avutil/hwcontext_cuda: fix yuv420p V/U plane overlap in cuda_get_buffer()

Odd-height yuv420p result in incorrect calculations of the U-plane
address offset. The last row of the V-plane overlapped with and was
overwritten by the first row of the U-plane, leading to chroma artifacts.

```
ffmpeg -init_hw_device cuda=cu -filter_hw_device cu -f lavfi -i \
testsrc=s=1920x1081,format=yuv420p -vf hwupload -c:v hevc_nvenc \
-vframes 1 -y <OUTPUT>
```

Signed-off-by: nyanmisaka <nst799610810@gmail.com>
(cherry picked from commit 3f6bf150cb)
Signed-off-by: Marvin Scholz <epirat07@gmail.com>

.forgejo/CODEOWNERS

@@ -57,7 +57,6 @@ libavcodec/.*png.* @Traneptora
 libavcodec/.*prores.* @lynne
 libavcodec/rangecoder.* @michaelni
 libavcodec/ratecontrol.* @michaelni
-libavcodec/rkmpp* @quink
 libavcodec/rv60.* @pross
 libavcodec/sgirle.* @pross
 libavcodec/.*siren.* @lynne
@@ -110,7 +109,6 @@ libavfilter/vf_find_rect.* @michaelni
 libavfilter/vf_icc.* @haasn
 libavfilter/vf_libplacebo.* @haasn
 libavfilter/vf_libvmaf.* @kylophone
-libavfilter/vf_mpdecimate.* @dana-feng
 libavfilter/vf_premultiply.* @haasn
 libavfilter/vf_scale.* @haasn
 libavfilter/vf_scale_vt.* @quink
@@ -141,8 +139,6 @@ libavformat/electronicarts.* @pross
 libavformat/.*exif.* @Traneptora
 libavformat/filmstrip.* @pross
 libavformat/frm.* @pross
-libavformat/hls.* @kasper93
-libavformat/hxvs.* @quink
 libavformat/iamf.* @jamrial
 libavformat/icecast.c @ePirat
 libavformat/ico.* @pross
@@ -157,7 +153,6 @@ libavformat/mlv.* @pross
 libavformat/mm.* @pross
 libavformat/msp.* @pross
 libavformat/mv.* @pross
-libavformat/ogg.* @toots
 libavformat/pp_bnk.* @zane
 libavformat/rm.* @pross
 libavformat/sauce.* @pross
@@ -198,7 +193,6 @@ libavutil/aarch64/.* @lynne @mstorsjo
 libavutil/arm/.* @mstorsjo
 libavutil/ppc/.* @sean_mcg
 libavutil/riscv/.* @Courmisch
-libavutil/wasm/.* @quink
 libavutil/x86/.* @lynne
 
 # swresample
@@ -232,26 +226,8 @@ doc/.* @GyanD
 # tests
 # =====
 tests/checkasm/riscv/.* @Courmisch
-libavutil/tests/ambient_viewing_environment.* @MarcosAsh
-libavutil/tests/buffer.* @MarcosAsh
-libavutil/tests/csp.* @MarcosAsh
-libavutil/tests/dovi_meta.* @MarcosAsh
-libavutil/tests/hdr_dynamic_vivid_metadata.* @MarcosAsh
-libavutil/tests/mastering_display_metadata.* @MarcosAsh
-libavutil/tests/tdrdi.* @MarcosAsh
-libavutil/tests/timestamp.* @MarcosAsh
-libavutil/tests/video_hint.* @MarcosAsh
 tests/ref/.*drawvg.* @ayosec
-tests/ref/fate/ambient_viewing_environment @MarcosAsh
-tests/ref/fate/buffer @MarcosAsh
-tests/ref/fate/csp @MarcosAsh
-tests/ref/fate/dovi_meta @MarcosAsh
-tests/ref/fate/hdr_dynamic_vivid_metadata @MarcosAsh
-tests/ref/fate/mastering_display_metadata @MarcosAsh
 tests/ref/fate/sub-mcc.* @programmerjake
-tests/ref/fate/tdrdi @MarcosAsh
-tests/ref/fate/timestamp @MarcosAsh
-tests/ref/fate/video_hint @MarcosAsh
 
 # Forgejo
 # =======

.forgejo/PULL_REQUEST_TEMPLATE.md

@@ -1,14 +0,0 @@
-# Summary of changes
-
-Briefly describe what this PR does and why.
-
-<!--
-If this PR requires new FATE test samples, attach them to the PR and
-list their target paths below (relative to the fate-suite root).
-
-Attached filenames must match the sample's filename:
-
-```fate-samples
-# e.g. vorbis/new-sample.ogg
-```
--->

.forgejo/actions/rebase-pr/action.yml

@@ -1,40 +0,0 @@
-name: Rebase PR onto target
-
-inputs:
-  path:
-    description: Directory to run in
-    required: false
-    default: .
-
-runs:
-  using: composite
-  steps:
-    - name: Rebase onto target branch
-      if: ${{ forge.event_name == 'pull_request' }}
-      working-directory: ${{ inputs.path }}
-      run: |
-        set -x
-        # Fetch all PR commits + merge-base
-        commits=$(curl -s -o /dev/null -w '%header{X-Total-Count}' '${{ forge.api_url }}/repos/${{ forge.repository }}/pulls/${{ forge.event.pull_request.number }}/commits?limit=1&verification=false&files=false')
-        git fetch -vv --deepen=$commits origin pull/${{ forge.event.pull_request.number }}/head
-
-        if [ "$(wc -l < .git/shallow)" -ne 1 ]; then
-            echo "Complex PR history detected, falling back to merge"
-            git fetch -vv --unshallow origin pull/${{ forge.event.pull_request.number }}/head
-            git fetch -vv origin ${{ forge.base_ref }}
-            git merge --no-edit origin/${{ forge.base_ref }} \
-              || { echo "Does not merge cleanly with ${{ forge.base_ref }}"; exit 1; }
-            exit 0
-        fi
-
-        # Unfortunately, Forgejo does not update the merge-base when PR branch
-        # is updated, the `forge.event.pull_request.merge_base` is stale value,
-        # from the time when PR was created. We cannot rely on it, so try to
-        # infer parent commit of the PR from the shallow clone that we have.
-        base=$(cat .git/shallow)
-
-        # Fetch target branch
-        git fetch -vv --depth=1 origin ${{ forge.base_ref }}
-        # Rebase PR commits onto the current state of target branch
-        git rebase --onto origin/${{ forge.base_ref }} $base \
-          || { echo "Does not rebase cleanly onto ${{ forge.base_ref }}"; exit 1; }

.forgejo/inject-pr-samples.py

@@ -1,174 +0,0 @@
-#!/usr/bin/env python3
-# Copyright (c) 2026 Romain Beauxis <romain.beauxis@gmail.com>
-#
-# Redistribution and use in source and binary forms, with or without
-# modification, are permitted provided that the following conditions are met:
-#
-# 1. Redistributions of source code must retain the above copyright notice,
-#    this list of conditions and the following disclaimer.
-# 2. Redistributions in binary form must reproduce the above copyright notice,
-#    this list of conditions and the following disclaimer in the documentation
-#    and/or other materials provided with the distribution.
-#
-# THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS"
-# AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
-# IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
-# ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
-# LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
-# CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
-# SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
-# INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
-# CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
-# ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
-# POSSIBILITY OF SUCH DAMAGE.
-
-"""Inject PR attachment samples into the fate-suite directory.
-
-Usage: inject-pr-samples.py <pr-number>
-
-Reads SAMPLES from the environment (defaults to fate-suite).  For each path
-listed in a ```fate-samples``` block in the PR description, downloads the
-matching PR attachment into $SAMPLES/<path>.
-
-The PR description should contain a block like:
-
-  ```fate-samples
-  vorbis/tos.ogg
-  mov/some-new-sample.mov
-  ```
-
-Each filename must match a file attached to the PR.
-"""
-
-import hashlib
-import json
-import os
-import re
-import sys
-import tempfile
-import urllib.request
-from pathlib import Path, PurePosixPath
-
-FORGEJO_API = "https://code.ffmpeg.org/api/v1/repos/ffmpeg/ffmpeg/issues"
-ATTACHMENT_BASE = "https://code.ffmpeg.org/attachments/"
-
-
-def fetch_json(url):
-    with urllib.request.urlopen(url) as r:
-        return json.load(r)
-
-
-def parse_fate_samples(body):
-    paths = []
-    in_block = False
-    for line in body.splitlines():
-        if line == "```fate-samples":
-            in_block = True
-        elif line == "```" and in_block:
-            break
-        elif in_block:
-            parts = line.split()
-            if len(parts) == 1:
-                paths.append(parts[0])
-    return paths
-
-
-MAX_PATH_DEPTH = 3
-
-
-def validate_path(path):
-    p = PurePosixPath(path)
-    if p.is_absolute():
-        raise ValueError(f"path must be relative: {path!r}")
-    if ".." in p.parts:
-        raise ValueError(f"path must not contain '..': {path!r}")
-    if not p.parts:
-        raise ValueError(f"empty path")
-    if len(p.parts) > MAX_PATH_DEPTH:
-        raise ValueError(f"path too deep (max {MAX_PATH_DEPTH} components): {path!r}")
-
-
-def validate_url(url):
-    if not url.startswith(ATTACHMENT_BASE):
-        raise ValueError(f"unexpected attachment URL: {url!r}")
-
-
-def digest(path):
-    h = hashlib.sha256()
-    with open(path, "rb") as f:
-        while chunk := f.read(1 << 16):
-            h.update(chunk)
-    return h.digest()
-
-
-def download(url, dst):
-    dst.parent.mkdir(parents=True, exist_ok=True)
-    with tempfile.NamedTemporaryFile(dir=dst.parent, delete=False) as tmp:
-        tmp_path = Path(tmp.name)
-        try:
-            with urllib.request.urlopen(url) as r:
-                while chunk := r.read(1 << 16):
-                    tmp.write(chunk)
-            if dst.exists() and digest(dst) != digest(tmp_path):
-                raise ValueError(f"already exists with different content: {dst}")
-            tmp_path.rename(dst)
-        except:
-            tmp_path.unlink(missing_ok=True)
-            raise
-
-
-def main():
-    if len(sys.argv) != 2 or not re.fullmatch(r"[0-9]+", sys.argv[1]):
-        print(f"Usage: {sys.argv[0]} <pr-number>", file=sys.stderr)
-        sys.exit(1)
-
-    pr_number = sys.argv[1]
-    samples_dir = Path(os.environ.get("SAMPLES", "fate-suite"))
-
-    pr = fetch_json(f"{FORGEJO_API}/{pr_number}")
-    assets = {a["name"]: a["browser_download_url"] for a in pr.get("assets", [])}
-    paths = parse_fate_samples(pr.get("body", ""))
-
-    if not paths:
-        sys.exit(0)
-
-    new_samples = False
-
-    for path in paths:
-        try:
-            validate_path(path)
-        except ValueError as e:
-            print(f"fate-samples: {e}", file=sys.stderr)
-            sys.exit(1)
-
-        name = PurePosixPath(path).name
-        url = assets.get(name)
-        if url is None:
-            print(f"fate-samples: no attachment named {name!r}", file=sys.stderr)
-            sys.exit(1)
-
-        try:
-            validate_url(url)
-        except ValueError as e:
-            print(f"fate-samples: {e}", file=sys.stderr)
-            sys.exit(1)
-
-        dst = samples_dir / path
-        is_new = not dst.exists()
-        try:
-            download(url, dst)
-        except ValueError as e:
-            print(f"fate-samples: {e}", file=sys.stderr)
-            sys.exit(1)
-        if is_new:
-            new_samples = True
-        print(f"Injected: {path}")
-
-    output_file = os.environ.get("FORGEJO_OUTPUT")
-    if output_file:
-        with open(output_file, "a") as f:
-            print(f"new_samples={'true' if new_samples else 'false'}", file=f)
-
-
-if __name__ == "__main__":
-    main()

.forgejo/pre-commit/config.yaml

@@ -1,14 +1,5 @@
 exclude: ^tests/ref/
 
-# This repository keeps its pre-commit config in this nondefault path.
-# Install hooks from here with:
-#   pre-commit install -c .forgejo/pre-commit/config.yaml
-# The command above installs both the pre-commit and commit-msg hooks because
-# default_install_hook_types includes them below.
-default_install_hook_types:
-  - pre-commit
-  - commit-msg
-
 repos:
 - repo: https://github.com/pre-commit/pre-commit-hooks
   rev: v5.0.0
@@ -29,15 +20,9 @@ repos:
     - id: trailing-whitespace
 - repo: local
   hooks:
-    - id: check-commit-message
-      name: validate commit message format
-      language: script
-      entry: ./tools/check_commit_msg.sh
-      stages: [commit-msg]
-      always_run: true
-    - id: arm-asm-indent
-      name: fix arm/aarch64 assembly indentation
-      files: ^.*/(arm|aarch64)/.*\.S$
+    - id: aarch64-asm-indent
+      name: fix aarch64 assembly indentation
+      files: ^.*/aarch64/.*\.S$
       language: script
       entry: ./tools/check_arm_indent.sh --apply
       pass_filenames: false
@@ -48,4 +33,4 @@ repos:
       args:
         - --ignore-words=.forgejo/pre-commit/ignored-words.txt
         - --ignore-multiline-regex=codespell:off.*?(codespell:on|\Z)
-      exclude: ^tools/(patcheck|clean-diff)|tests/checkasm/ext/.*$
+      exclude: ^tools/(patcheck|clean-diff)$

.forgejo/pre-commit/ignored-words.txt

@@ -69,7 +69,6 @@ pEvents
 PixelX
 Psot
 quater
-re-use
 readd
 recuse
 redY

.forgejo/workflows/autolabel.yml

@@ -22,7 +22,6 @@ jobs:
         with:
           configuration-path: .forgejo/labeler/labeler.yml
           repo-token: ${{ secrets.AUTOLABELER_TOKEN }}
-          sync-labels: true
       - name: Label by title-match
         uses: actions/github-script@v8
         with:

.forgejo/workflows/lint.yml

@@ -16,8 +16,6 @@ jobs:
     steps:
       - name: Checkout
         uses: actions/checkout@v6
-      - name: Rebase onto target branch
-        uses: ./.forgejo/actions/rebase-pr
       - name: Install pre-commit CI
         id: install
         run: |

.forgejo/workflows/test.yml

@@ -3,7 +3,7 @@ name: Test
 on:
   push:
     branches:
-      - master
+      - release/8.1
   pull_request:
 
 concurrency:
@@ -29,8 +29,6 @@ jobs:
     steps:
       - name: Checkout
         uses: actions/checkout@v6
-      - name: Rebase onto target branch
-        uses: ./.forgejo/actions/rebase-pr
       - name: Configure
         run: |
           ./configure --enable-gpl --enable-nonfree --enable-memory-poisoning --assert-level=2 \
@@ -60,42 +58,26 @@ jobs:
         with:
           path: fate-suite
           key: fate-suite-${{ steps.fate.outputs.hash }}
-      - name: Inject PR Samples
-        id: inject
-        if: ${{ forge.event_name == 'pull_request' }}
-        run: SAMPLES=$PWD/fate-suite .forgejo/inject-pr-samples.py ${{ forge.event.pull_request.number }}
       - name: Run Fate
         run: |
           LD_LIBRARY_PATH="$(printf "%s:" "$PWD"/lib*)$PWD" make fate fate-build SAMPLES="$PWD/fate-suite" -j$(nproc) || FATERES=$?
           find . -name "*.err" -exec printf '::group::%s\n' {} \; -exec cat {} \; -exec printf '::endgroup::\n' \;
           exit ${FATERES:-0}
-      - name: Fail if new samples were injected
-        if: ${{ steps.inject.outputs.new_samples == 'true' }}
-        run: |
-          echo "New FATE samples were injected from PR attachments. Please add them to the official fate-suite before merging."
-          exit 1
   run_fate_full:
     name: Fate (Full, ${{ matrix.target_exec }})
     strategy:
       fail-fast: false
       matrix:
-        image: ['ghcr.io/btbn/ffmpeg-builds/win64-gpl:latest']
+        image: ['ghcr.io/btbn/ffmpeg-builds/win64-gpl-8.1:latest']
         target_exec: ['wine']
     runs-on: linux-amd64
     container: ${{ matrix.image }}
     steps:
       - name: Checkout
         uses: actions/checkout@v6
-        with:
-          path: ffmpeg
-      - name: Rebase onto target branch
-        uses: ./ffmpeg/.forgejo/actions/rebase-pr
-        with:
-          path: ffmpeg
       - name: Configure
         run: |
-          mkdir build && cd build
-          ../ffmpeg/configure --pkg-config-flags="--static" $FFBUILD_TARGET_FLAGS $FF_CONFIGURE \
+          ./configure --pkg-config-flags="--static" $FFBUILD_TARGET_FLAGS $FF_CONFIGURE \
               --cc="$CC" --cxx="$CXX" --ar="$AR" --ranlib="$RANLIB" --nm="$NM" \
               --extra-cflags="$FF_CFLAGS" --extra-cxxflags="$FF_CXXFLAGS" \
               --extra-libs="$FF_LIBS" --extra-ldflags="$FF_LDFLAGS" --extra-ldexeflags="$FF_LDEXEFLAGS" \
@@ -105,7 +87,7 @@ jobs:
           cat ffbuild/config.log
           exit ${CFGRES:-0}
       - name: Build
-        run: make -C build -j$(nproc)
+        run: make -j$(nproc)
       - name: Restore Cached Fate-Suite
         id: cache
         uses: actions/cache/restore@v5
@@ -117,7 +99,7 @@ jobs:
       - name: Sync Fate-Suite
         id: fate
         run: |
-          make -C build fate-rsync SAMPLES=$PWD/fate-suite
+          make fate-rsync SAMPLES=$PWD/fate-suite
           echo "hash=$(find fate-suite -type f -printf "%P %s %T@\n" | sort | sha256sum | cut -d' ' -f1)" >> $FORGEJO_OUTPUT
       - name: Cache Fate-Suite
         uses: actions/cache/save@v5
@@ -125,21 +107,12 @@ jobs:
         with:
           path: fate-suite
           key: fate-suite-${{ steps.fate.outputs.hash }}
-      - name: Inject PR Samples
-        id: inject
-        if: ${{ forge.event_name == 'pull_request' }}
-        run: SAMPLES=$PWD/fate-suite ffmpeg/.forgejo/inject-pr-samples.py ${{ forge.event.pull_request.number }}
       - name: Run Fate
         run: |
           if [[ "${{ matrix.target_exec }}" == "wine" ]]; then
             export WINEDEBUG=-all
             export FREI0R_PATH="$(winepath -w "$FREI0R_PATH")"
           fi
-          LD_LIBRARY_PATH="$(printf "%s:" "$PWD"/lib*)$PWD" make -C build fate fate-build SAMPLES="$PWD/fate-suite" -j$(nproc) || FATERES=$?
+          LD_LIBRARY_PATH="$(printf "%s:" "$PWD"/lib*)$PWD" make fate fate-build SAMPLES="$PWD/fate-suite" -j$(nproc) || FATERES=$?
           find . -name "*.err" -exec printf '::group::%s\n' {} \; -exec cat {} \; -exec printf '::endgroup::\n' \;
           exit ${FATERES:-0}
-      - name: Fail if new samples were injected
-        if: ${{ steps.inject.outputs.new_samples == 'true' }}
-        run: |
-          echo "New FATE samples were injected from PR attachments. Please add them to the official fate-suite before merging."
-          exit 1

.gitignore

@@ -26,9 +26,6 @@
 *.spv
 *.spv.c
 *.spv.gz
-*.gen.asm
-*.gen.c
-*.gen.S
 *.ptx
 *.ptx.c
 *.ptx.gz
@@ -40,9 +37,6 @@
 /ffmpeg
 /ffplay
 /ffprobe
-/checkasm_config_generated.asm
-/checkasm_config_generated.h
-/checkasm_header_config_generated.h
 /config.asm
 /config.h
 /config_components.asm

CONTRIBUTING.md

@@ -1,5 +1,4 @@
 # Note to Github users
-
 Patches should be submitted to [Forgejo](https://code.ffmpeg.org/FFmpeg/FFmpeg/pulls) or the [ffmpeg-devel mailing list](https://ffmpeg.org/mailman/listinfo/ffmpeg-devel) using `git format-patch` or `git send-email`. Github pull requests should be avoided because they are not part of our review process and **will be ignored**.
 
 See [https://ffmpeg.org/developer.html#Contributing](https://ffmpeg.org/developer.html#Contributing) for more information.

Changelog

@@ -1,32 +1,245 @@
 Entries are sorted chronologically from oldest to youngest within each release,
 releases are sorted from youngest to oldest.
 
-version <next>:
-- extensively improved AAC encoder
-- APV Vulkan encoder
+
+version 8.1.2:
+ swscale/x86/rgb_2_rgb: fix uyvytoyuv422 overwrite on odd width
+ swscale/aarch64: fix uyvy/yuyv to yuv420p/yuv422p on odd width
+ avcodec/snowenc: fix SIGFPE in get_dc() when a block lies outside the plane
+ avcodec/snowenc: fix out-of-bounds OBMC read in get_dc() for narrow planes
+ (fforge/pr/23505) avformat/tls_gnutls:fix crash when connecting to peer
+ libavcodec/jpeg2000htdec: remove trailing whitespace
+ libavdevice/alsa.c: fix NULL pointer dereference
+ avcodec/libjxlenc: check orientation tag metadata before reading
+ avformat/icecast: reject CR/LF in metadata header values
+ avfilter/avf_showspectrum: Fix allocation check
+ avformat/sctp: add size check in sctp_read() matching sctp_write()
+ avcodec/jpeg2000: Fix undefined behavior on ROI shift-up
+ aacdec_usac_mps212: reject reserved freq_res value
+ avcodec/aac/aacdec_usac_mps212: Off-by-one bounds check in ff_aac_ec_data_deci()
+ Update for 8.1.2
+ avcodec/snowenc: fix out-of-bounds memcpy in get_block_rd() for narrow planes
+ avutil/eval: apply unary sign to print, squish, gauss and lerp
+ avcodec/cbs_av1_syntax_template: reset seen_frame_header on sequence headers
+ avfilter/convolution: compute user matrix products in unsigned
+ avformat/mpegts: use av_fast_realloc() for prg
+ avfilter/avf_showcwt: fix DIRECTION_DU EOF fill clearing the wrong rows
+ avfilter/avf_showcwt: fix DIRECTION_RL EOF fill clearing the wrong columns
+ avfilter/avf_showcwt: avoid undefined float to int conversion of nb_consumed_samples
+ avfilter/avf_showcwt: fix out of array read in compute_kernel
+ avfilter/v360: compute remap table offsets in 64bit
+ avfilter/v360: reject out-of-range dimensions
+ swresample/x86/resample: write only int16 in the int16 resampler
+ avformat/rtpenc_amr: Check input size
+ swscale/ppc: fix ASAN stack-buffer-overflow in yuv2planeX
+ swscale/ppc: fix LOAD_FILTER overread in VSX path
+ avcodec/sanm: reject codec37 frames taller than the allocated buffer
+ avcodec/sanm: reject codec47 frames taller than the allocated buffer
+ avfilter: use ff_slice_pos() for per-slice boundary computation
+ avfilter: add ff_slice_pos() helper for slice boundaries
+ (origin/release/8.1, fforge/release/8.1) avformat/iamf_writer: reject muxing PCM streams
+ avfilter/estdif: avoid signed overflow in slice boundary calculation
+ swscale: support sliced input with cascaded scaling contexts
+ avformat/rtspdec: bound Content-Length in the ANNOUNCE handler to SDP_MAX_SIZE
+ avcodec/cbs_h266_syntax_template: reject subpic info with res_change_in_clvs
+ avcodec/misc4: Check nb channels
+ avcodec/rv10, rv34: check init_get_bits8() before RealVideo bit access
+ avformat/http: reject request-line tokens not terminated by whitespace
+ avformat/mov: reject out of range ispe dimensions, avoid overflow summing HEIF tile dimensions
+ avcodec/agm: validate actual src_y against prev plane in decode_inter_plane
+ avformat/dhav: Fix second integer overflow in get_duration()
+ fftools/ffmpeg_dec: deep-copy subtitle_header to fix use-after-free
+ avcodec/hevc/ps: Check window parameters
+ avcodec/hevc/ps: Factor window reading out
+ avcodec/truespeech: reject iterations count whose * 240 product overflows 32-bit
+ libavcodec/options_table: gamma22 and gamma28 aliases
+ avcodec/on2avc: reject subframe count whose * SUBFRAME_SIZE product overflows 32-bit
+ avfilter/zmq: initialize send_buf before shared cleanup on parse failure
+ avcodec/adpcm: fix signed integer overflow in get_nb_samples()
+ avformat/matroskadec: avoid signed overflow in DASH cue time differences
+ avcodec/fastaudio: reject subframes count whose * 256 product overflows 32-bit
+ avcodec/vc2enc_dwt: avoid signed overflow in the 9/7 DWT lifting
+ avcodec/vc2enc_dwt: avoid signed overflow in the 5/3 and Haar DWT
+ avformat/dashdec: Fail with any inner stream count being 0
+ avcodec/mjpegdec: require progress in AVRn interlaced field loop
+ avcodec/mwsc: do not dereference a missing reference frame
+ avcodec/misc4: reject invalid sample rate
+ swscale/output: avoid signed overflow in yuv2rgba64_1 alpha
+ swscale/output: avoid signed overflow in yuv2rgba64_full_1 alpha
+ tools/target_dem_fuzzer: do not exit on io_buffer allocation failure
+ avformat/gxfenc: Check timecode and propagate error
+ swscale/rgb2rgb_template: use unsigned for <<24
+ avformat/iamf_parse: bound substream count by remaining OBU size
+ avformat/matroskadec: bound TRACKENTRY parsing by max_streams
+ avcodec/diracdec: fix heap buffer overflow in edge_emu_buffer
+ avformat/rtmppkt: Check recursion depth
+ avcodec/tdsc: propagate max_pixels to the JPEG tile decoder
+ avcodec/imm5: propagate max_pixels to the H264/HEVC sub-decoders
+ avcodec/cri: propagate max_pixels to the JPEG tile decoder
+ avcodec/jpeglsdec: only apply color transform to decoded rows
+ avcodec/adpcm: require block_align to be a multiple of channels in ADPCM_PSXC init
+ avfilter/avf_showcwt: fix out-of-bounds read in du scroll
+ avfilter/f_ebur128: avoid signed-int wrap when sizing per-channel cache
+ avformat/mov: cap HEIF ICC profile copies via c*max_streams to bound CPU and memory
+ avcodec/aac/aacdec_usac: reject explicit usacSamplingFrequency of 0
+ avcodec/aac/aacdec_usac: avoid signed overflow in decode_tsd
+ avcodec/aac/aacdec: reject decoded frame without a valid sample rate
+ avformat/iff: check av_get_packet() result in ANIM branch
+ avcodec/cbs_h266_syntax_template: Fix pps_exp_slice_height_in_ctus_minus1 range
+ avfilter/vf_scale: split rational multiply
+ avcodec/bsf/dts2pts: fix binary tree invariant violation on selective dec_poc
+ avfilter/vf_drawtext: Avoid double free in glyph_enu_border_free()
+ avfilter/vf_drawtext: plug error-path leaks in measure_text/draw_text
+ avfilter/vf_drawtext: shape_text_hb() free allocated things on error
+ avfilter/vf_drawtext: avoid double-free of aliased FT_Glyph in glyph_enu_free
+ avfilter/vf_drawtext: don't double-free glyph that has been cached in tree
+ avfilter/vf_drawtext: always check pixel_mode == FT_PIXEL_MODE_MONO
+ avcodec/cook: bound subpacket channel sum against channel count
+ avcodec/apv_decode: avoid using apv_cbc
+ avformat/mxfdec: Remove unneeded check
+ avformat/ftp: Check string used for RNTO
+ avformat/ftp: Check for Telnet IAC characters and other non printable ASCII chars
+ avformat/ftp: reject CR/LF in the URL path to prevent FTP command injection
+ avcodec/jpeg2000dec: Clear header derived variables
+ avcodec/bsf/smpte436m_to_eia608: properly frees stuff on errors in ff_smpte436m_to_eia608_filter()
+ tests/checkasm/crc: retain offset values between calls
+ avformat/avc: Adjust get_ue_golomb() to handle 32 bit
+ avformat/whip: require remote DTLS fingerprint in SDP answer
+ avcodec/x86/vorbisdsp: change cmpleps to cmpltps in inverse coupling
+ avcodec/liboapvenc: derive and validate APV profile from pixel format
+ avformat/dashdec: bound manifest reloads and fragment-open retries
+ avfilter/af_join: fix wrong loop bound in buffer dedup (use-after-free)
+ avcodec/nvenc: fix compatibility with Video Codec SDK 13.1
+ avformat/mov: validate APV access unit length before passing to decoder
+ avcodec/h264_slice: guard color_frame() against chroma-width underflow
+ avcodec/magicyuv: reject slice_height misaligned with chroma vshift
+ avcodec/magicyuv: Expand the s->interlaced slice-height sanity check
+ avcodec/magicyuv: Fix 1 line MEDIAN slices
+ avformat/mxfdec: zero-init Sony MPEG-4 extradata and add padding
+ avformat/soxdec: Check sample_rate for nan
+ tests/tiny_ssim: fixed mistake in ssim_c1 calculation
+ avformat/hls: Check TIME-OFFSET value
+ avformat/hls: Check url_offset and size
+ avcodec/prores_raw: reject invalid tile alignment values
+ avformat/ty: check rec_size
+ avformat/ogg: Fix overflow and stale oggvorbis_private values
+ avformat/rtpenc_xiph: bail out when the max payload size underflows
+ avformat/rtpenc_aac: reject packets smaller than the ADTS header
+ avcodec/jpeg2000dec: compute mask in decode_clnpass() like in decode_sigpass()
+ avformat/flvdec: Check size at the top of the main loop
+ avformat/vividas: fix misaligned access
+ avformat/mccdec: dont pass NULL to bytestream2_put_buffer()
+ avcodec/g2meet: The stack is EPIC_PIX_STACK_SIZE
+ avcodec/h2645_sei: Initialize side data before deallocation
+ avfilter/boxblur: Fix off by one errors
+ avformat/assenc: Add the missing parentheses
+ avcodec/diracdec: Enlarge `mctmp` to cover the worst-case `blheight·ybsep + yblen` rows, and break the MC loop when no output rows remain
+ tools/zmqsend: free the AVBprint buffer after using it
+ [Wave] Fix issues with unaligned metadata chunks.
+ avformat/mpegts: Dont assume fc->priv_data is a MpegTSContext
+ fftools/graph: Add missing include "libavutil/mem.h" for fftools/graph/graphprint.c
+ avcodec/cbs_h266_syntax_template: tighten sh_num_tiles_in_slice_minus1 upper bound
+ avcodec/hevc: limit missing-ref fill to coded planes
+ avformat/mov: Fix negative index given to can_seek_to_key_sample()
+ avcodec/hdrdec: fix pixel count decrement in RLE decompress loop
+ (fforge/pr/23464) forgejo/workflows: update test workflow for 8.1 release
+ (fforge/pr/23445) aarch64: vp9lpf: Fix GCS violations
+ (fforge/pr/23303) avformat/mov: don't abort on unsupported or invalid chnl boxes
+ avformat/mov_chan: keep the layout untouched on chan/chnl box failure
+ (fforge/pr/23240) avformat/oggparsevorbis.c: Prevent integer overflow when summing header lengths; add bounds check.
+ (fforge/pr/23187) avformat/oggparsecelt: bound extra_headers to avoid an effectively infinite loop
+ (fforge/pr/23111) avformat/demux: use correct close function for custom io
+ avformat/hlsenc: use correct close function for custom io
+ avformat/hlsenc: respect io_open set in AVFormatContext
+ avformat/dashenc: respect io_open set in AVFormatContext
+ avformat/dashdec: respect io_open set in AVFormatContext
+ vulkan: fix using encode caps before querying
+ avformat/rtpdec_av1: fix buffer overflow due to variable confusion
 
 
-version 9.0:
-- Extend AMF Color Converter (vf_vpp_amf) HDR capabilities
-- LCEVC track muxing support in MP4 muxer
-- Playdate video encoder and muxer
-- Add v360_vulkan filter
-- HE-AAC 960 decoding (DAB+)
-- transpose_cuda filter
-- Add AMF Frame Rate Converter (vf_frc_amf) filter
-- SMPTE 2094-50 metadata support and passthrough
-- ProRes RAW VideoToolbox hwaccel
-- APV Vulkan hwaccel
-- Animated WebP decoder
-- Animated WebP demuxer
-- Remove CELT decoding support (doesn't affect Opus CELT)
-- Remove ogg/celt parsing
-- Bitstream filter to split Dolby Vision multi-layer HEVC
-- Add AMF hardware memory mapping support.
-- ONNX Runtime DNN backend with GPU execution provider support
-- Remove deprecated NVENC options and support for pre-11.1 SDK versions
-- iTerm2 inline image protocol muxer
-
+version 8.1.1:
+ avcodec/aac/aacdec_usac_mps212: fix attach_lsb() OOB after huff_decode
+ avcodec/dfpwmdec: Check nb_samples
+ avcodec/alsdec: do not set nbits invalidly
+ swscale/swscale_unscaled: adjust last line copy
+ swscale/swscale: Check srcSliceY and srcSliceH
+ avformat/avidec: check LIST size in avi_load_index()
+ avformat/avidec: validate INFO list size before parsing
+ avformat/matroskadec: Check audio.sub_packet_h * audio.frame_size
+ libavformat/xwma: fix overflow in seek position
+ avformat/pcm: Use 64bit for byte_rate
+ avcodec/adpcm: signed integer overflow in ADPCM_N64
+ avcodec/hevc/ps: validate rep_format dimensions in multi-layer SPS
+ avfilter/vf_kerndeint: Check for minimum height
+ avcodec/ralf: Add the missing return statement after the error log
+ avfilter/vf_codecview: Clamp block to the visible frame region
+ avcodec/zmbv: reject XOR data that overruns the decompression buffer
+ avcodec/rasc: fix heap use-after-free in decode_move()
+ avformat/rtpdec_mpeg4: reject zero-length AU header sections
+ avcodec/hevc/refs: Check multiplication in alloc_frame()
+ fftools/ffmpeg_opt: validate stream index in negative map handling
+ avformat/rtmpproto: prevent integer overflow accumulating FLV buffer size
+ avformat/rtmpproto: validate compressed SWF header length
+ avformat/rtsp: Fix out-of-bounds read in SDP parser when control_url is empty
+ avformat/rtpdec_latm: avoid integer overflow in LATM length parsing
+ avcodec/h264: recompute per-slice direct mode state for every slice
+ avcodec/h264_refs: Clear stale pointers from ref_list
+ avcodec/pdvdec: Check input space before buffer allocation
+ avformat/concat: guard total_size overflow
+ avformat/iamf_parse.c: Fix potential integer overflow in opus_decoder_config()
+ avcodec/wmaenc: Fix missing padding in extradata
+ avformat: Fix various extradata padding issues
+ avcodec/tdsc: remove double stride adjustment
+ avformat/cafdec: fix negative index use in read_seek
+ avcodec/notchlc: Check 255 loops
+ avformat/rtpdec_jpeg: check qtable_len
+ avformat/vividas: use-of-uninitialized-value in keybuffer
+ avcodec/tdsc: Check jpeg size
+ avcodec/tdsc: Better input size check
+ avcodec/tdsc: Check tile_size
+ avcodec/decode: Better documentation for ff_set_dimensions()
+ avformat/mov: Limit maximum box size for mov_read_lhvc()
+ avformat/mov: reject dimg references with zero entries
+ avformat/mov: check extradata in mov_read_dops()
+ avformat/mov: Check read size for opus extradata
+ avformat/rtspdec: reject non-positive ANNOUNCE Content-Length
+ avformat/wavdec: Fix use-of-uninitialized-value in find_guid()
+ lavfi: vf_drawtext: check memory allocation
+ avcodec/svq1dec: Check input space for minimum
+ avcodec/vp9: Rollback dimensions when format is rejected
+ avformat/rtpdec_qdm2: Check block_size
+ avfilter/vf_ssim360: fix integer overflow in tape_length allocation
+ avcodec/escape130: Initialize old_y_avg
+ avutil/samplefmt: Dont claim that av_get_sample_fmt_string checks sample_fmt
+ swscale: fix signed integer overflow in color conversion arithmetic
+ avformat: check avio_read() return values in dss/dtshd/mlv
+ avformat/whip: check RTP history packet size before RTX retransmission
+ avcodec/aac/aacdec_usac: Implement missing bits of otts_bands_phase and residual_bands computation
+ avcodec/alsdec: preserve full float value in zero-truncated samples
+ avcodec/alsdec: propagate read_diff_float_data() errors in read_frame_data()
+ avcodec/alsdec: fix mantissa unpacking in compressed Part A path
+ avfilter/vf_drawtext: fix newline rendered as .notdef glyph
+ avcodec/pngdec: fix dead overflow check in decode_text_to_exif()
+ avcodec/vp9: fix cbs fragment leak on error
+ libavfilter/vf_v360: fix operator precedence in stereo loop condition
+ avcodec/alsdec: fix abs(INT_MIN) UB in read_diff_float_data()
+ avformat/rsd: reject short ADPCM_THP extradata reads
+ avformat/mov: Handle integer overflow in MOV parser
+ avcodec/dvdsub_parser: Fix buf_size check
+ avfilter/af_pan: fix sscanf() return value checks in parse_channel_name
+ avutil/bswap: fix implicit conversion warning in av_bswap64
+ avcodec/h2645_sei: don't use provider_code uninitialized
+ (fforge/pr/22943) avformat/hls_sample_encryption: add missing padding for audio setup buffer
+ (fforge/pr/22826) lavc/videotoolbox_vp9: fix vpcC flags offset
+ lavc/videotoolboxenc: return SEI parse errors
+ vulkan: fix encode feedback query handling
+ avcodec/cbs_lcevc: fix writing process blocks with size 6
+ avformat/mov: don't try to create an LCEVC group if there's a single track
+ (fforge/pr/22563) aarch64: Add Armv9.3-A GCS (Guarded Control Stack) support
+ aarch64: hevcdsp: Make returns match the call site
+ (fforge/pr/22545) avcodec/lcevcdec: don't try to derive final dimensions from SAR
+ avcodec/lcevc_parser: move the resolution type table to a header
+ avcodec/cbs_lcevc: don't look for process blocks if the unit was not decomposed
 
 version 8.1:
 - ffprobe -codec option

INSTALL.md

@@ -1,4 +1,4 @@
-# Installing FFmpeg
+## Installing FFmpeg
 
 0. If you like to include source plugins, merge them before configure
 for example run tools/merge-all-source-plugins
@@ -14,11 +14,15 @@ path when launching `configure`, e.g. `/ffmpegdir/ffmpeg/configure`.
 
 3. Type `make install` to install all binaries and libraries you built.
 
-## NOTICE
+NOTICE
+------
 
-- Non system dependencies (e.g. libx264, libvpx) are disabled by default.
+ - Non system dependencies (e.g. libx264, libvpx) are disabled by default.
 
-## NOTICE for Package Maintainers
+NOTICE for Package Maintainers
+------------------------------
 
-- It is recommended to build FFmpeg twice, first with minimal external dependencies so that 3rd party packages, which depend on FFmpegs libavutil/libavfilter/libavcodec/libavformat
-can then be built. And last build FFmpeg with full dependencies (which may in turn depend on some of these 3rd party packages). This avoids circular dependencies during build.
+ - It is recommended to build FFmpeg twice, first with minimal external dependencies so
+   that 3rd party packages, which depend on FFmpegs libavutil/libavfilter/libavcodec/libavformat
+   can then be built. And last build FFmpeg with full dependencies (which may in turn depend on
+   some of these 3rd party packages). This avoids circular dependencies during build.

MAINTAINERS

@@ -65,14 +65,14 @@ Communication
 =============
 website                                 (T: https://git.ffmpeg.org/ffmpeg-web) Deby Barbara Lepage
 fate.ffmpeg.org                         (L: fate-admin@ffmpeg.org) (W: https://fate.ffmpeg.org) (P: https://ffmpeg.org/fate.html) (S: https://git.ffmpeg.org/fateserver) Timo Rothenpieler
-Trac bug tracker                    [X] (W: https://trac.ffmpeg.org) Alexander Strasser, Michael Niedermayer, Carl Eugen Hoyos
-Forgejo issue tracker                   (W: https://code.ffmpeg.org/FFmpeg/FFmpeg/issues)
+Trac bug tracker                        (W: https://trac.ffmpeg.org) Alexander Strasser, Michael Niedermayer, Carl Eugen Hoyos
 Patchwork                           [2] (W: https://patchwork.ffmpeg.org) Andriy Gelman
 mailing lists                           (W: https://ffmpeg.org/contact.html#MailingLists) Baptiste Coudurier
 Twitter                                 Reynaldo H. Verdejo Pinochet
 Launchpad                               Timothy Gu
 ffmpeg-security                     [2] (L: ffmpeg-security@ffmpeg.org) (W: https://ffmpeg.org/security.html) Michael Niedermayer, Reimar Doeffinger
 
+
 libavutil
 =========
 
@@ -93,7 +93,6 @@ Other:
   hash                                  Reimar Doeffinger
   hwcontext_cuda*                       Timo Rothenpieler
   hwcontext_d3d12va*                    Wu Jianhua
-  hwcontext_oh*                         Zhao Zhili
   hwcontext_vulkan*                 [2] Lynne
   intfloat*                             Michael Niedermayer
   integer.c, integer.h                  Michael Niedermayer
@@ -159,7 +158,7 @@ Codecs:
   asv*                                  Michael Niedermayer
   atrac3plus*                           Maxim Poliakovski
   audiotoolbox*                         rcombs
-  avs2*                                 Huiwen Ren, Zhao Zhili
+  avs2*                                 Huiwen Ren
   bgmc.c, bgmc.h                        Thilo Borgmann
   binkaudio.c                           Peter Ross
   cavs*                                 Stefan Gehrer
@@ -233,7 +232,7 @@ Codecs:
   msvideo1.c                            Mike Melanson
   nuv.c                                 Reimar Doeffinger
   nvdec*, nvenc*                        Timo Rothenpieler
-  oh*                                   Zhao Zhili
+  omx.c                                 Martin Storsjo, Aman Gupta
   opus*                                 Rostislav Pehlivanov
   pcx.c                                 Ivo van Poorten
   pgssubdec.c                           Reimar Doeffinger
@@ -244,7 +243,6 @@ Codecs:
   qtrle.c                               Mike Melanson
   ra144.c, ra144.h, ra288.c, ra288.h    Roberto Togni
   resample2.c                           Michael Niedermayer
-  rkmppenc*                             Zhao Zhili
   rl2.c                                 Sascha Sommer
   rpza.c                                Roberto Togni
   rtjpeg.c, rtjpeg.h                    Reimar Doeffinger
@@ -252,6 +250,7 @@ Codecs:
   sanm.c                                Manuel Lauss
   smc.c                                 Mike Melanson
   snow*                                 Michael Niedermayer, Loren Merritt
+  sonic.c                               Alex Beregszaszi
   speedhq.c                             Steinar H. Gunderson
   srt*                                  Aurelien Jacobs
   sunrast.c                             Ivo van Poorten
@@ -318,10 +317,6 @@ libavfilter
 ===========
 
 Generic parts:
-
-  Framework and orphaned filters        Nicolas George
-  (except hardware acceleration)
-
   graphdump.c                           Nicolas George
 
   motion_estimation.c                   Davinder Singh
@@ -353,9 +348,7 @@ Filters:
   vf_minterpolate.c                     Davinder Singh
   vf_readvitc.c                         Tobias Rapp (CC t.rapp at noa-archive dot com)
   vf_scale.c                        [2] Michael Niedermayer
-  vf_scale_vt.c                         Zhao Zhili
   vf_tonemap_opencl.c                   Ruiling Song
-  vf_transpose_vt.c                     Zhao Zhili
   vf_yadif.c                        [2] Michael Niedermayer
   vf_xfade_vulkan.c                 [2] Marvin Scholz (CC <epirat07@gmail.com>)
 
@@ -413,9 +406,7 @@ Muxers/Demuxers:
   flvenc.c                              Michael Niedermayer, Steven Liu
   gxf.c                                 Reimar Doeffinger
   gxfenc.c                              Baptiste Coudurier
-  hls.c                                 Kacper Michajłow
   hlsenc.c                              Christian Suloway, Steven Liu
-  hxvs.c                                Zhao Zhili
   iamf*                             [2] James Almer
   idcin.c                               Mike Melanson
   idroqdec.c                            Mike Melanson
@@ -451,9 +442,9 @@ Muxers/Demuxers:
   nsvdec.c                              Francois Revol
   nut*                                  Michael Niedermayer
   nuv.c                                 Reimar Doeffinger
-  oggdec.c, oggdec.h                    David Conrad, Romain Beauxis
-  oggenc.c                              Baptiste Coudurier, Romain Beauxis
-  oggparse*.c                           David Conrad, Romain Beauxis
+  oggdec.c, oggdec.h                    David Conrad
+  oggenc.c                              Baptiste Coudurier
+  oggparse*.c                           David Conrad
   oma.c                                 Maxim Poliakovski
   pp_bnk.c                              Zane van Iperen
   psxstr.c                              Mike Melanson

Makefile

@@ -106,7 +106,7 @@ ffbuild/.config: $(CONFIGURABLE_COMPONENTS)
 	@-printf '\nWARNING: $(?) newer than config_components.h, rerun configure\n\n'
 	@-tput sgr0 2>/dev/null
 
-SUBDIR_VARS := CLEANFILES FFLIBS DEVPROGS HOSTPROGS TESTPROGS TOOLS      \
+SUBDIR_VARS := CLEANFILES FFLIBS HOSTPROGS TESTPROGS TOOLS               \
                HEADERS ARCH_HEADERS BUILT_HEADERS SKIPHEADERS            \
                ARMV5TE-OBJS ARMV6-OBJS ARMV8-OBJS VFP-OBJS NEON-OBJS     \
                ALTIVEC-OBJS VSX-OBJS X86ASM-OBJS                         \
@@ -143,10 +143,8 @@ ifeq ($(STRIPTYPE),direct)
 else
 	$(RM) $@
 	$(CP) $< $@
-ifneq ($(STRIPTYPE),nostrip)
 	$(STRIP) $@
 endif
-endif
 
 %$(PROGSSUF)_g$(EXESUF): $(FF_DEP_LIBS)
 	$(call LINK,$(LDFLAGS) $(LDEXEFLAGS) $(LD_O) $(OBJS-$*) $(FF_EXTRALIBS))
@@ -189,8 +187,7 @@ clean::
 	$(RM) -rf coverage.info coverage.info.in lcov
 
 distclean:: clean
-	$(RM) .version config.asm config.h config_components.* mapfile \
-		checkasm_config_generated.* checkasm_header_config_generated.h \
+	$(RM) .version config.asm config.h config_components.* mapfile  \
 		ffbuild/.config ffbuild/config.* libavutil/avconfig.h \
 		version.h libavutil/ffversion.h libavcodec/codec_names.h \
 		libavcodec/bsf_list.c libavformat/protocol_list.c \
@@ -205,8 +202,8 @@ endif
 config:
 	$(SRC_PATH)/configure $(value FFMPEG_CONFIGURATION)
 
-build: all alltools devprogs examples testprogs
-check: all alltools devprogs examples testprogs fate
+build: all alltools examples testprogs
+check: all alltools examples testprogs fate
 
 include $(SRC_PATH)/tests/Makefile
 

README.md

@@ -1,4 +1,5 @@
-# FFmpeg README
+FFmpeg README
+=============
 
 FFmpeg is a collection of libraries and tools to process multimedia content
 such as audio, video, subtitles and related metadata.

RELEASE

@@ -1 +1 @@
-8.0.git
+8.1.2

RELEASE_NOTES

@@ -0,0 +1,15 @@
+
+              ┌──────────────────────────────────────┐
+              │ RELEASE NOTES for FFmpeg 8.1 "Hoare" │
+              └──────────────────────────────────────┘
+
+   The FFmpeg Project proudly presents FFmpeg 8.1 "Hoare", about 7
+   months after the release of FFmpeg 8.0.
+
+   A complete Changelog is available at the root of the project, and the
+   complete Git history on https://git.ffmpeg.org/gitweb/ffmpeg.git
+
+   We hope you will like this release as much as we enjoyed working on it, and
+   as usual, if you have any questions about it, or any FFmpeg related topic,
+   feel free to join us on the #ffmpeg IRC channel (on irc.libera.chat) or ask
+   on the mailing-lists.

compat/cuda/cuda_runtime.h

@@ -182,7 +182,6 @@ static inline __device__ float fabsf(float a) { return __builtin_fabsf(a); }
 static inline __device__ float fabs(float a) { return __builtin_fabsf(a); }
 static inline __device__ double fabs(double a) { return __builtin_fabs(a); }
 static inline __device__ float sqrtf(float a) { return __builtin_sqrtf(a); }
-static inline __device__ float rintf(float a) { return __builtin_rintf(a); }
 
 static inline __device__ float __saturatef(float a) { return __nvvm_saturate_f(a); }
 static inline __device__ float __sinf(float a) { return __nvvm_sin_approx_f(a); }

compat/windows/makedef

@@ -33,11 +33,6 @@ if [ ! -f "$vscript" ]; then
 fi
 
 for object in "$@"; do
-    case "$object" in
-    @*)
-        object="${object#@}"
-        ;;
-    esac
     if [ ! -f "$object" ]; then
         echo "Object does not exist: ${object}" >&2
         exit 1

doc/APIchanges

@@ -1,77 +1,11 @@
-The last version increases of all libraries were on 2026-06-23.
+The last version increases of all libraries were on 2025-03-28
 
 API changes, most recent first:
 
-2026-06-23 - 61693f6c35f - lavu 60.34.100 - hwcontext_vulkan.h
-  Switch AVVkFrame.access to VkAccessFlagBits2.
-
-2026-06-23 - 191be14fb6d - lavc 62.37.100 - packet.h
-  av_packet_pack_dictionary() now accepts const AVDictionary*.
-
-2026-06-10 - 12dc67b6fe0 - lavu 60.33.100 - frame.h
-  Add AV_FRAME_DATA_RAW_COLOR_PARAMS.
-
-2026-06-03 - 972c0cf91f4 - lsws 9.8.100 - swscale.h
-  Add enum SwsBackend and SwsContext.backends
-
-2026-06-02 - 9fe5758da5c - lavu 60.32.100 - hwcontext_vulkan.h
-  Add AVVulkanDeviceContext.queue_flags.
-
-2026-05-31 - 1e3883df9f0 - lavf 62.19.100 - avformat.h
-  Add AVStreamGroupLayeredVideo
-  Add AVStreamGroup.params.layered_video
-  Add AV_STREAM_GROUP_PARAMS_DOLBY_VISION
-  Deprecate AVStreamGroupLCEVC
-  Deprecate AVStreamGroup.params.lcevc
-
-2026-05-30 - 5f6dff5e7db - lavc 62.35.100 - packet.h
-  Add AV_PKT_DATA_HEVC_CONF side data type.
-
-2026-05-14 - xxxxxxxxxxx - lavf 62.18.100 - avformat.h
-  Add AV_FDEBUG_ID3V2 debug flag, deprecate FF_FDEBUG_TS in favor of
-  AV_FDEBUG_TS.
-
-2026-05-17 - a5822fca94f - lavf 62.17.100 - avformat.h
-  Add AV_STREAM_GROUP_PARAMS_TREF.
-  Add AVStreamGroupTREF.
-
-2026-05-16 - c0bdc3b62a5 - lavf 62.16.100 - avformat.h
-  Add AVFMT_FIXED_FRAMESIZE.
-
-2026-05-16 - b1120b1ed80 - lavc 62.33.100 - avcodec.h
-  Add AV_CODEC_FLAG2_FIXED_FRAME_SIZE.
-
-2026-05-12 - 99908c6e05e - lavu 60.31.100 - frame.h
-Add IAMF frame side data types to enum AVFrameSideDataType:
-- AV_FRAME_DATA_IAMF_MIX_GAIN_PARAM
-- AV_FRAME_DATA_IAMF_DEMIXING_INFO_PARAM
-- AV_FRAME_DATA_IAMF_RECON_GAIN_INFO_PARAM
-
-2026-05-05 - 5c557dd5d55 - lavf 62.15.100 - avformat.h
-  Add av_program_copy().
-
-2026-05-05 - 7623379a770 - lavf 62.14.100 - avformat.h
-  Add av_program_add_stream_index2().
-
-2026-04-14 - 7faa6ee2aa - lavc 62.30.100 - packet.h
-  Add AV_PKT_DATA_DYNAMIC_HDR_SMPTE_2094_APP5 side data type.
-
-2026-04-09 - 6ba6db4f19 - lavu 60.30.100 - hdr_dynamic_metadata.h frame.h
-  Add AVDynamicHDRSmpte2094App5 struct and functions.
-  Add AV_FRAME_DATA_DYNAMIC_HDR_SMPTE_2094_APP5 side data type.
-
-2026-03-14 - c102e89448a - lavu 60.29.100 - hwcontext_vulkan.h
+2026-03-14 - xxxxxxxxxx - lavu 60.26.100 - hwcontext_vulkan.h
   Deprecate AVVulkanDeviceContext.lock_queue and
   AVVulkanDeviceContext.unlock_queue without replacement.
 
-2026-03-12 - 3503b19711c - lsws 9.7.100 - swscale.h
-  Add enum SwsScaler, and SwsContext.scaler/scaler_sub.
-
-2026-03-11 - 910000fe59d - lavu 60.28.100 - hwcontext_amf.h
-  Add av_amf_display_mastering_meta_to_hdrmeta(), av_amf_light_metadata_to_hdrmeta().
-  Add av_amf_extract_hdr_metadata(), av_amf_attach_hdr_metadata().
-  Add av_amf_get_color_profile().
-
 2026-03-07 - c23d56b173a - lavc 62.26.100 - codec_desc.h
   Add AV_CODEC_PROP_ENHANCEMENT.
 

doc/CVSS.txt

@@ -1,40 +0,0 @@
-This document is work in progress
-
-*What is CVSS*
-    The Common Vulnerability Scoring System (CVSS) is an open, industry-standard framework used to measure and communicate the severity of software vulnerabilities, ranging from 0.0 to 10.0.
-
-*Why we need this Document*
-    It is important that FFmpeg CVEs have consistent and correct CVSS, not only for the obvious reason that one can recognize the severity of an issue at first glance.
-    But also as these numbers form the basis of rewards paid in bug bounty systems. Inconsistent CVSS could lead to unfair payouts.
-
-*What is this Document*
-    Prior 2026, FFmpeg had no guideline about CVSS.
-    This document describes how to select the CVSS for a FFmpeg related CVE. It currently only covers the Base Score.
-
-*What is the CVSS Base Score*
-    AV Attack Vector    (Network, Adjacent, Local, Physical)
-    AC Attack Complexity (Low, High)
-    PR Privileges Required (None, Low, High)
-    UI User Interaction (None, Required)
-    S  Scope (Unchanged, Changed)
-    C  Confidentiality (None, Low, High)
-    I  Integrity (None, Low, High)
-    A  Availability (None, Low, High)
-
-
-*Things people have set incorrectly*
-
-Below are general guidelines and in specific cases other things may apply.
-
-Attack Vector.
-    Quote from https://www.first.org/cvss/v3.1/user-guide
-        "Specifically, analysts should only score for Network or Adjacent when a vulnerability is bound to the network stack.
-         Vulnerabilities which require user interaction to download or receive malicious content (which could also be delivered locally, e.g., via USB drives) should be scored as Local."
-
-Availability.
-    FFmpeg Crashes -> AVAILABILITY IMPACT: Low
-    FFmpeg is frequently used as a short-lived, single-run process instead of a continuously running service that handles ongoing streams of user input. In that usage model, a crash usually causes only limited disruption.
-
-User Interaction
-    Please consider if an attacker can actually set the parameters required for an attack.
-    In general arbitrary filter parameters cannot be set by an attacker and require the user/account owner/admin to set them

doc/Doxyfile

@@ -38,7 +38,7 @@ PROJECT_NAME           = FFmpeg
 # could be handy for archiving the generated documentation or if some version
 # control system is used.
 
-PROJECT_NUMBER         =
+PROJECT_NUMBER         = 8.1.2
 
 # Using the PROJECT_BRIEF tag one can provide an optional one line description
 # for a project that appears at the top of each page and should give viewer a

doc/Makefile

@@ -54,7 +54,7 @@ TEXIDEP = perl $(SRC_PATH)/doc/texidep.pl $(SRC_PATH) $< $@ >$(@:%=%.d)
 doc/%.txt: TAG = TXT
 doc/%.txt: doc/%.texi
 	$(Q)$(TEXIDEP)
-	$(M)$(MAKEINFO) --force --no-headers -o $@ $< 2>/dev/null
+	$(M)makeinfo --force --no-headers -o $@ $< 2>/dev/null
 
 GENTEXI  = format codec
 GENTEXI := $(GENTEXI:%=doc/avoptions_%.texi)
@@ -69,11 +69,11 @@ doc/%-all.html: TAG = HTML
 ifdef HAVE_MAKEINFO_HTML
 doc/%.html: doc/%.texi $(SRC_PATH)/doc/t2h.pm $(GENTEXI)
 	$(Q)$(TEXIDEP)
-	$(M)$(MAKEINFO) --html -I doc --no-split -D config-not-all --init-file=$(SRC_PATH)/doc/t2h.pm --output $@ $<
+	$(M)makeinfo --html -I doc --no-split -D config-not-all --init-file=$(SRC_PATH)/doc/t2h.pm --output $@ $<
 
 doc/%-all.html: doc/%.texi $(SRC_PATH)/doc/t2h.pm $(GENTEXI)
 	$(Q)$(TEXIDEP)
-	$(M)$(MAKEINFO) --html -I doc --no-split -D config-all --init-file=$(SRC_PATH)/doc/t2h.pm --output $@ $<
+	$(M)makeinfo --html -I doc --no-split -D config-all --init-file=$(SRC_PATH)/doc/t2h.pm --output $@ $<
 else
 doc/%.html: doc/%.texi $(SRC_PATH)/doc/t2h.init $(GENTEXI)
 	$(Q)$(TEXIDEP)

doc/bitstream_filters.texi

@@ -124,33 +124,6 @@ that this level currently behaves the same as @samp{limited} in libavcodec.
 @end table
 @end table
 
-@section dovi_split
-
-Split a Dolby Vision Profile 7 multi-layer HEVC bitstream. Profile 7 carries the
-enhancement-layer HEVC bitstream interleaved inside the base-layer access units,
-wrapped in user-unspecified NAL units of type 63 (UNSPEC63), and the RPU metadata
-as a sibling user-unspecified NAL of type 62 (UNSPEC62).
-
-@table @option
-@item mode
-Which Dolby Vision components to keep in the output bitstream.
-@table @samp
-@item bl
-Base layer only: drop every UNSPEC63 (EL) and every UNSPEC62 (RPU).
-The output is a plain HEVC stream with no Dolby Vision markers. This is the
-default.
-@item bl_rpu
-Base layer with the RPU NAL kept.
-@item el
-Enhancement layer only: for every UNSPEC63 NAL, strip the two-byte outer NAL
-header and emit the inner payload. The result is a standalone HEVC bitstream.
-UNSPEC62 (RPU) is dropped.
-@item el_rpu
-Enhancement layer with the RPU NAL kept verbatim. Same as @samp{el}, but the
-UNSPEC62 RPU NALs are also emitted alongside the unwrapped EL NALs.
-@end table
-@end table
-
 @section dump_extra
 
 Add extradata to the beginning of the filtered packets except when

doc/build_system.txt

@@ -7,6 +7,11 @@ V
     Disable the default terse mode, the full command issued by make and its
     output will be shown on the screen.
 
+DBG
+    Preprocess x86 external assembler files to a .dbg.asm file in the object
+    directory, which then gets compiled. Helps in developing those assembler
+    files.
+
 DESTDIR
     Destination directory for the install targets, useful to prepare packages
     or install FFmpeg in cross-environments.

doc/codecs.texi

@@ -646,8 +646,6 @@ Do not skip samples and export skip information as frame side data.
 Do not reset ASS ReadOrder field on flush.
 @item icc_profiles
 Generate/parse embedded ICC profiles from/to colorimetry tags.
-@item fixed_frame_size
-Force audio encoders to use a fixed frame size.
 @end table
 
 @item export_side_data @var{flags} (@emph{decoding/encoding,audio,video,subtitles})

doc/decoders.texi

@@ -307,6 +307,14 @@ This decoder generates wave patterns according to predefined sequences. Its
 use is purely internal and the format of the data it accepts is not publicly
 documented.
 
+@section libcelt
+
+libcelt decoder wrapper.
+
+libcelt allows libavcodec to decode the Xiph CELT ultra-low delay audio codec.
+Requires the presence of the libcelt headers and library during configuration.
+You need to explicitly configure the build with @code{--enable-libcelt}.
+
 @section libgsm
 
 libgsm decoder wrapper.

doc/demuxers.texi

@@ -1181,51 +1181,10 @@ RIFF Wave Audio demuxer.
 
 This demuxer accepts the following options:
 @table @option
-@item ignore_length @var{bool}
-Ignore the size of the @code{data} chunk and keep reading until the end of the file if set.
-May be useful to read broken or partial files where the header was not properly updated,
-but will misinterpret files with non-audio chunks after the @code{data} chunk.
-Default is disabled.
-
 @item max_size
 Specify the maximum packet size in bytes for the demuxed packets. By default
 this is set to 0, which means that a sensible value is chosen based on the
 input format.
 @end table
 
-@section webp
-
-Animated WebP demuxer.
-
-It accepts the following options:
-
-@table @option
-@item -min_delay @var{int}
-Set the minimum valid delay between frames in milliseconds.
-Range is 0 to 60000. Default value is 10.
-
-@item -max_webp_delay @var{int}
-Set the maximum valid delay between frames in milliseconds.
-Range is 0 to 16777215. Default value is 16777215 (over four hours),
-the maximum value allowed by the specification.
-
-@item -default_delay @var{int}
-Set the default delay between frames in milliseconds.
-Range is 0 to 60000. Default value is 100.
-
-@item -ignore_loop @var{bool}
-WebP files can contain information to loop a certain number of times
-(or infinitely). If @option{ignore_loop} is set to true, then the loop
-setting from the input will be ignored and looping will not occur.
-If set to false, then looping will occur and will cycle the number
-of times according to the WebP. Default value is true.
-
-@item usebgcolor @var{bool}
-WebP files contain a background color hint in the ANIM chunk, but the
-WebP specification says that viewer applications are not required to
-use it. If @option{usebgcolor} is set to true, then the background
-color hint will be used, otherwise transparent black will be used for
-the background. Default value is false.
-@end table
-
 @c man end DEMUXERS

doc/developer.texi

@@ -225,25 +225,6 @@ AVStream *stream;
 AVStream* stream;
 @end example
 
-@item
-When sensible, prefer a narrow variable scope, especially in for loops:
-
-@example c, good
-// Good
-for (unsigned i = 0; i < submix->nb_elements; i++) @{
-    // Do something...
-@}
-@end example
-
-@example c, bad
-// Bad style
-unsigned i;
-//...
-for (i = 0; i < submix->nb_elements; i++) @{
-    // Do something...
-@}
-@end example
-
 @end itemize
 
 If you work on a file that does not follow these guidelines consistently,
@@ -387,7 +368,7 @@ symbols. If in doubt, just avoid names starting with @code{_} altogether.
 Casts should be used only when necessary. Unneeded parentheses
 should also be avoided if they don't make the code easier to understand.
 @item
-Where applicable, SI units shall be used. For example timeouts should use seconds as the fundamental unit not microseconds.
+Where applicable, SI units shall be used. For example timeouts should use seconds as the fundamental unit not micro seconds.
 That means a bare value like @samp{1.0} must mean 1 second, @samp{50m} means 50 milliseconds. For weight, gram shall be used.
 @end itemize
 
@@ -488,25 +469,15 @@ later on.
 Also if you have doubts about splitting or not splitting, do not hesitate to
 ask/discuss it on the developer mailing list.
 
-@subheading Bug fixes intended for backporting should stay focused.
-Keep bug fixes as small as possible while fully fixing the reported issue.
-Changes not needed for correctness or robustness should be submitted separately,
-especially if they increase complexity or affect backporting.
-
 @subheading Cosmetic changes should be kept in separate patches.
 We refuse source indentation and other cosmetic changes if they are mixed
-with functional changes, such commits will be rejected and removed. However,
-indentation changes that can be ignored by @code{git diff --ignore-all-space}
-(e.g. changes in whitespace amount, leading/trailing spaces) may be mixed with
-functional changes, since reviewers can use @code{git diff -w} or
-@code{git log -p --ignore-all-space} to review only the functional parts of
-the change. Forgejo's pull request interface also provides a
-``Hide whitespace changes'' option for this purpose. Every
+with functional changes, such commits will be rejected and removed. Every
 developer has his own indentation style, you should not change it. Of course
 if you (re)write something, you can use your own style, even though we would
 prefer if the indentation throughout FFmpeg was consistent (Many projects
 force a given indentation style - we do not.). If you really need to make
-non-whitespace cosmetic changes, separate them strictly from real changes.
+indentation changes (try to avoid this), separate them strictly from real
+changes.
 
 NOTE: If you had to put if()@{ .. @} over a large (> 5 lines) chunk of code,
 then either do NOT change the indentation of the inner part within (do not

doc/encoders.texi

@@ -1045,10 +1045,6 @@ Other values include 0 for mono and stereo, 1 for surround sound with masking
 and LFE bandwidth optimizations, and 255 for independent streams with an
 unspecified channel layout.
 
-@item dtx (N.A.)
-Allow discontinuous transmission when set to 1. The default value is 0
-(disabled).
-
 @item apply_phase_inv (N.A.) (requires libopus >= 1.2)
 If set to 0, disables the use of phase inversion for intensity stereo,
 improving the quality of mono downmixes, but slightly reducing normal stereo

doc/examples/decode_video.c

@@ -42,11 +42,7 @@ static void pgm_save(unsigned char *buf, int wrap, int xsize, int ysize,
     FILE *f;
     int i;
 
-    f = fopen(filename, "wb");
-    if (!f) {
-        fprintf(stderr, "Could not open %s\n", filename);
-        return;
-    }
+    f = fopen(filename,"wb");
     fprintf(f, "P5\n%d %d\n%d\n", xsize, ysize, 255);
     for (i = 0; i < ysize; i++)
         fwrite(buf + i * wrap, 1, xsize, f);

doc/examples/demux_decode.c

@@ -336,17 +336,15 @@ int main (int argc, char **argv)
 
     if (video_stream) {
         printf("Play the output video file with the command:\n"
-               "ffplay -f rawvideo -pixel_format %s -video_size %dx%d %s\n",
+               "ffplay -f rawvideo -pix_fmt %s -video_size %dx%d %s\n",
                av_get_pix_fmt_name(pix_fmt), width, height,
                video_dst_filename);
     }
 
     if (audio_stream) {
         enum AVSampleFormat sfmt = audio_dec_ctx->sample_fmt;
-        AVChannelLayout mono = AV_CHANNEL_LAYOUT_MONO;
-        AVChannelLayout *ch_layout = &audio_dec_ctx->ch_layout;
+        int n_channels = audio_dec_ctx->ch_layout.nb_channels;
         const char *fmt;
-        char buf[64];
 
         if (av_sample_fmt_is_planar(sfmt)) {
             const char *packed = av_get_sample_fmt_name(sfmt);
@@ -354,18 +352,15 @@ int main (int argc, char **argv)
                    "(%s). This example will output the first channel only.\n",
                    packed ? packed : "?");
             sfmt = av_get_packed_sample_fmt(sfmt);
-            ch_layout = &mono;
+            n_channels = 1;
         }
 
         if ((ret = get_format_from_sample_fmt(&fmt, sfmt)) < 0)
             goto end;
 
-        if ((ret = av_channel_layout_describe(ch_layout, buf, sizeof(buf))) < 0)
-            goto end;
-
         printf("Play the output audio file with the command:\n"
-               "ffplay -f %s -ch_layout %s -sample_rate %d %s\n",
-               fmt, buf, audio_dec_ctx->sample_rate,
+               "ffplay -f %s -ac %d -ar %d %s\n",
+               fmt, n_channels, audio_dec_ctx->sample_rate,
                audio_dst_filename);
     }
 

doc/examples/encode_audio.c

@@ -41,17 +41,7 @@
 /* check that a given sample format is supported by the encoder */
 static int check_sample_fmt(const AVCodec *codec, enum AVSampleFormat sample_fmt)
 {
-    const void *out_config;
-    int ret = avcodec_get_supported_config(NULL, codec, AV_CODEC_CONFIG_SAMPLE_FORMAT,
-                                           0, &out_config, NULL);
-    if (ret < 0) {
-        fprintf(stderr, "Error getting supported sample formats\n");
-        exit(1);
-    }
-    const enum AVSampleFormat *p = out_config;
-
-    if (!p)
-        return 1;
+    const enum AVSampleFormat *p = codec->sample_fmts;
 
     while (*p != AV_SAMPLE_FMT_NONE) {
         if (*p == sample_fmt)
@@ -64,19 +54,13 @@ static int check_sample_fmt(const AVCodec *codec, enum AVSampleFormat sample_fmt
 /* just pick the highest supported samplerate */
 static int select_sample_rate(const AVCodec *codec)
 {
-    const void *out_config;
-    int ret = avcodec_get_supported_config(NULL, codec, AV_CODEC_CONFIG_SAMPLE_RATE,
-                                           0, &out_config, NULL);
-    if (ret < 0) {
-        fprintf(stderr, "Error getting supported sample rates\n");
-        exit(1);
-    }
-    const int *p = out_config;
+    const int *p;
     int best_samplerate = 0;
 
-    if (!p)
+    if (!codec->supported_samplerates)
         return 44100;
 
+    p = codec->supported_samplerates;
     while (*p) {
         if (!best_samplerate || abs(44100 - *p) < abs(44100 - best_samplerate))
             best_samplerate = *p;
@@ -88,19 +72,13 @@ static int select_sample_rate(const AVCodec *codec)
 /* select layout with the highest channel count */
 static int select_channel_layout(const AVCodec *codec, AVChannelLayout *dst)
 {
-    const void *out_config;
-    int ret = avcodec_get_supported_config(NULL, codec, AV_CODEC_CONFIG_CHANNEL_LAYOUT,
-                                           0, &out_config, NULL);
-    if (ret < 0) {
-        fprintf(stderr, "Error getting supported channel layouts\n");
-        exit(1);
-    }
-    const AVChannelLayout *p = out_config, *best_ch_layout;
+    const AVChannelLayout *p, *best_ch_layout;
     int best_nb_channels   = 0;
 
-    if (!p)
+    if (!codec->ch_layouts)
         return av_channel_layout_copy(dst, &(AVChannelLayout)AV_CHANNEL_LAYOUT_STEREO);
 
+    p = codec->ch_layouts;
     while (p->nb_channels) {
         int nb_channels = p->nb_channels;
 
@@ -240,10 +218,10 @@ int main(int argc, char **argv)
         samples = (uint16_t*)frame->data[0];
 
         for (j = 0; j < c->frame_size; j++) {
-            samples[c->ch_layout.nb_channels*j] = (int)(sin(t) * 10000);
+            samples[2*j] = (int)(sin(t) * 10000);
 
             for (k = 1; k < c->ch_layout.nb_channels; k++)
-                samples[c->ch_layout.nb_channels*j + k] = samples[c->ch_layout.nb_channels*j];
+                samples[2*j + k] = samples[2*j];
             t += tincr;
         }
         encode(c, frame, pkt, f);

doc/examples/hw_decode.c

@@ -132,9 +132,8 @@ static int decode_write(AVCodecContext *avctx, AVPacket *packet)
             goto fail;
         }
 
-        if (fwrite(buffer, 1, size, output_file) != size) {
+        if ((ret = fwrite(buffer, 1, size, output_file)) < 0) {
             fprintf(stderr, "Failed to dump raw data.\n");
-            ret = -1;
             goto fail;
         }
 
@@ -233,10 +232,6 @@ int main(int argc, char *argv[])
 
     /* open the file to dump raw data */
     output_file = fopen(argv[3], "w+b");
-    if (!output_file) {
-        fprintf(stderr, "Cannot open output file '%s'\n", argv[3]);
-        return -1;
-    }
 
     /* actual decoding and dump the raw data */
     while (ret >= 0) {

doc/examples/mux.c

@@ -127,7 +127,7 @@ static void add_stream(OutputStream *ost, AVFormatContext *oc,
                        enum AVCodecID codec_id)
 {
     AVCodecContext *c;
-    int ret;
+    int i;
 
     /* find the encoder */
     *codec = avcodec_find_encoder(codec_id);
@@ -157,37 +157,21 @@ static void add_stream(OutputStream *ost, AVFormatContext *oc,
     ost->enc = c;
 
     switch ((*codec)->type) {
-    case AVMEDIA_TYPE_AUDIO: {
-        const void *codec_config;
+    case AVMEDIA_TYPE_AUDIO:
+        c->sample_fmt  = (*codec)->sample_fmts ?
+            (*codec)->sample_fmts[0] : AV_SAMPLE_FMT_FLTP;
         c->bit_rate    = 64000;
-        ret = avcodec_get_supported_config(c, NULL, AV_CODEC_CONFIG_SAMPLE_FORMAT,
-                                           0, &codec_config, NULL);
-        if (ret < 0) {
-            fprintf(stderr, "Failed to get supported sample formats\n");
-            exit(1);
-        }
-        c->sample_fmt  = codec_config ? *(const enum AVSampleFormat*)codec_config
-                                      : AV_SAMPLE_FMT_FLTP;
-        ret = avcodec_get_supported_config(c, NULL, AV_CODEC_CONFIG_SAMPLE_RATE,
-                                           0, &codec_config, NULL);
-        if (ret < 0) {
-            fprintf(stderr, "Failed to get supported sample rates\n");
-            exit(1);
-        }
-        if (codec_config) {
-            const int *supported_samplerates = codec_config;
-            c->sample_rate = supported_samplerates[0];
-            for (; *supported_samplerates; supported_samplerates++) {
-                if (*supported_samplerates == 44100)
+        c->sample_rate = 44100;
+        if ((*codec)->supported_samplerates) {
+            c->sample_rate = (*codec)->supported_samplerates[0];
+            for (i = 0; (*codec)->supported_samplerates[i]; i++) {
+                if ((*codec)->supported_samplerates[i] == 44100)
                     c->sample_rate = 44100;
             }
-        } else {
-            c->sample_rate = 44100;
         }
         av_channel_layout_copy(&c->ch_layout, &(AVChannelLayout)AV_CHANNEL_LAYOUT_STEREO);
         ost->st->time_base = (AVRational){ 1, c->sample_rate };
         break;
-    }
 
     case AVMEDIA_TYPE_VIDEO:
         c->codec_id = codec_id;

doc/examples/qsv_decode.c

@@ -39,7 +39,6 @@
 #include <libavutil/error.h>
 #include <libavutil/hwcontext.h>
 #include <libavutil/hwcontext_qsv.h>
-#include <libavutil/imgutils.h>
 #include <libavutil/mem.h>
 
 static int get_format(AVCodecContext *avctx, const enum AVPixelFormat *pix_fmts)
@@ -89,16 +88,9 @@ static int decode_packet(AVCodecContext *decoder_ctx,
             goto fail;
         }
 
-        for (i = 0; i < FF_ARRAY_ELEMS(sw_frame->data) && sw_frame->data[i]; i++) {
-            int h = sw_frame->height >> (i > 0);
-            int linesize = av_image_get_linesize(sw_frame->format, sw_frame->width, i);
-            if (linesize < 0) {
-                ret = linesize;
-                goto fail;
-            }
-            for (j = 0; j < h; j++)
-                avio_write(output_ctx, sw_frame->data[i] + j * sw_frame->linesize[i], linesize);
-        }
+        for (i = 0; i < FF_ARRAY_ELEMS(sw_frame->data) && sw_frame->data[i]; i++)
+            for (j = 0; j < (sw_frame->height >> (i > 0)); j++)
+                avio_write(output_ctx, sw_frame->data[i] + j * sw_frame->linesize[i], sw_frame->width);
 
 fail:
         av_frame_unref(sw_frame);

doc/examples/qsv_transcode.c

@@ -430,9 +430,7 @@ int main(int argc, char **argv)
 
 end:
     avformat_close_input(&ifmt_ctx);
-    if (ofmt_ctx && !(ofmt_ctx->oformat->flags & AVFMT_NOFILE))
-        avio_closep(&ofmt_ctx->pb);
-    avformat_free_context(ofmt_ctx);
+    avformat_close_input(&ofmt_ctx);
     avcodec_free_context(&decoder_ctx);
     avcodec_free_context(&encoder_ctx);
     av_buffer_unref(&hw_device_ctx);

doc/examples/remux.c

@@ -184,7 +184,7 @@ end:
     avformat_close_input(&ifmt_ctx);
 
     /* close output */
-    if (ofmt_ctx && !(ofmt_ctx->oformat->flags & AVFMT_NOFILE))
+    if (ofmt_ctx && !(ofmt->flags & AVFMT_NOFILE))
         avio_closep(&ofmt_ctx->pb);
     avformat_free_context(ofmt_ctx);
 

doc/examples/transcode.c

@@ -177,7 +177,7 @@ static int open_output_file(const char *filename)
                 enc_ctx->width = dec_ctx->width;
                 enc_ctx->sample_aspect_ratio = dec_ctx->sample_aspect_ratio;
 
-                ret = avcodec_get_supported_config(enc_ctx, NULL,
+                ret = avcodec_get_supported_config(dec_ctx, NULL,
                                                    AV_CODEC_CONFIG_PIX_FORMAT, 0,
                                                    (const void**)&pix_fmts, NULL);
 
@@ -195,7 +195,7 @@ static int open_output_file(const char *filename)
                 if (ret < 0)
                     return ret;
 
-                ret = avcodec_get_supported_config(enc_ctx, NULL,
+                ret = avcodec_get_supported_config(dec_ctx, NULL,
                                                    AV_CODEC_CONFIG_SAMPLE_FORMAT, 0,
                                                    (const void**)&sample_fmts, NULL);
 

doc/examples/transcode_aac.c

@@ -152,7 +152,6 @@ static int open_output_file(const char *filename,
     AVIOContext *output_io_context = NULL;
     AVStream *stream               = NULL;
     const AVCodec *output_codec    = NULL;
-    const void *sample_fmts;
     int error;
 
     /* Open the output file to write to it. */
@@ -208,12 +207,8 @@ static int open_output_file(const char *filename,
     /* Set the basic encoder parameters.
      * The input file's sample rate is used to avoid a sample rate conversion. */
     av_channel_layout_default(&avctx->ch_layout, OUTPUT_CHANNELS);
-    error = avcodec_get_supported_config(avctx, NULL, AV_CODEC_CONFIG_SAMPLE_FORMAT,
-                                         0, &sample_fmts, NULL);
-    av_assert0(error >= 0);
-    avctx->sample_fmt     = sample_fmts ? *(const enum AVSampleFormat*)sample_fmts
-                                        : input_codec_context->sample_fmt;
     avctx->sample_rate    = input_codec_context->sample_rate;
+    avctx->sample_fmt     = output_codec->sample_fmts[0];
     avctx->bit_rate       = OUTPUT_BIT_RATE;
 
     /* Set the sample rate for the container. */

doc/examples/vaapi_encode.c

@@ -96,6 +96,7 @@ static int encode_write(AVCodecContext *avctx, AVFrame *frame, FILE *fout)
 
 end:
     av_packet_free(&enc_pkt);
+    ret = ((ret == AVERROR(EAGAIN)) ? 0 : -1);
     return ret;
 }
 
@@ -117,7 +118,7 @@ int main(int argc, char *argv[])
     height = atoi(argv[2]);
     size   = width * height;
 
-    if (!(fin = fopen(argv[3], "rb"))) {
+    if (!(fin = fopen(argv[3], "r"))) {
         fprintf(stderr, "Fail to open input file : %s\n", strerror(errno));
         return -1;
     }
@@ -197,8 +198,7 @@ int main(int argc, char *argv[])
             goto close;
         }
 
-        err = encode_write(avctx, hw_frame, fout);
-        if (err != AVERROR(EAGAIN) && err < 0) {
+        if ((err = (encode_write(avctx, hw_frame, fout))) < 0) {
             fprintf(stderr, "Failed to encode.\n");
             goto close;
         }

doc/examples/vaapi_transcode.c

@@ -294,9 +294,7 @@ int main(int argc, char **argv)
 
 end:
     avformat_close_input(&ifmt_ctx);
-    if (ofmt_ctx && !(ofmt_ctx->oformat->flags & AVFMT_NOFILE))
-        avio_closep(&ofmt_ctx->pb);
-    avformat_free_context(ofmt_ctx);
+    avformat_close_input(&ofmt_ctx);
     avcodec_free_context(&decoder_ctx);
     avcodec_free_context(&encoder_ctx);
     av_buffer_unref(&hw_device_ctx);

doc/ffmpeg.texi

@@ -1569,27 +1569,6 @@ Set whether on display the image should be vertically flipped.
 
 See the @code{-display_rotation} option for more details.
 
-@item -mastering_display[:@var{stream_specifier}] @var{G(%u,%u)B(%u,%u)R(%u,%u)WP(%u,%u)L(%u,%u)} (@emph{input,per-stream})
-Set video mastering display metadata.
-
-@var{G(%u,%u)B(%u,%u)R(%u,%u)WP(%u,%u)L(%u,%u)} is a string specifying
-X,Y display primaries for GBR channels and white point (WP) in units of
-0.00002, and max-min luminance (L) values in units of 0.0001 candela per
-meter square. The values are unsigned integers representing the numerator
-of a rational with an implicit denominator of 50000 for GBR and (WP), and
-implicit denominator 10000 for (L).
-
-This option overrides the mastering display metadata stored in the file,
-if any.
-
-@item -content_light[:@var{stream_specifier}] @var{%u,%u} (@emph{input,per-stream})
-Set video content light metadata.
-
-@var{%u,%u} is a string specifying max content light level and maximum picture
-average light level.
-
-This option overrides the content light metadata stored in the file, if any.
-
 @item -vn (@emph{input/output})
 As an input option, blocks all video streams of a file from being filtered or
 being automatically selected or mapped for any output. See @code{-discard}
@@ -2388,19 +2367,25 @@ If either the input or output is blocked leading to actual read speed falling be
 specified readrate, then this rate takes effect till the input catches up with the
 specified readrate. Must not be lower than the primary readrate.
 
-@item -fps_mode[:@var{stream_specifier}] @var{parameter} (@emph{output,per-stream})
-Set video sync method / framerate mode.
+@item -vsync @var{parameter} (@emph{global})
+@itemx -fps_mode[:@var{stream_specifier}] @var{parameter} (@emph{output,per-stream})
+Set video sync method / framerate mode. vsync is applied to all output video streams
+but can be overridden for a stream by setting fps_mode. vsync is deprecated and will be
+removed in the future.
+
+For compatibility reasons some of the values for vsync can be specified as numbers (shown
+in parentheses in the following table).
 
 @table @option
-@item passthrough
+@item passthrough (0)
 Each frame is passed with its timestamp from the demuxer to the muxer.
-@item cfr
+@item cfr (1)
 Frames will be duplicated and dropped to achieve exactly the requested
 constant frame rate.
-@item vfr
+@item vfr (2)
 Frames are passed through with their timestamp or dropped so as to
 prevent 2 frames from having the same timestamp.
-@item auto
+@item auto (-1)
 Chooses between cfr and vfr depending on muxer capabilities. This is the
 default method.
 @end table
@@ -2430,7 +2415,7 @@ Do not process input timestamps, but keep their values without trying
 to sanitize them. In particular, do not remove the initial start time
 offset value.
 
-Note that, depending on the @option{fps_mode} option or on specific muxer
+Note that, depending on the @option{vsync} option or on specific muxer
 processing (e.g. in case the format option @option{avoid_negative_ts}
 is enabled) the output timestamps may mismatch with the input
 timestamps even when this option is selected.

doc/ffprobe.xsd

@@ -357,7 +357,6 @@
   <xsd:complexType name="streamGroupComponentType">
     <xsd:sequence>
       <xsd:element name="subcomponents" type="ffprobe:streamGroupSubComponentList" minOccurs="0" maxOccurs="1"/>
-      <xsd:element name="side_data_list" type="ffprobe:packetSideDataListType" minOccurs="0" maxOccurs="1"/>
       <xsd:element name="component_entry" type="ffprobe:streamGroupEntryType" minOccurs="0" maxOccurs="unbounded"/>
     </xsd:sequence>
   </xsd:complexType>

doc/filters.texi

@@ -7743,11 +7743,6 @@ The file path of the downloaded whisper.cpp model (mandatory).
 The language to use for transcription ('auto' for auto-detect).
 Default value: @code{"auto"}
 
-@item translate
-If enabled, translate the transcription from the source language to English. A
-multilingual model is required to enable this option.
-Default value: @code{"false"}
-
 @item queue
 The maximum size that will be queued into the filter before processing the audio
 with whisper. Using a small value the audio stream will be processed more often,
@@ -7780,8 +7775,8 @@ Default value: @code{"text"}
 
 @item max_len
 Maximum segment length in characters. When set to a value greater than 0,
-transcription segments will be split by word to not exceed this length. This is
-useful for generating subtitles with shorter lines.
+transcription segments will be split to not exceed this length. This is useful
+for generating subtitles with shorter lines.
 Default value: @code{"0"}
 
 @item vad_model
@@ -8611,24 +8606,21 @@ Same as the @ref{subtitles} filter, except that it doesn't require libavcodec
 and libavformat to work. On the other hand, it is limited to ASS (Advanced
 Substation Alpha) subtitles files.
 
-This filter accepts @option{filename}/@option{f}, @option{original_size},
-@option{fontsdir}, and @option{alpha} from the @ref{subtitles} filter, plus the
-following option:
+This filter accepts the following option in addition to the common options from
+the @ref{subtitles} filter:
 
 @table @option
 @item shaping
-Set the shaping engine.
+Set the shaping engine
 
 Available values are:
 @table @samp
 @item auto
 The default libass shaping engine, which is the best available.
 @item simple
-Fast, font-agnostic shaper that can do only substitutions.
+Fast, font-agnostic shaper that can do only substitutions
 @item complex
-Slower shaper using OpenType for substitutions and positioning. Required
-for correct rendering of complex scripts such as Arabic, Hebrew, Devanagari
-and Thai. Requires libass to be built with HarfBuzz.
+Slower shaper using OpenType for substitutions and positioning
 @end table
 
 The default is @code{auto}.
@@ -12224,42 +12216,18 @@ and configure FFmpeg with @code{--enable-libtorch
 --extra-cflags=-I/libtorch_root/libtorch/include/torch/csrc/api/include
 --extra-ldflags=-L/libtorch_root/libtorch/lib/}
 
-@item onnx
-ONNX Runtime backend. To enable this backend you need to install the
-ONNX Runtime library (see @url{https://onnxruntime.ai/}) and configure
-FFmpeg with @code{--enable-libonnxruntime}.
-
-The current ONNX Runtime backend expects 4-D input and output tensors
-with NCHW layout and 32-bit floating-point element type (ONNX
-@code{FLOAT}); models with integer or other element types (e.g.
-@code{UINT8}) are not supported and will be rejected at load time.
-Models using NHWC layout or other ranks are not yet
-supported. Only single-input models are supported; the backend binds
-exactly one input tensor when running the model.
-
-The @option{input} and @option{output} options are optional for the
-ONNX Runtime backend; when they are omitted the backend resolves the
-tensor names from the session.
-
-The ONNX Runtime backend runs inference synchronously using a single
-inference request. The shared @option{async} and @option{nireq} options
-therefore have no effect for @code{dnn_backend=onnx}; inference always
-runs synchronously regardless of their values.
-
 @end table
 
 @item model
 Set path to model file specifying network architecture and its parameters.
-Note that different backends use different file formats. TensorFlow, OpenVINO,
-Libtorch, and ONNX Runtime backends can load files for only their respective formats.
+Note that different backends use different file formats. TensorFlow, OpenVINO
+and Libtorch backend can load files for only its format.
 
 @item input
-Set the input name of the dnn network. Required for the TensorFlow backend;
-optional for the ONNX Runtime backend.
+Set the input name of the dnn network.
 
 @item output
-Set the output name of the dnn network. Required for the TensorFlow backend;
-optional for the ONNX Runtime backend.
+Set the output name of the dnn network.
 
 @item backend_configs
 Set the configs to be passed into backend. To use async execution, set async (default: set).
@@ -12268,20 +12236,6 @@ Roll back to sync execution if the backend does not support async.
 For tensorflow backend, you can set its configs with @option{sess_config} options,
 please use tools/python/tf_sess_config.py to get the configs of TensorFlow backend for your system.
 
-@item device
-Set the device to run the model. For the ONNX Runtime backend this selects the
-execution provider: @code{cpu} (default), @code{cuda} (NVIDIA GPU),
-@code{dml} (DirectML, Windows only) or @code{vitisai} (AMD Ryzen AI NPU).
-
-@item device_id
-Set the device index used by GPU execution providers (e.g. @code{cuda} or
-@code{dml}) for the ONNX Runtime backend. Default is 0.
-
-@item threads_per_operation
-ONNX Runtime backend only. Set the number of CPU threads used per ONNX
-Runtime operator when running with @code{device=cpu}. Default is 0 (let
-ONNX Runtime choose automatically). Has no effect for GPU/NPU providers.
-
 @end table
 
 @subsection Examples
@@ -14790,73 +14744,6 @@ This flag is enabled by default.
 @end table
 @end table
 
-@section frc_amf
-
-Double the frame rate, using Frame Rate Converter (FRC) provided by
-AMD Advanced Media Framework library for hardware acceleration.
-
-The filter accepts the following options:
-
-@table @option
-@item engine_type
-Specify the engine used to run shaders.
-@table @samp
-@item dx11
-DirectX 11.
-
-@item dx12
-DirectX 12 (Default value).
-@end table
-
-@item enable
-Boolean value: enable/disable FRC. Dynamic value, can be altered at a runtime
-without re-initializing the filter. (Default value: enabled).
-
-@item fallback_mode
-Fallback behavior in case of low interpolation confidence.
-@table @samp
-@item duplicate
-Duplicate frame.
-
-@item blend
-Blend two frames together (Default value).
-@end table
-
-@item indicator
-Boolean value: show FRC indicator square in the top left corner of the video
-(Default value: disabled).
-
-@item profile
-Level of hierarchical motion search.
-@table @samp
-@item low
-Less levels of hierarchical motion search.
-Only recommended for extremely low resolutions.
-
-@item high
-Recommended for any resolution up to 1440p. (Default value)
-
-@item super
-More levels of hierarchical motion search. Recommended for resolutions 1440p
-or higher.
-@end table
-
-@item mv_search_mode
-Performance mode of the motion search.
-@table @samp
-@item native
-Conduct motion search on the full resolution of source images.
-
-@item performance
-Conduct motion search on the down scaled source images.
-Recommended for APU or low end GPU for better performance.
-@end table
-
-@item use_future_frame
-Boolean value: enable dependency on future frame, improves quality for the cost
-of latency (Default value: enabled).
-@end table
-
 @section framestep
 
 Select one frame every N-th frame.
@@ -23392,23 +23279,6 @@ at least libass release 0.17.0 (or LIBASS_VERSION 0x01600010), @emph{and} libass
 have been built with libunibreak.
 
 The option is enabled by default except for native ASS.
-
-@item shaping
-Set the shaping engine.
-
-Available values are:
-@table @samp
-@item auto
-The default libass shaping engine, which is the best available.
-@item simple
-Fast, font-agnostic shaper that can do only substitutions.
-@item complex
-Slower shaper using OpenType for substitutions and positioning. Required
-for correct rendering of complex scripts such as Arabic, Hebrew, Devanagari
-and Thai. Requires libass to be built with HarfBuzz.
-@end table
-
-The default is @code{auto}.
 @end table
 
 If the first key is not specified, it is assumed that the first value
@@ -23777,9 +23647,9 @@ The value must be between @code{0} and @var{nb_frames - 1}. Default is @code{0}.
 @item
 Produce 8x8 PNG tiles of all keyframes (@option{-skip_frame nokey}) in a movie:
 @example
-ffmpeg -skip_frame nokey -i file.avi -vf 'scale=128:72,tile=8x8' -an -fps_mode passthrough keyframes%03d.png
+ffmpeg -skip_frame nokey -i file.avi -vf 'scale=128:72,tile=8x8' -an -vsync 0 keyframes%03d.png
 @end example
-The @option{-fps_mode passthrough} is necessary to prevent @command{ffmpeg} from
+The @option{-vsync 0} is necessary to prevent @command{ffmpeg} from
 duplicating each output frame to accommodate the originally detected frame
 rate.
 
@@ -25778,22 +25648,6 @@ Work the same as the identical @ref{scale} filter options.
 @item reset_sar
 Works the same as the identical @ref{scale} filter option.
 
-@item in_color_range
-Override input color range.
-
-@item out_color_range
-Specify output color range.
-
-The accepted values for in_trc and out_trc are:
-@table @samp
-@item studio
-Studio (or restricted, or MPEG) color range.
-
-@item full
-Full (or JPEG) color range.
-
-@end table
-
 @anchor{color_profile}
 @item color_profile
 Specify all color properties at once.
@@ -25811,13 +25665,10 @@ BT.2020
 
 @end table
 
-@item in_trc
-Override input transfer characteristics.
-
-@item out_trc
+@item trc
 Specify output transfer characteristics.
 
-The accepted values for in_trc and out_trc are:
+The accepted values are:
 @table @samp
 @item bt709
 BT.709
@@ -25869,13 +25720,10 @@ ARIB_STD_B67
 
 @end table
 
-@item in_primaries
-Override input color primaries.
-
-@item out_primaries
+@item primaries
 Specify output color primaries.
 
-The accepted values for in_primaries and out_primaries are:
+The accepted values are:
 @table @samp
 @item bt709
 BT.709
@@ -25927,13 +25775,6 @@ Upscale to 4K and change color profile to bt2020.
 @example
 vpp_amf=4096:2160:color_profile=bt2020
 @end example
-
-@item
-Override input primaries and input transfer characteristics, change both to bt709.
-
-@example
-vpp_amf=color_profile=bt2020:in_trc=smpte2084:in_primaries=bt2020:out_trc=bt709:out_primaries=bt709
-@end example
 @end itemize
 
 @anchor{vstack}
@@ -27115,19 +26956,7 @@ value.
 @chapter CUDA Video Filters
 @c man begin CUDA Video Filters
 
-Below is a description of the currently available Nvidia CUDA video filters.
-
-Prerequisites:
-@itemize
-@item Install Nvidia CUDA Toolkit
-@end itemize
-
-Note: If FFmpeg detects the Nvidia CUDA Toolkit during configuration, it will enable CUDA filters automatically without requiring any additional flags. If you want to explicitly enable them, use the following options:
-
-@itemize
-@item Configure FFmpeg with @code{--enable-cuda-nvcc --enable-nonfree}.
-@item Configure FFmpeg with @code{--enable-cuda-llvm}. Additional requirement: @code{llvm} lib must be installed.
-@end itemize
+To enable CUDA and/or NPP filters please refer to configuration guidelines for @ref{CUDA} and for @ref{CUDA NPP} filters.
 
 Running CUDA filters requires you to initialize a hardware device and to pass that device to all filters in any filter graph.
 @table @option
@@ -27162,7 +26991,23 @@ Since CUDA filters operate exclusively on GPU memory, frame data must sometimes
 @end itemize
 Note that @ref{hwupload} uploads data to a surface with the same layout as the software frame, so it may be necessary to add a @ref{format} filter immediately before @ref{hwupload} to ensure the input is in the correct format. Similarly, @ref{hwdownload} may not support all output formats, so an additional @ref{format} filter may need to be inserted immediately after @ref{hwdownload} in the filter graph to ensure compatibility.
 
-@section bilateral_cuda
+@anchor{CUDA}
+@section CUDA
+Below is a description of the currently available Nvidia CUDA video filters.
+
+Prerequisites:
+@itemize
+@item Install Nvidia CUDA Toolkit
+@end itemize
+
+Note: If FFmpeg detects the Nvidia CUDA Toolkit during configuration, it will enable CUDA filters automatically without requiring any additional flags. If you want to explicitly enable them, use the following options:
+
+@itemize
+@item Configure FFmpeg with @code{--enable-cuda-nvcc --enable-nonfree}.
+@item Configure FFmpeg with @code{--enable-cuda-llvm}. Additional requirement: @code{llvm} lib must be installed.
+@end itemize
+
+@subsection bilateral_cuda
 CUDA accelerated bilateral filter, an edge preserving filter.
 This filter is mathematically accurate thanks to the use of GPU acceleration.
 For best output quality, use one to one chroma subsampling, i.e. yuv444p format.
@@ -27182,7 +27027,7 @@ Set window size of the bilateral function to determine the number of neighbours
 If the number entered is even, one will be added automatically.
 Allowed range is 1 to 255. Default is 1.
 @end table
-@subsection Examples
+@subsubsection Examples
 
 @itemize
 @item
@@ -27201,7 +27046,7 @@ Apply the bilateral filter on a video.
 
 @end itemize
 
-@section bwdif_cuda
+@subsection bwdif_cuda
 
 Deinterlace the input video using the @ref{bwdif} algorithm, but implemented
 in CUDA so that it can work as part of a GPU accelerated pipeline with nvdec
@@ -27253,13 +27098,13 @@ Only deinterlace frames marked as interlaced.
 The default value is @code{all}.
 @end table
 
-@section chromakey_cuda
+@subsection chromakey_cuda
 CUDA accelerated YUV colorspace color/chroma keying.
 
 This filter works like normal chromakey filter but operates on CUDA frames.
 for more details and parameters see @ref{chromakey}.
 
-@subsection Examples
+@subsubsection Examples
 
 @itemize
 @item
@@ -27295,7 +27140,7 @@ Process two software sources, explicitly uploading the frames:
 
 @end itemize
 
-@section colorspace_cuda
+@subsection colorspace_cuda
 
 CUDA accelerated implementation of the colorspace filter.
 
@@ -27328,7 +27173,7 @@ JPEG (full) range
 @end table
 
 @anchor{overlay_cuda}
-@section overlay_cuda
+@subsection overlay_cuda
 
 Overlay one video on top of another.
 
@@ -27407,13 +27252,13 @@ This filter also supports the @ref{framesync} options.
 
 
 @anchor{pad_cuda}
-@section pad_cuda
+@subsection pad_cuda
 Add paddings to an input video stream using CUDA.
 
 This filter is the CUDA-accelerated version of the @ref{pad} filter. It accepts the same options and expressions and provides the same core functionality.
 For a detailed description of available options, please see the documentation for the @ref{pad} filter.
 
-@subsection Examples
+@subsubsection Examples
 
 @itemize
 @item
@@ -27430,7 +27275,7 @@ ffmpeg -hwaccel cuda -hwaccel_output_format cuda -i input.mp4 -vf "pad_cuda=w=ih
 @end itemize
 
 @anchor{scale_cuda}
-@section scale_cuda
+@subsection scale_cuda
 
 Scale (resize) and convert (pixel format) the input video, using accelerated CUDA kernels.
 Setting the output width and height works in the same way as for the @ref{scale} filter.
@@ -27499,7 +27344,7 @@ Works the same as the identical @ref{scale} filter option.
 
 @end table
 
-@subsection Examples
+@subsubsection Examples
 
 @itemize
 @item
@@ -27523,7 +27368,7 @@ scale_cuda=passthrough=0
 @end example
 @end itemize
 
-@section thumbnail_cuda
+@subsection thumbnail_cuda
 
 Select the most representative frame in a given sequence of consecutive frames using CUDA.
 
@@ -27539,7 +27384,7 @@ the end. Default is @code{100}.
 Since the filter keeps track of the whole frames sequence, a bigger @var{n}
 value will result in a higher memory usage, so a high value is not recommended.
 
-@subsection Example
+@subsubsection Example
 
 @itemize
 
@@ -27551,57 +27396,7 @@ Thumbnails are extracted from every @var{n}=150-frame batch, selecting one per b
 
 @end itemize
 
-@section transpose_cuda
-
-Transpose rows with columns in the input video and optionally flip it.
-For more in depth examples see the @ref{transpose} video filter, which shares mostly the same options.
-
-It accepts the following parameters:
-
-@table @option
-
-@item dir
-Specify the transposition direction.
-
-Can assume the following values:
-@table @samp
-@item cclock_flip
-Rotate by 90 degrees counterclockwise and vertically flip. (default)
-
-@item clock
-Rotate by 90 degrees clockwise.
-
-@item cclock
-Rotate by 90 degrees counterclockwise.
-
-@item clock_flip
-Rotate by 90 degrees clockwise and vertically flip.
-
-@item reversal
-Rotate by 180 degrees.
-
-@item hflip
-Flip horizontally.
-
-@item vflip
-Flip vertically.
-@end table
-
-@item passthrough
-Do not apply the transposition if the input geometry matches the one
-specified by the specified value. It accepts the following values:
-@table @samp
-@item none
-Always apply transposition. (default)
-@item portrait
-Preserve portrait geometry (when @var{height} >= @var{width}).
-@item landscape
-Preserve landscape geometry (when @var{width} >= @var{height}).
-@end table
-
-@end table
-
-@section yadif_cuda
+@subsection yadif_cuda
 
 Deinterlace the input video using the @ref{yadif} algorithm, but implemented
 in CUDA so that it can work as part of a GPU accelerated pipeline with nvdec
@@ -27659,6 +27454,274 @@ Only deinterlace frames marked as interlaced.
 The default value is @code{all}.
 @end table
 
+@anchor{CUDA NPP}
+@section CUDA NPP
+Below is a description of the currently available NVIDIA Performance Primitives (libnpp) video filters.
+
+Prerequisites:
+@itemize
+@item Install Nvidia CUDA Toolkit
+@item Install libnpp
+@end itemize
+
+To enable CUDA NPP filters:
+
+@itemize
+@item Configure FFmpeg with @code{--enable-nonfree --enable-libnpp}.
+@end itemize
+
+
+@anchor{scale_npp}
+@subsection scale_npp
+
+Use the NVIDIA Performance Primitives (libnpp) to perform scaling and/or pixel
+format conversion on CUDA video frames. Setting the output width and height
+works in the same way as for the @var{scale} filter.
+
+The following additional options are accepted:
+@table @option
+@item format
+The pixel format of the output CUDA frames. If set to the string "same" (the
+default), the input format will be kept. Note that automatic format negotiation
+and conversion is not yet supported for hardware frames
+
+@item interp_algo
+The interpolation algorithm used for resizing. One of the following:
+@table @option
+@item nn
+Nearest neighbour.
+
+@item linear
+@item cubic
+@item cubic2p_bspline
+2-parameter cubic (B=1, C=0)
+
+@item cubic2p_catmullrom
+2-parameter cubic (B=0, C=1/2)
+
+@item cubic2p_b05c03
+2-parameter cubic (B=1/2, C=3/10)
+
+@item super
+Supersampling
+
+@item lanczos
+@end table
+
+@item force_original_aspect_ratio
+Enable decreasing or increasing output video width or height if necessary to
+keep the original aspect ratio. Possible values:
+
+@table @samp
+@item disable
+Scale the video as specified and disable this feature.
+
+@item decrease
+The output video dimensions will automatically be decreased if needed.
+
+@item increase
+The output video dimensions will automatically be increased if needed.
+
+@end table
+
+One useful instance of this option is that when you know a specific device's
+maximum allowed resolution, you can use this to limit the output video to
+that, while retaining the aspect ratio. For example, device A allows
+1280x720 playback, and your video is 1920x800. Using this option (set it to
+decrease) and specifying 1280x720 to the command line makes the output
+1280x533.
+
+Please note that this is a different thing than specifying -1 for @option{w}
+or @option{h}, you still need to specify the output resolution for this option
+to work.
+
+@item force_divisible_by
+Ensures that both the output dimensions, width and height, are divisible by the
+given integer when used together with @option{force_original_aspect_ratio}. This
+works similar to using @code{-n} in the @option{w} and @option{h} options.
+
+This option respects the value set for @option{force_original_aspect_ratio},
+increasing or decreasing the resolution accordingly. The video's aspect ratio
+may be slightly modified.
+
+This option can be handy if you need to have a video fit within or exceed
+a defined resolution using @option{force_original_aspect_ratio} but also have
+encoder restrictions on width or height divisibility.
+
+@item reset_sar
+Works the same as the identical @ref{scale} filter option.
+
+@item eval
+Specify when to evaluate @var{width} and @var{height} expression. It accepts the following values:
+
+@table @samp
+@item init
+Only evaluate expressions once during the filter initialization or when a command is processed.
+
+@item frame
+Evaluate expressions for each incoming frame.
+
+@end table
+
+@end table
+
+The values of the @option{w} and @option{h} options are expressions
+containing the following constants:
+
+@table @var
+@item in_w
+@item in_h
+The input width and height
+
+@item iw
+@item ih
+These are the same as @var{in_w} and @var{in_h}.
+
+@item out_w
+@item out_h
+The output (scaled) width and height
+
+@item ow
+@item oh
+These are the same as @var{out_w} and @var{out_h}
+
+@item a
+The same as @var{iw} / @var{ih}
+
+@item sar
+input sample aspect ratio
+
+@item dar
+The input display aspect ratio. Calculated from @code{(iw / ih) * sar}.
+
+@item n
+The (sequential) number of the input frame, starting from 0.
+Only available with @code{eval=frame}.
+
+@item t
+The presentation timestamp of the input frame, expressed as a number of
+seconds. Only available with @code{eval=frame}.
+
+@item pos
+The position (byte offset) of the frame in the input stream, or NaN if
+this information is unavailable and/or meaningless (for example in case of synthetic video).
+Only available with @code{eval=frame}.
+Deprecated, do not use.
+@end table
+
+@subsection scale2ref_npp
+
+Use the NVIDIA Performance Primitives (libnpp) to scale (resize) the input
+video, based on a reference video.
+
+See the @ref{scale_npp} filter for available options, scale2ref_npp supports the same
+but uses the reference video instead of the main input as basis. scale2ref_npp
+also supports the following additional constants for the @option{w} and
+@option{h} options:
+
+@table @var
+@item main_w
+@item main_h
+The main input video's width and height
+
+@item main_a
+The same as @var{main_w} / @var{main_h}
+
+@item main_sar
+The main input video's sample aspect ratio
+
+@item main_dar, mdar
+The main input video's display aspect ratio. Calculated from
+@code{(main_w / main_h) * main_sar}.
+
+@item main_n
+The (sequential) number of the main input frame, starting from 0.
+Only available with @code{eval=frame}.
+
+@item main_t
+The presentation timestamp of the main input frame, expressed as a number of
+seconds. Only available with @code{eval=frame}.
+
+@item main_pos
+The position (byte offset) of the frame in the main input stream, or NaN if
+this information is unavailable and/or meaningless (for example in case of synthetic video).
+Only available with @code{eval=frame}.
+@end table
+
+@subsubsection Examples
+
+@itemize
+@item
+Scale a subtitle stream (b) to match the main video (a) in size before overlaying
+@example
+'scale2ref_npp[b][a];[a][b]overlay_cuda'
+@end example
+
+@item
+Scale a logo to 1/10th the height of a video, while preserving its display aspect ratio.
+@example
+[logo-in][video-in]scale2ref_npp=w=oh*mdar:h=ih/10[logo-out][video-out]
+@end example
+@end itemize
+
+@subsection sharpen_npp
+Use the NVIDIA Performance Primitives (libnpp) to perform image sharpening with
+border control.
+
+The following additional options are accepted:
+@table @option
+
+@item border_type
+Type of sampling to be used ad frame borders. One of the following:
+@table @option
+
+@item replicate
+Replicate pixel values.
+
+@end table
+@end table
+
+@subsection transpose_npp
+
+Transpose rows with columns in the input video and optionally flip it.
+For more in depth examples see the @ref{transpose} video filter, which shares mostly the same options.
+
+It accepts the following parameters:
+
+@table @option
+
+@item dir
+Specify the transposition direction.
+
+Can assume the following values:
+@table @samp
+@item cclock_flip
+Rotate by 90 degrees counterclockwise and vertically flip. (default)
+
+@item clock
+Rotate by 90 degrees clockwise.
+
+@item cclock
+Rotate by 90 degrees counterclockwise.
+
+@item clock_flip
+Rotate by 90 degrees clockwise and vertically flip.
+@end table
+
+@item passthrough
+Do not apply the transposition if the input geometry matches the one
+specified by the specified value. It accepts the following values:
+@table @samp
+@item none
+Always apply transposition. (default)
+@item portrait
+Preserve portrait geometry (when @var{height} >= @var{width}).
+@item landscape
+Preserve landscape geometry (when @var{width} >= @var{height}).
+@end table
+
+@end table
+
 @c man end CUDA Video Filters
 
 @chapter OpenCL Video Filters
@@ -32276,7 +32339,7 @@ select=n=2:e='mod(n, 2)+1' [odd][even]; [odd] pad=h=2*ih [tmp]; [tmp][even] over
 Select useful frames from an ffconcat file which is using inpoints and
 outpoints but where the source files are not intra frame only.
 @example
-ffmpeg -copyts -segment_time_metadata 1 -i input.ffconcat -fps_mode passthrough -vf select=concatdec_select -af aselect=concatdec_select output.avi
+ffmpeg -copyts -vsync 0 -segment_time_metadata 1 -i input.ffconcat -vf select=concatdec_select -af aselect=concatdec_select output.avi
 @end example
 @end itemize
 

doc/formats.texi

@@ -100,9 +100,6 @@ Print specific debug info.
 Possible values:
 @table @samp
 @item ts
-Print timestamp information.
-@item id3v2
-Print ID3v2 frame data when demuxing files containing ID3v2 tags.
 @end table
 
 @item max_delay @var{integer} (@emph{input/output})

doc/general_contents.texi

@@ -205,39 +205,6 @@ FFmpeg can make use of this library, originating in Modplug-XMMS, to read from M
 See @url{https://github.com/Konstanty/libmodplug}. Pass @code{--enable-libmodplug} to configure to
 enable it.
 
-@section ONNX Runtime
-
-FFmpeg can make use of the ONNX Runtime library as a backend for DNN based filters.
-
-Go to @url{https://onnxruntime.ai/} and follow the instructions for installing
-the library. Pre-built packages are available for various platforms.
-
-Configure FFmpeg with ONNX Runtime support:
-
-@example
-./configure \
-    --enable-libonnxruntime \
-    --extra-cflags="-I/path/to/onnxruntime/include" \
-    --extra-ldflags="-L/path/to/onnxruntime/lib"
-@end example
-
-On ELF-based systems (Linux, BSD) you may also want to embed the
-library search path so that the installed @file{ffmpeg} binary finds the
-shared library at run time without setting @env{LD_LIBRARY_PATH}:
-
-@example
-./configure \
-    --enable-libonnxruntime \
-    --extra-cflags="-I/path/to/onnxruntime/include" \
-    --extra-ldflags="-L/path/to/onnxruntime/lib -Wl,-rpath,/path/to/onnxruntime/lib"
-@end example
-
-The @code{-Wl,-rpath} flag is ELF-specific and should be omitted on
-other platforms (Windows, macOS).
-
-For MinGW GCC builds, add the @code{-D_stdcall=__stdcall} flag for compatibility
-with ONNX Runtime headers.
-
 @section OpenCORE, VisualOn, and Fraunhofer libraries
 
 Spun off Google Android sources, OpenCore, VisualOn and Fraunhofer
@@ -1328,6 +1295,8 @@ following image formats are supported:
 @item Bink Audio             @tab     @tab  X
     @tab Used in Bink and Smacker files in many games.
 @item Bonk audio             @tab     @tab  X
+@item CELT                   @tab     @tab  E
+    @tab decoding supported through external library libcelt
 @item codec2                 @tab  E  @tab  E
     @tab en/decoding supported through external library libcodec2
 @item CRI HCA                @tab     @tab X

doc/infra.txt

@@ -52,8 +52,7 @@ Web, mail, and public facing git, also website git
 
 fftrac VM:
 ----------
-trac.ffmpeg.org         Legacy issue tracking
-code.ffmpeg.org         Primary issue tracking
+trac.ffmpeg.org         Issue tracking
 gpg encrypted backups of the trac repositories are created once a day
 and can be downloaded by any of the admins.
 

doc/issue_tracker.txt

@@ -21,10 +21,7 @@ a mail for every change to every issue.
 
 The subscription URL for the ffmpeg-trac list is:
 https://lists.ffmpeg.org/mailman/listinfo/ffmpeg-trac
-The primary issue tracker is:
-https://code.ffmpeg.org/FFmpeg/FFmpeg/issues
-
-Legacy issues may still exist on:
+The URL of the webinterface of the tracker is:
 https://trac.ffmpeg.org
 
 Type:

doc/muxers.texi

@@ -1330,21 +1330,11 @@ element. Imply that the @option{single_file} option is set to
 @var{true}. In the template, @code{$ext$} is replaced with the file
 name extension specific for the segment format.
 
-@item availability_start_time_ms @var{milliseconds}
-Set the MPD @code{availabilityStartTime} attribute as Unix epoch time in
-milliseconds. If set to @code{0}, the wall clock is used when the first
-packet is written. This is useful for live outputs that need a stable
-availability start time across restarts.
-
 @item streaming @var{bool}
 Enable or disable chunk streaming mode of output. In chunk streaming
 mode, each frame will be a @code{moof} fragment which forms a
 chunk. This is disabled by default.
 
-@item suggested_presentation_delay @var{duration}
-Set the MPD @code{suggestedPresentationDelay} attribute. If set to
-@code{0}, the last segment duration is used.
-
 @item target_latency @var{target_latency}
 Set an intended target latency in seconds for serving (fractional
 value can be set). Applicable only when the @option{streaming} and
@@ -1769,7 +1759,7 @@ See also the @ref{framehash} and @ref{md5} muxers.
 Animated GIF muxer.
 
 Note that the GIF format has a very large time base: the delay between two frames can
-therefore not be smaller than one centisecond.
+therefore not be smaller than one centi second.
 
 @subsection Options
 @table @option
@@ -2654,12 +2644,6 @@ writing is completed. Default is disabled.
 @item protocol_opts @var{options_list}
 Set protocol options as a :-separated list of key=value parameters. Values
 containing the @code{:} special character must be escaped.
-
-@item update_filemtime @var{bool}
-If set to 1, set each output file's modification time to the
-@code{creation_time} metadata value plus the frame's PTS offset.
-If @code{creation_time} is missing or unparsable, a warning is
-logged and the option is ignored. Default value is 0.
 @end table
 
 @subsection Examples
@@ -2668,7 +2652,7 @@ logged and the option is ignored. Default value is 0.
 Use @command{ffmpeg} for creating a sequence of files @file{img-001.jpeg},
 @file{img-002.jpeg}, ..., taking one image every second from the input video:
 @example
-ffmpeg -i in.avi -fps_mode cfr -r 1 -f image2 'img-%03d.jpeg'
+ffmpeg -i in.avi -vsync cfr -r 1 -f image2 'img-%03d.jpeg'
 @end example
 
 Note that with @command{ffmpeg}, if the format is not specified with the
@@ -2676,7 +2660,7 @@ Note that with @command{ffmpeg}, if the format is not specified with the
 format, the image2 muxer is automatically selected, so the previous
 command can be written as:
 @example
-ffmpeg -i in.avi -fps_mode cfr -r 1 'img-%03d.jpeg'
+ffmpeg -i in.avi -vsync cfr -r 1 'img-%03d.jpeg'
 @end example
 
 Note also that the pattern must not necessarily contain "%d" or
@@ -2728,58 +2712,6 @@ computer-generated compositions.
 
 This muxer accepts a single audio stream containing PCM data.
 
-@section iterm2
-iTerm2 inline image protocol muxer.
-
-This muxer writes video frames as OSC 1337 inline images for display in
-terminals that support the iTerm2 image protocol. Use @option{-re} to limit
-the output rate to the source framerate; without it, frames are emitted as
-fast as they are encoded, which is usually not desired for live display.
-
-Frames are sent with the multipart form of the protocol, which splits each
-image across several short control sequences. This avoids the per-sequence
-size limit that otherwise discards large frames, and requires iTerm2 3.5 or
-newer.
-
-The output is a self-contained byte stream and can be redirected to a file.
-Replaying the file with @command{cat} displays the images in the terminal.
-
-@subsection Options
-@table @option
-@item display_width @var{size}
-Set the displayed image width. @var{size} can be @samp{auto}, @var{N} terminal
-cells, @var{N}px pixels, or @var{N}% of the terminal width. When unset, the
-terminal derives the width from the image.
-
-@item display_height @var{size}
-Set the displayed image height. @var{size} uses the same syntax as
-@option{display_width}. When unset, the terminal derives the height from the
-image.
-
-@item keep_aspect @var{bool}
-Preserve the input aspect ratio when scaling. Default is enabled.
-
-@item tmux @var{bool}
-Wrap image data in tmux DCS passthrough. This requires a tmux version whose
-passthrough sequence size limit is large enough for image data, with
-passthrough enabled via @command{tmux set -g allow-passthrough on}. Default is
-disabled.
-@end table
-
-@subsection Examples
-
-Display a video in an iTerm2 terminal:
-@example
-ffmpeg -re -i input.mp4 -f iterm2 -
-@end example
-
-Scale the displayed image to 40 terminal cells tall. Inside tmux, enable
-passthrough first with @command{tmux set -g allow-passthrough on}, then add
-@option{tmux}:
-@example
-ffmpeg -re -i input.mp4 -f iterm2 -display_height 40 -tmux 1 -
-@end example
-
 @section ivf
 On2 IVF muxer.
 
@@ -2819,14 +2751,6 @@ song lyrics with an audio file, such as MP3, Vorbis, or MIDI.
 
 This muxer accepts a single @samp{subrip} or @samp{text} subtitles stream.
 
-@subsection Options
-@table @option
-@item precision @var{number}
-Set the number of decimal digits used for the fractional (sub-second)
-part of the timestamps. Range is @code{1} to @code{6}. Default is
-@code{2}, i.e. centiseconds.
-@end table
-
 @subsection Metadata
 The following metadata tags are converted to the format corresponding
 metadata:
@@ -3145,7 +3069,7 @@ Default is @code{0x0001}.
 Set the @samp{original_network_id}. This is unique identifier of a
 network in DVB. Its main use is in the unique identification of a service
 through the path @samp{Original_Network_ID, Transport_Stream_ID}. Default
-is @code{0xff01}.
+is @code{0x0001}.
 
 @item mpegts_service_id @var{integer}
 Set the @samp{service_id}, also known as program in DVB. Default is
@@ -3172,8 +3096,6 @@ MPEG2 Digital HDTV service.
 Advanced Codec Digital SDTV service.
 @item advanced_codec_digital_hdtv
 Advanced Codec Digital HDTV service.
-@item hevc_digital_hdtv
-HEVC Digital Television service.
 @end table
 
 @item mpegts_pmt_start_pid @var{integer}
@@ -3350,19 +3272,6 @@ ogg files can be safely chained.
 
 @end table
 
-@section pdv
-
-Playdate Video muxer.
-
-This muxer writes the Playdate video container used by Panic's Playdate SDK.
-It requires a seekable output and a single PDV video stream.
-
-@table @option
-@item max_frames @var{frames}
-Reserve space for at most @var{frames} video frames in the file header. This
-option is mandatory.
-@end table
-
 @anchor{rcwtenc}
 @section rcwt
 
@@ -3608,13 +3517,6 @@ Wrap around segment index once it reaches @var{limit}.
 @item segment_start_number @var{number}
 Set the sequence number of the first segment. Defaults to @code{0}.
 
-@item segment_wrap_number @var{number}
-Set the number of times the segment index is assumed to have already wrapped
-when generating the indexes reported in the segment list. Together with
-@option{segment_wrap} and @option{segment_start_number} it offsets the reported
-index, which is computed as @code{segment_start_number + segment_wrap *
-segment_wrap_number}. Defaults to @code{0}.
-
 @item strftime @var{1|0}
 Use the @code{strftime} function to define the name of the new
 segments to write. If this is selected, the output segment name must
@@ -3641,25 +3543,6 @@ argument must be a time duration specification, and defaults to 0.
 If enabled, write an empty segment if there are no packets during the period a
 segment would usually span. Otherwise, the segment will be filled with the next
 packet written. Defaults to @code{0}.
-
-@item write_header_trailer @var{bool}
-Write a header to the first segment and a trailer to the last one, instead of
-writing a header and a trailer to every individual segment. Disabling it
-(@code{false}) also forces @option{individual_header_trailer} to @code{false},
-so that no segment is given a header or trailer. Default value is @code{true}.
-
-@item individual_header_trailer @var{bool}
-If enabled, write a complete header and trailer to every segment, making each
-segment an independently usable file. If disabled (@code{false}), only the first
-segment is given a header and only the last one a trailer. It is forced to
-@code{false} when @option{write_header_trailer} is @code{false} or when
-@option{segment_header_filename} is set. Default value is @code{true}.
-
-@item segment_header_filename @var{name}
-Write the stream header to a separate file named @var{name} instead of storing
-it in the segments. Setting it forces @option{write_header_trailer} to
-@code{true} and @option{individual_header_trailer} to @code{false}, so the
-header is written only once, to @var{name}. Unset by default.
 @end table
 
 Make sure to require a closed GOP when encoding and to set the GOP
@@ -3925,76 +3808,6 @@ ffmpeg -i ... -map 0 -flags +global_header -c:v libx264 -c:a aac
 @end example
 @end itemize
 
-@section wav
-
-RIFF Wave Audio muxer.
-
-@subsection Options
-
-@table @option
-
-@item rf64 @var{mode}
-Choose whether to use the RF64 file format instead of RIFF.
-RF64 allows files larger than 4 gigabytes, but RF64 support is not as
-universal as RIFF.
-
-The following modes are recognized:
-@table @samp
-@item auto
-Start writing a standard RIFF file to retain compatibility with non-RF64-aware
-software, but switch to RF64 if the output file grows larger than 4 gigabytes.
-Reserves a small amount of extra space in the header using a @code{JUNK} chunk,
-which should be ignored by all compliant RIFF readers, but some extremely
-simplistic WAV readers may be confused by this.
-
-@item always
-Always use the RF64 format regardless of file size. The output file will only
-be readable by RF64-compatible software and is no longer a standard RIFF WAVE
-file.
-
-@item never
-Always use plain RIFF, never RF64. This mode is the most compatible with legacy
-software, but if the output file grows beyond 4 gigabytes, the 32-bit size field
-in the @code{data} chunk will no longer be able to correctly represent the size,
-and most audio software will not be able to read some or all of the data.
-(The @ref{wav} demuxer's @code{ignore_length} option can be used to recover such
-a file.)
-This is the default mode.
-@end table
-
-@item write_bext @var{bool}
-Add a Broadcast Wave Format @code{BEXT} chunk containing extended audio
-metadata if enabled. Defaults to false.
-
-@item write_peak @var{mode}
-Add a Peak Envelope chunk (EBU Tech 3285 Supplement 3) if enabled.
-Defaults to off.
-
-@table @samp
-@item off
-Do not add the Peak Envelope chunk.
-
-@item on
-Add the Peak Envelope chunk, calculating the peak values based on the
-@code{peak_block_size}, @code{peak_format}, and @code{peak_ppv} options.
-
-@item only
-Like @code{on}, but skip writing the actual audio data (no @code{data}
-chunk will be written to the file; only the Peak Envelope and other metadata
-chunks will be written.)
-@end table
-
-@item peak_block_size @var{int}
-Number of audio samples used to generate each peak frame, up to 65536.
-Default is 256.
-
-@item peak_format @var{format}
-The format of the peak envelope data (1: uint8, 2: uint16). Default is 2 (uint16).
-
-@item peak_ppv @var{int}
-Number of peak points per peak value (1 or 2). Default is 2.
-@end table
-
 @section webm_chunk
 
 WebM Live Chunk Muxer.
@@ -4138,10 +3951,6 @@ This muxer supports the following options:
 Set the timeout in milliseconds for ICE and DTLS handshake.
 Default value is 5000.
 
-@item timeout @var{integer}
-Set timeout in seconds for socket I/O operations. Applicable only for HTTP output.
-Default value is -1.
-
 @item pkt_size @var{integer}
 Set the maximum size, in bytes, of RTP packets that send out.
 Default value is 1200.

doc/optimization.txt

@@ -20,7 +20,7 @@ architecture-specific versions. It is recommended to look at older
 revisions of the interesting files (web frontends for the various FFmpeg
 branches are listed at http://ffmpeg.org/download.html).
 Alternatively, look into the other architecture-specific versions in
-the x86/, ppc/, aarch64/ subdirectories. Even if you don't exactly
+the x86/, ppc/, alpha/ subdirectories. Even if you don't exactly
 comprehend the instructions, it could help understanding the functions
 and how they can be optimized.
 
@@ -191,6 +191,11 @@ __asm__() block.
 Use external asm (nasm) or inline asm (__asm__()), do not use intrinsics.
 The latter requires a good optimizing compiler which gcc is not.
 
+When debugging a x86 external asm compilation issue, if lost in the macro
+expansions, add DBG=1 to your make command-line: the input file will be
+preprocessed, stripped of the debug/empty lines, then compiled, showing the
+actual lines causing issues.
+
 Inline asm vs. external asm
 ---------------------------
 Both inline asm (__asm__("..") in a .c file, handled by a compiler such as gcc)

doc/protocols.texi

@@ -169,7 +169,6 @@ Read angle 2 of playlist 4 from BluRay mounted to /mnt/bluray, start from chapte
 -playlist 4 -angle 2 -chapter 2 bluray:/mnt/bluray
 @end example
 
-@anchor{cache}
 @section cache
 
 Caching wrapper for input stream.
@@ -344,8 +343,7 @@ time, which is valuable for files on slow medium.
 If set to 1, the protocol will retry reading at the end of the file, allowing
 reading files that still are being written. In order for this to terminate,
 you either need to use the rw_timeout option, or use the interrupt callback
-(for API users). Setting this option also ignores the file size reported by
-the file system.
+(for API users).
 
 @item seekable
 Controls if seekability is advertised on the file. 0 means non-seekable, -1
@@ -453,16 +451,15 @@ string describing the libavformat build. ("Lavf/<version>")
 Set the Referer header. Include 'Referer: URL' header in HTTP request.
 
 @item multiple_requests
-Force persistent connections if set to 1, or disable if 0. Default is -1,
-which means auto (implies keep-alive when using -request_size or
--initial_request_size).
+Use persistent connections if set to 1, default is 0.
 
 @item request_size
 Limit the size of requests made. This is useful for some pathological servers
 that throttle unbounded range requests, as well as when expecting to seek
 frequently. Disabled (set to 0) by default.
 
-Note that if enabling this option, it's strongly recommended to also set
+Note that if enabling this option, it's strongly recommended to also enable
+the @option{multiple_requests} option, as well as setting
 @option{short_seek_size} to the same value or higher. Doing so allows FFmpeg
 to reuse a single HTTP connection wherever possible.
 
@@ -1572,67 +1569,6 @@ If set to any value, listen for an incoming connection. Outgoing connection is d
 Set the maximum number of streams. By default no limit is set.
 @end table
 
-@section shared
-
-Thread-safe, persistent, cross-process cache wrapper for input streams. Caches
-the input stream to a file in the specified directory. This is similar to
-@ref{cache}, but with a persistent cache on disk that may be shared between
-multiple processes or demuxers, even concurrently.
-
-The filename of the cache is obtained by hashing the input URL, so there is
-some risk of collision if the same URL is used for different content (e.g.
-as a result of extra POST data being used to select the stream ID). To avoid
-this, make sure to only use this protocol for URLs that uniquely identify
-the content; otherwise you may get a garbled mix of sources.
-
-The accepted options are:
-@table @option
-
-@item cache_dir
-Path to the directory where cache files are stored. This option is required.
-
-@item block_shift
-Shift factor (log2) of the block size used for internal reads/writes. Defaults
-to 15, i.e. 32KB blocks. If this does not match the value that was specified
-when an existing cache file was created, the previously specified value will
-be used instead.
-
-@item read_only
-If true, use the shared cache for reads but don't write any new blocks to it.
-Default is false. Note that even with this option enabled, the cache file will
-be initialized if it does not already exist.
-
-@item cache_verify
-If true, verify any data read from the cache against the underlying input
-stream, and report any mismatches. Note that this will make the cache layer
-effectively useless. This is purely a debug option.
-
-@item cache_timeout
-If set to a nonzero value, specifies the maximum time (in microseconds) to wait
-for data to become available, if another process is currently trying to fetch
-and cache the same block at the same time. If this timeout elapses, it's
-assumed that the other process may have gotten stuck or died in the meantime.
-
-If set to zero, no waiting is done and all processes will immediately race
-to try and fetch the same missing blocks themselves. Defaults to 10000 (10 ms).
-
-@item retry_errors
-If true (the default), transient read errors from the underlying input stream
-are ignored and retried again. If false, any blocks that previously failed
-being read from will be treated as permanently inaccessible.
-
-@item retry_corrupt
-If true (the default), blocks whose contents fail the CRC integrity check are
-re-fetched from the underlying input stream, overwriting the corrupt cached
-data. If false, cache corruption is treated as a fatal read error.
-
-@end table
-
-URL Syntax is
-@example
-shared:@var{URL}
-@end example
-
 @section srt
 
 Haivision Secure Reliable Transport Protocol via libsrt.
@@ -1921,10 +1857,6 @@ When true, use Timestamp-based Packet Delivery mode. The default behavior
 depends on the transmission type: enabled in live mode, disabled in file
 mode.
 
-@item ipv6only=@var{1|0}
-Accept IPv4 or not while using the IPv6 wildcard address. This should be set
-when listening on a IPv6 wildcard address.
-
 @end table
 
 For more information see: @url{https://github.com/Haivision/srt}.
@@ -2091,8 +2023,8 @@ database, but it does not validate that the certificate actually
 matches the host name we are trying to connect to. (With other backends,
 the host name is validated as well.)
 
-This is enabled by default. Verifying the peer requires a CA database,
-which in some cases has to be provided by the caller.
+This is disabled by default since it requires a CA database to be
+provided by the caller in many cases.
 
 @item cert_file, cert=@var{filename}
 A file containing a certificate to use in the handshake with the peer.
@@ -2163,8 +2095,8 @@ peer certificate is signed by one of the root certificates in the CA
 database, but it does not validate that the certificate actually
 matches the host name we are trying to connect to.
 
-This is enabled by default. Verifying the peer requires a CA database,
-which in some cases has to be provided by the caller.
+This is disabled by default since it requires a CA database to be
+provided by the caller in many cases.
 
 @item cert_file, cert=@var{filename}
 A file containing a certificate to use in the handshake with the peer.

doc/scaler.texi

@@ -11,88 +11,48 @@ For programmatic use, they can be set explicitly in the
 
 @table @option
 
-@anchor{scaler}
-@item scaler, scaler_sub
-Choose the scaling algorithm to use. Default value is @samp{auto} for both.
-It accepts the following values:
-
-@table @samp
-@item auto
-Aumotic choice. For @samp{scaler_sub}, this means the same algorithm as
-@samp{scaler}. For @samp{scaler}, this defaults to the scaler flag selected
-by @samp{sws_flags}.
-
-@item bilinear
-Bilinear filter. (AKA triangle filter)
-
-@item bicubic
-2-tap cubic BC-spline (AKA Mitchell-Netravali spline). The B and C parameters
-can be configured by setting @code{param0} and @code{param1}, defaulting to
-0.0 and 0.6 respectively.
-
-@item point, neighbor
-Point sampling (AKA nearest neighbor).
-
-@item area
-Area averaging. Equivalent to @samp{bilinear} for upscaling.
-
-@item gaussian
-2-tap Gaussian filter approximation. The sharpness parameter can be configured
-by setting @code{param0}, defaulting to 3.0.
-
-@item sinc
-Unwindowed sinc filter.
-
-@item lanczos
-Lanczos resampling (sinc windowed sinc). The number of filter taps can
-be configured by setting @code{param0}, defaulting to 3.
-
-@item spline
-Unwindowed natural bicubic spline.
-@end table
-
 @anchor{sws_flags}
 @item sws_flags
 Set the scaler flags. This is also used to set the scaling
-algorithm, though this usage is deprecated in favor of setting @samp{scaler}.
-Only a single algorithm may be selected. Default value is @samp{bicubic}.
+algorithm. Only a single algorithm should be selected. Default
+value is @samp{bicubic}.
 
 It accepts the following values:
 @table @samp
 @item fast_bilinear
-Select fast bilinear scaling algorithm. (Deprecated)
+Select fast bilinear scaling algorithm.
 
 @item bilinear
-Select bilinear scaling algorithm. (Deprecated)
+Select bilinear scaling algorithm.
 
 @item bicubic
-Select bicubic scaling algorithm. (Deprecated)
+Select bicubic scaling algorithm.
 
 @item experimental
-Select experimental scaling algorithm. (Deprecated)
+Select experimental scaling algorithm.
 
 @item neighbor
-Select nearest neighbor rescaling algorithm. (Deprecated)
+Select nearest neighbor rescaling algorithm.
 
 @item area
-Select averaging area rescaling algorithm. (Deprecated)
+Select averaging area rescaling algorithm.
 
 @item bicublin
 Select bicubic scaling algorithm for the luma component, bilinear for
-chroma components. (Deprecated)
+chroma components.
 
 @item gauss
-Select Gaussian rescaling algorithm. (Deprecated)
+Select Gaussian rescaling algorithm.
 
 @item sinc
-Select sinc rescaling algorithm. (Deprecated)
+Select sinc rescaling algorithm.
 
 @item lanczos
 Select Lanczos rescaling algorithm. The default width (alpha) is 3 and can be
-changed by setting @code{param0}. (Deprecated)
+changed by setting @code{param0}.
 
 @item spline
-Select natural bicubic spline rescaling algorithm. (Deprecated)
+Select natural bicubic spline rescaling algorithm.
 
 @item print_info
 Enable printing/debug logging.
@@ -191,47 +151,6 @@ No blending
 
 @end table
 
-@item sws_backends
-Set the allowed swscale backends. This is a flags option, so multiple backends
-may be combined.
-
-@table @samp
-@item auto
-Automatic selection. Equal to either @samp{stable} or @samp{all} depending on
-whether or not the @samp{unstable} flag is set. This is the default value.
-
-@item stable
-All stable backends.
-
-@item unstable
-All unstable backends.
-
-@item all
-All available backends.
-
-@item legacy
-Legacy swscale code.
-
-@item c
-Template-based reference code.
-
-@item memcpy
-Fast path using libc @code{memcpy}.
-
-@item x86
-x86 SIMD kernels.
-
-@item aarch64
-AArch64 NEON kernels.
-
-@item spirv
-Vulkan SPIR-V backend.
-
-@item glsl
-Vulkan GLSL backend.
-
-@end table
-
 @end table
 
 @c man end SCALER OPTIONS

doc/swscale-v2.txt

@@ -13,9 +13,8 @@ threaded error diffusion pass following a multi-threaded scaling pass.
 
 SwsGraph is internally recreated whenever the image format, dimensions or
 settings change in any way. sws_scale_frame() is itself just a light-weight
-wrapper that runs ff_sws_graph_reinit() initially and on format changes,
-splits interlaced images into separate fields, and calls ff_sws_graph_run()
-on each.
+wrapper that runs ff_sws_graph_create() whenever the format changes, splits
+interlaced images into separate fields, and calls ff_sws_graph_run() on each.
 
 From the point of view of SwsGraph itself, all inputs are progressive.
 

doc/t2h.pm

@@ -55,8 +55,7 @@ sub get_formatting_function($$) {
 
 # determine texinfo version
 my $package_version = ff_get_conf('PACKAGE_VERSION');
-$package_version =~ s/\+nc$//;
-$package_version =~ s/\+?dev$//;
+$package_version =~ s/\+dev$//;
 my $program_version_num = version->declare($package_version)->numify;
 my $program_version_6_8 = $program_version_num >= 6.008000;
 
@@ -120,8 +119,29 @@ sub ffmpeg_heading_command($$$$$)
     }
 
     my $heading_level;
-    # Never use node for heading
-    if ($cmdname ne 'node') {
+    # node is used as heading if there is nothing else.
+    if ($cmdname eq 'node') {
+        if (!$output_unit or
+            (((!$output_unit->{'extra'}->{'section'}
+              and $output_unit->{'extra'}->{'node'}
+              and $output_unit->{'extra'}->{'node'} eq $command)
+             or
+             ((($output_unit->{'extra'}->{'unit_command'}
+                and $output_unit->{'extra'}->{'unit_command'} eq $command)
+               or
+               ($output_unit->{'unit_command'}
+                and $output_unit->{'unit_command'} eq $command))
+              and $command->{'extra'}
+              and not $command->{'extra'}->{'associated_section'}))
+             # bogus node may not have been normalized
+            and defined($command->{'extra'}->{'normalized'}))) {
+            if ($command->{'extra'}->{'normalized'} eq 'Top') {
+                $heading_level = 0;
+            } else {
+                $heading_level = 3;
+            }
+        }
+    } else {
         if (defined($command->{'extra'})
             and defined($command->{'extra'}->{'section_level'})) {
           $heading_level = $command->{'extra'}->{'section_level'};
@@ -133,58 +153,58 @@ sub ffmpeg_heading_command($$$$$)
         }
     }
 
-    if (defined($heading_level)) {
-        my $heading = $self->command_text($command);
-        # empty heading corresponds to an empty @top
-        if ($heading ne '') {
-            if ($root_commands{$cmdname}
-                and $sectioning_commands{$cmdname}) {
-                my $content_href = $self->command_contents_href($command, 'contents',
-                                                                $self->{'current_filename'});
-                if ($content_href) {
-                    my $this_href = $content_href =~ s/^\#toc-/\#/r;
-                    $heading .= '<span class="pull-right">'.
-                                  '<a class="anchor hidden-xs" '.
-                                     "href=\"$this_href\" aria-hidden=\"true\">".
-                ($ENV{"FA_ICONS"} ? '<i class="fa fa-link"></i>'
-                                  : '#').
-                                  '</a> '.
-                                  '<a class="anchor hidden-xs"'.
-                                     "href=\"$content_href\" aria-hidden=\"true\">".
-                ($ENV{"FA_ICONS"} ? '<i class="fa fa-navicon"></i>'
-                                  : 'TOC').
-                                  '</a>'.
-                                '</span>';
-                }
-            }
+    my $heading = $self->command_text($command);
+    # $heading not defined may happen if the command is a @node, for example
+    # if there is an error in the node.
+    if (defined($heading) and $heading ne '' and defined($heading_level)) {
 
-            my $in_preformatted;
-            if ($program_version_num >= 7.001090) {
-              $in_preformatted = $self->in_preformatted_context();
-            } else {
-              $in_preformatted = $self->in_preformatted();
+        if ($root_commands{$cmdname}
+            and $sectioning_commands{$cmdname}) {
+            my $content_href = $self->command_contents_href($command, 'contents',
+                                                            $self->{'current_filename'});
+            if ($content_href) {
+                my $this_href = $content_href =~ s/^\#toc-/\#/r;
+                $heading .= '<span class="pull-right">'.
+                              '<a class="anchor hidden-xs" '.
+                                 "href=\"$this_href\" aria-hidden=\"true\">".
+            ($ENV{"FA_ICONS"} ? '<i class="fa fa-link"></i>'
+                              : '#').
+                              '</a> '.
+                              '<a class="anchor hidden-xs"'.
+                                 "href=\"$content_href\" aria-hidden=\"true\">".
+            ($ENV{"FA_ICONS"} ? '<i class="fa fa-navicon"></i>'
+                              : 'TOC').
+                              '</a>'.
+                            '</span>';
             }
-            if ($in_preformatted) {
-                $result .= $heading."\n";
-            } else {
-                # if the level was changed, set the command name right
-                if ($cmdname ne 'node'
-                    and $heading_level ne $Texinfo::Common::command_structuring_level{$cmdname}) {
-                    $cmdname
-                        = $Texinfo::Common::level_to_structuring_command{$cmdname}->[$heading_level];
-                }
-                if ($program_version_num >= 7.000000) {
-                    $result .= &{get_formatting_function($self,'format_heading_text')}($self,
-                         $cmdname, [$cmdname], $heading,
-                         $heading_level +$self->get_conf('CHAPTER_HEADER_LEVEL') -1,
-                         $heading_id, $command);
+        }
 
-                } else {
-                  $result .= &{get_formatting_function($self,'format_heading_text')}(
-                            $self, $cmdname, $heading,
-                            $heading_level +
-                            $self->get_conf('CHAPTER_HEADER_LEVEL') - 1, $command);
-                }
+        my $in_preformatted;
+        if ($program_version_num >= 7.001090) {
+          $in_preformatted = $self->in_preformatted_context();
+        } else {
+          $in_preformatted = $self->in_preformatted();
+        }
+        if ($in_preformatted) {
+            $result .= $heading."\n";
+        } else {
+            # if the level was changed, set the command name right
+            if ($cmdname ne 'node'
+                and $heading_level ne $Texinfo::Common::command_structuring_level{$cmdname}) {
+                $cmdname
+                    = $Texinfo::Common::level_to_structuring_command{$cmdname}->[$heading_level];
+            }
+            if ($program_version_num >= 7.000000) {
+                $result .= &{get_formatting_function($self,'format_heading_text')}($self,
+                     $cmdname, [$cmdname], $heading,
+                     $heading_level +$self->get_conf('CHAPTER_HEADER_LEVEL') -1,
+                     $heading_id, $command);
+
+            } else {
+              $result .= &{get_formatting_function($self,'format_heading_text')}(
+                        $self, $cmdname, $heading,
+                        $heading_level +
+                        $self->get_conf('CHAPTER_HEADER_LEVEL') - 1, $command);
             }
         }
     }

doc/transforms.md

@@ -1,17 +1,14 @@
-# Transforms
-
-The basis transforms used for FFT and various other derived functions are based on the following unrollings.
+The basis transforms used for FFT and various other derived functions are based
+on the following unrollings.
 The functions can be easily adapted to double precision floats as well.
 
-## Parity permutation
-
+# Parity permutation
 The basis transforms described here all use the following permutation:
 
 ``` C
 void ff_tx_gen_split_radix_parity_revtab(int *revtab, int len, int inv,
                                          int basis, int dual_stride);
 ```
-
 Parity means even and odd complex numbers will be split, e.g. the even
 coefficients will come first, after which the odd coefficients will be
 placed. For example, a 4-point transform's coefficients after reordering:
@@ -36,8 +33,7 @@ register or 0. This allows to reuse SSE functions as dual-transform
 functions in AVX mode.
 If length is smaller than basis/2 this function will not do anything.
 
-## 4-point FFT transform
-
+# 4-point FFT transform
 The only permutation this transform needs is to swap the `z[1]` and `z[2]`
 elements when performing an inverse transform, which in the assembly code is
 hardcoded with the function itself being templated and duplicated for each
@@ -84,8 +80,7 @@ static void fft4(FFTComplex *z)
 }
 ```
 
-## 8-point AVX FFT transform
-
+# 8-point AVX FFT transform
 Input must be pre-permuted using the parity lookup table, generated via
 `ff_tx_gen_split_radix_parity_revtab`.
 
@@ -198,8 +193,7 @@ This theme continues throughout the document. Note that in the actual assembly c
 the paths are interleaved to improve unit saturation and CPU dependency tracking, so
 to more clearly see them, you'll need to deinterleave the instructions.
 
-## 8-point SSE/ARM64 FFT transform
-
+# 8-point SSE/ARM64 FFT transform
 Input must be pre-permuted using the parity lookup table, generated via
 `ff_tx_gen_split_radix_parity_revtab`.
 
@@ -311,8 +305,7 @@ static void fft8(FFTComplex *z)
 Most functions here are highly tuned to use x86's addsub instruction to save on
 external sign mask loading.
 
-## 16-point AVX FFT transform
-
+# 16-point AVX FFT transform
 This version expects the output of the 8 and 4-point transforms to follow the
 even/odd convention established above.
 
@@ -452,8 +445,7 @@ static void fft16(FFTComplex *z)
 }
 ```
 
-## AVX split-radix synthesis
-
+# AVX split-radix synthesis
 To create larger transforms, the following unrolling of the C split-radix
 function is used.
 
@@ -713,8 +705,8 @@ beginning to overlap, particularly `[o1]` with `[0]` after the second iteration.
 To iterate further, set `z = &z[16]` via `z += 8` for the second iteration. After
 the 4th iteration, the layout resets, so repeat the same.
 
-## 15-point AVX FFT transform
 
+# 15-point AVX FFT transform
 The 15-point transform is based on the following unrolling. The input
 must be permuted via the following loop:
 

ffbuild/common.mak

@@ -2,6 +2,14 @@
 # common bits used by all libraries
 #
 
+DEFAULT_X86ASMD=.dbg
+
+ifeq ($(DBG),1)
+X86ASMD=$(DEFAULT_X86ASMD)
+else
+X86ASMD=
+endif
+
 ifndef SUBDIR
 
 LINK = $(LD) $(1)
@@ -27,7 +35,7 @@ M      = @$(call ECHO,$(TAG),$@);
 $(foreach VAR,$(BRIEF), \
     $(eval override $(VAR) = @$$(call ECHO,$(VAR),$$(MSG)); $($(VAR))))
 $(foreach VAR,$(SILENT),$(eval override $(VAR) = @$($(VAR))))
-$(eval INSTALL = @$(call ECHO,INSTALL,$$(^:$(SRC_PATH)/%=%)); $(INSTALL))
+$(eval INSTALL = @$(call ECHO,INSTALL,$$(^:$(SRC_DIR)/%=%)); $(INSTALL))
 endif
 
 # Prepend to a recursively expanded variable without making it simply expanded.
@@ -97,6 +105,10 @@ COMPILE_LASX = $(call COMPILE,CC,LASXFLAGS)
 %_host.o: %.c
 	$(COMPILE_HOSTC)
 
+%$(DEFAULT_X86ASMD).asm: %.asm
+	$(DEPX86ASM) $(X86ASMFLAGS) -M -o $@ $< > $(@:.asm=.d)
+	$(X86ASM) $(X86ASMFLAGS) -e $< | sed '/^%/d;/^$$/d;' > $@
+
 %.o: %.asm
 	$(COMPILE_X86ASM)
 	-$(if $(ASMSTRIPFLAGS), $(STRIP) $(ASMSTRIPFLAGS) $@)
@@ -114,8 +126,8 @@ $(BIN2CEXE): ffbuild/bin2c_host.o
 	$(HOSTLD) $(HOSTLDFLAGS) $(HOSTLD_O) $^ $(HOSTEXTRALIBS)
 
 RUN_BIN2C = $(BIN2C) $(patsubst $(SRC_PATH)/%,$(SRC_LINK)/%,$<) $@ $(subst .,_,$(basename $(notdir $@)))
-RUN_GZIP  = $(M)mkdir -p $(dir $@) && gzip -nc9 $(patsubst $(SRC_PATH)/%,$(SRC_LINK)/%,$<) >$@
-RUN_MINIFY = $(M)mkdir -p $(dir $@) && sed 's!/\\*.*\\*/!!g' $< | tr '\n' ' ' | tr -s ' ' | sed 's/^ //; s/ $$//' > $@
+RUN_GZIP  = $(M)gzip -nc9 $(patsubst $(SRC_PATH)/%,$(SRC_LINK)/%,$<) >$@
+RUN_MINIFY = $(M)sed 's!/\\*.*\\*/!!g' $< | tr '\n' ' ' | tr -s ' ' | sed 's/^ //; s/ $$//' > $@
 %.gz: TAG = GZIP
 %.min: TAG = MINIFY
 
@@ -185,7 +197,7 @@ endif
 clean::
 	$(RM) $(BIN2CEXE) $(CLEANSUFFIXES:%=ffbuild/%)
 
-%.c %.h %.S %.pc %.ver %.version: TAG = GEN
+%.c %.h %.pc %.ver %.version: TAG = GEN
 
 # Dummy rule to stop make trying to rebuild removed or renamed headers
 %.h %_template.c:
@@ -205,7 +217,6 @@ OBJS      += $(OBJS-yes)
 SHLIBOBJS += $(SHLIBOBJS-yes)
 STLIBOBJS += $(STLIBOBJS-yes)
 FFLIBS    := $($(NAME)_FFLIBS) $(FFLIBS-yes) $(FFLIBS)
-DEVPROGS  += $(DEVPROGS-yes)
 TESTPROGS += $(TESTPROGS-yes)
 
 LDLIBS       = $(FFLIBS:%=%$(BUILDSUF))
@@ -214,8 +225,7 @@ FFEXTRALIBS := $(LDLIBS:%=$(LD_LIB)) $(foreach lib,EXTRALIBS-$(NAME) $(FFLIBS:%=
 OBJS      := $(sort $(OBJS:%=$(SUBDIR)%))
 SHLIBOBJS := $(sort $(SHLIBOBJS:%=$(SUBDIR)%))
 STLIBOBJS := $(sort $(STLIBOBJS:%=$(SUBDIR)%))
-TESTOBJS  := $(TESTOBJS:%=$(SUBDIR)tests/%) $(TESTPROGS:%=$(SUBDIR)tests/%.o) $(DEVPROGS:%=$(SUBDIR)%.o)
-DEVPROGS  := $(DEVPROGS:%=$(SUBDIR)%$(EXESUF))
+TESTOBJS  := $(TESTOBJS:%=$(SUBDIR)tests/%) $(TESTPROGS:%=$(SUBDIR)tests/%.o)
 TESTPROGS := $(TESTPROGS:%=$(SUBDIR)tests/%$(EXESUF))
 HOSTOBJS  := $(HOSTPROGS:%=$(SUBDIR)%.o)
 HOSTPROGS := $(HOSTPROGS:%=$(SUBDIR)%$(HOSTEXESUF))
@@ -256,14 +266,14 @@ $(TOOLOBJS): | tools
 
 OUTDIRS := $(OUTDIRS) $(dir $(OBJS) $(HOBJS) $(HOSTOBJS) $(SHLIBOBJS) $(STLIBOBJS) $(TESTOBJS))
 
-CLEANSUFFIXES     = *.d *.gcda *.gcno *.h.c *.ho *.map *.o *.objs *.pc *.ptx *.ptx.gz *.ptx.c *.spv *.spv.gz *.spv.c *.gen.asm *.gen.c *.gen.S *.ver *.version *.html.gz *.html.c *.css.min.gz *.css.min *.css.c *~ *.ilk *.pdb
+CLEANSUFFIXES     = *.d *.gcda *.gcno *.h.c *.ho *.map *.o *.objs *.pc *.ptx *.ptx.gz *.ptx.c *.spv *.spv.gz *.spv.c *.ver *.version *.html.gz *.html.c *.css.min.gz *.css.min *.css.c  *$(DEFAULT_X86ASMD).asm *~ *.ilk *.pdb
 LIBSUFFIXES       = *.a *.lib *.so *.so.* *.dylib *.dll *.def *.dll.a
 
 define RULES
 clean::
-	$(RM) $(DEVPROGS) $(HOSTPROGS) $(TESTPROGS) $(TOOLS)
+	$(RM) $(HOSTPROGS) $(TESTPROGS) $(TOOLS)
 endef
 
 $(eval $(RULES))
 
--include $(wildcard $(OBJS:.o=.d) $(HOSTOBJS:.o=.d) $(TESTOBJS:.o=.d) $(HOBJS:.o=.d) $(SHLIBOBJS:.o=.d) $(STLIBOBJS:.o=.d) $(SPVOBJS:.spv.o=.d))
+-include $(wildcard $(OBJS:.o=.d) $(HOSTOBJS:.o=.d) $(TESTOBJS:.o=.d) $(HOBJS:.o=.d) $(SHLIBOBJS:.o=.d) $(STLIBOBJS:.o=.d) $(SPVOBJS:.spv.o=.d)) $(OBJS:.o=$(DEFAULT_X86ASMD).d)

ffbuild/library.mak

@@ -14,15 +14,6 @@ INSTHEADERS := $(INSTHEADERS) $(HEADERS:%=$(SUBDIR)%)
 all-$(CONFIG_STATIC): $(SUBDIR)$(LIBNAME)  $(SUBDIR)lib$(FULLNAME).pc
 all-$(CONFIG_SHARED): $(SUBDIR)$(SLIBNAME) $(SUBDIR)lib$(FULLNAME).pc
 
-# Make <4.0 does not support the built-in file function;
-# versions that do support it should use it, as it's
-# faster and isn't bound by command line length limits.
-ifeq (4.0,$(firstword $(sort 4.0 $(MAKE_VERSION))))
-HAVE_BUILTIN_FILE := yes
-else
-HAVE_BUILTIN_FILE := no
-endif
-
 LIBOBJS := $(OBJS) $(SHLIBOBJS) $(STLIBOBJS) $(SUBDIR)%.h.o $(TESTOBJS)
 $(LIBOBJS) $(LIBOBJS:.o=.s) $(LIBOBJS:.o=.i):   CPPFLAGS += -DHAVE_AV_CONFIG_H
 
@@ -45,11 +36,7 @@ endif
 $(SUBDIR)$(LIBNAME): $(OBJS) $(STLIBOBJS)
 	$(RM) $@
 ifeq ($(RESPONSE_FILES),yes)
-ifeq ($(HAVE_BUILTIN_FILE),yes)
-	$(file >$@.objs,$^)
-else
 	$(Q)echo $^ > $@.objs
-endif
 	$(AR) $(ARFLAGS) $(AR_O) @$@.objs
 else
 	$(AR) $(ARFLAGS) $(AR_O) $^
@@ -64,13 +51,15 @@ install-libs-$(CONFIG_SHARED): install-lib$(NAME)-shared
 
 define RULES
 $(TOOLS):     THISLIB = $(FULLNAME:%=$(LD_LIB))
-$(TESTPROGS) $(DEVPROGS): THISLIB = $(SUBDIR)$(LIBNAME)
+$(TESTPROGS): THISLIB = $(SUBDIR)$(LIBNAME)
+
+$(LIBOBJS): CPPFLAGS += -DBUILDING_$(NAME)
 
 $(NAME)LINK_EXE_ARGS = $(LDFLAGS) $(LDEXEFLAGS)
 $(NAME)LINK_SO_ARGS = $(SHFLAGS) $(LDFLAGS) $(LDSOFLAGS)
 $(NAME)LINK_EXTRA = $(FFEXTRALIBS)
 
-$(DEVPROGS) $(TESTPROGS) $(TOOLS): %$(EXESUF): %.o
+$(TESTPROGS) $(TOOLS): %$(EXESUF): %.o
 	$$(call LINK,$$(call $(NAME)LINK_EXE_ARGS) $$(LD_O) $$(filter %.o,$$^) $$(THISLIB) $$(call $(NAME)LINK_EXTRA) $$(EXTRALIBS-$$(*F)) $$(ELIBS))
 
 $(SUBDIR)lib$(NAME).version: $(SUBDIR)version.h $(SUBDIR)version_major.h | $(SUBDIR)
@@ -86,20 +75,14 @@ $(SUBDIR)$(SLIBNAME): $(SUBDIR)$(SLIBNAME_WITH_MAJOR)
 	$(Q)cd ./$(SUBDIR) && $(LN_S) $(SLIBNAME_WITH_MAJOR) $(SLIBNAME)
 
 $(SUBDIR)$(SLIBNAME_WITH_MAJOR): $(OBJS) $(SHLIBOBJS) $(SUBDIR)lib$(NAME).ver
+	$(SLIB_CREATE_DEF_CMD)
 ifeq ($(RESPONSE_FILES),yes)
-ifeq ($(HAVE_BUILTIN_FILE),yes)
-	$$(file >$$@.objs,$$(filter %.o,$$^))
-else
 	$(Q)echo $$(filter %.o,$$^) > $$@.objs
-endif
-endif
-	$(Q)$(SLIB_CREATE_DEF_CMD)
-ifeq ($(RESPONSE_FILES),yes)
 	$$(call LINK,$$(call $(NAME)LINK_SO_ARGS) $$(LD_O) @$$@.objs $$(call $(NAME)LINK_EXTRA))
 else
 	$$(call LINK,$$(call $(NAME)LINK_SO_ARGS) $$(LD_O) $$(filter %.o,$$^) $$(call $(NAME)LINK_EXTRA))
 endif
-	$(Q)$(SLIB_EXTRA_CMD)
+	$(SLIB_EXTRA_CMD)
 	-$(RM) $$@.objs
 
 ifdef SUBDIR
@@ -113,9 +96,7 @@ clean::
 install-lib$(NAME)-shared: $(SUBDIR)$(SLIBNAME)
 	$(Q)mkdir -p "$(SHLIBDIR)"
 	$$(INSTALL) -m 755 $$< "$(SHLIBDIR)/$(SLIB_INSTALL_NAME)"
-ifneq ($(STRIPTYPE),nostrip)
 	$$(STRIP) "$(SHLIBDIR)/$(SLIB_INSTALL_NAME)"
-endif
 	$(Q)$(foreach F,$(SLIB_INSTALL_LINKS),(cd "$(SHLIBDIR)" && $(LN_S) $(SLIB_INSTALL_NAME) $(F));)
 	$(if $(SLIB_INSTALL_EXTRA_SHLIB),$$(INSTALL) -m 644 $(SLIB_INSTALL_EXTRA_SHLIB:%=$(SUBDIR)%) "$(SHLIBDIR)")
 	$(if $(SLIB_INSTALL_EXTRA_LIB),$(Q)mkdir -p "$(LIBDIR)")
@@ -153,6 +134,6 @@ endef
 $(eval $(RULES))
 
 $(TOOLS):     $(DEP_LIBS) $(SUBDIR)$($(CONFIG_SHARED:yes=S)LIBNAME)
-$(TESTPROGS) $(DEVPROGS): $(DEP_LIBS) $(SUBDIR)$(LIBNAME)
+$(TESTPROGS): $(DEP_LIBS) $(SUBDIR)$(LIBNAME)
 
 testprogs: $(TESTPROGS)

fftools/cmdutils.c

@@ -1267,7 +1267,7 @@ unsigned stream_specifier_match(const StreamSpecifier *ss,
                 break;
             }
         }
-        av_fallthrough;
+        // fall-through
     case STREAM_LIST_GROUP_IDX:
         if (ss->stream_list == STREAM_LIST_GROUP_IDX &&
             ss->list_id >= 0 && ss->list_id < s->nb_stream_groups)

fftools/ffmpeg.c

@@ -216,16 +216,6 @@ void term_init(void)
 #endif
 
 #if HAVE_TERMIOS_H
-    /* A closed fd 0 is later reused by the first opened input file. read_key()
-     * would then read from that input instead of the terminal and corrupt the
-     * stream, so disable interaction when fd 0 is not an open descriptor.
-     */
-    if (stdin_interaction && fcntl(0, F_GETFD) == -1) {
-        av_log(NULL, AV_LOG_WARNING,
-               "fd 0 is not an open file descriptor, stdin interaction disabled\n");
-        stdin_interaction = 0;
-    }
-
     if (stdin_interaction) {
         struct termios tty;
         if (tcgetattr (0, &tty) == 0) {

fftools/ffmpeg.h

@@ -51,6 +51,16 @@
 
 #include "libswresample/swresample.h"
 
+// deprecated features
+#define FFMPEG_OPT_QPHIST 1
+#define FFMPEG_OPT_ADRIFT_THRESHOLD 1
+#define FFMPEG_OPT_ENC_TIME_BASE_NUM 1
+#define FFMPEG_OPT_TOP 1
+#define FFMPEG_OPT_FORCE_KF_SOURCE_NO_DROP 1
+#define FFMPEG_OPT_VSYNC_DROP 1
+#define FFMPEG_OPT_VSYNC 1
+#define FFMPEG_OPT_FILTER_SCRIPT 1
+
 #define FFMPEG_ERROR_RATE_EXCEEDED FFERRTAG('E', 'R', 'E', 'D')
 
 enum VideoSyncMethod {
@@ -59,6 +69,9 @@ enum VideoSyncMethod {
     VSYNC_CFR,
     VSYNC_VFR,
     VSYNC_VSCFR,
+#if FFMPEG_OPT_VSYNC_DROP
+    VSYNC_DROP,
+#endif
 };
 
 enum EncTimeBase {
@@ -205,17 +218,21 @@ typedef struct OptionsContext {
     SpecifierOptList display_rotations;
     SpecifierOptList display_hflips;
     SpecifierOptList display_vflips;
-    SpecifierOptList mastering_displays;
-    SpecifierOptList content_lights;
     SpecifierOptList rc_overrides;
     SpecifierOptList intra_matrices;
     SpecifierOptList inter_matrices;
     SpecifierOptList chroma_intra_matrices;
+#if FFMPEG_OPT_TOP
+    SpecifierOptList top_field_first;
+#endif
     SpecifierOptList metadata_map;
     SpecifierOptList presets;
     SpecifierOptList copy_initial_nonkeyframes;
     SpecifierOptList copy_prior_start;
     SpecifierOptList filters;
+#if FFMPEG_OPT_FILTER_SCRIPT
+    SpecifierOptList filter_scripts;
+#endif
     SpecifierOptList reinit_filters;
     SpecifierOptList drop_changed;
     SpecifierOptList fix_sub_duration;
@@ -417,6 +434,9 @@ enum DecoderFlags {
     // decoder should override timestamps by fixed framerate
     // from DecoderOpts.framerate
     DECODER_FLAG_FRAMERATE_FORCED = (1 << 2),
+#if FFMPEG_OPT_TOP
+    DECODER_FLAG_TOP_FIELD_FIRST  = (1 << 3),
+#endif
     DECODER_FLAG_SEND_END_TS      = (1 << 4),
     // force bitexact decoding
     DECODER_FLAG_BITEXACT         = (1 << 5),
@@ -480,6 +500,9 @@ typedef struct InputStream {
 
     /* framerate forced with -r */
     AVRational            framerate;
+#if FFMPEG_OPT_TOP
+    int                   top_field_first;
+#endif
 
     int                   fix_sub_duration;
 
@@ -581,6 +604,9 @@ typedef struct EncStats {
 
 enum {
     KF_FORCE_SOURCE         = 1,
+#if FFMPEG_OPT_FORCE_KF_SOURCE_NO_DROP
+    KF_FORCE_SOURCE_NO_DROP = 2,
+#endif
     // force keyframe if lavfi.scd.time metadata is set
     KF_FORCE_SCD_METADATA = 3,
 };
@@ -637,6 +663,10 @@ typedef struct OutputStream {
 
     Encoder *enc;
 
+    /* video only */
+#if FFMPEG_OPT_TOP
+    int top_field_first;
+#endif
     int bitexact;
     int bits_per_raw_sample;
 
@@ -783,7 +813,7 @@ int check_avoptions_used(const AVDictionary *opts, const AVDictionary *opts_used
 int assert_file_overwrite(const char *filename);
 int find_codec(void *logctx, const char *name,
                enum AVMediaType type, int encoder, const AVCodec **codec);
-int parse_and_set_vsync(const char *arg, enum VideoSyncMethod *vsync_var, int file_idx, int st_idx);
+int parse_and_set_vsync(const char *arg, enum VideoSyncMethod *vsync_var, int file_idx, int st_idx, int is_global);
 
 int filtergraph_is_simple(const FilterGraph *fg);
 int fg_create_simple(FilterGraph **pfg,

fftools/ffmpeg_dec.c

@@ -387,6 +387,13 @@ fail:
 static int video_frame_process(DecoderPriv *dp, AVFrame *frame,
                                unsigned *outputs_mask)
 {
+#if FFMPEG_OPT_TOP
+    if (dp->flags & DECODER_FLAG_TOP_FIELD_FIRST) {
+        av_log(dp, AV_LOG_WARNING, "-top is deprecated, use the setfield filter instead\n");
+        frame->flags |= AV_FRAME_FLAG_TOP_FIELD_FIRST;
+    }
+#endif
+
     if (frame->format == dp->hwaccel_pix_fmt) {
         int err = hwaccel_retrieve_data(dp->dec_ctx, frame);
         if (err < 0)

fftools/ffmpeg_demux.c

@@ -28,7 +28,6 @@
 #include "libavutil/display.h"
 #include "libavutil/error.h"
 #include "libavutil/intreadwrite.h"
-#include "libavutil/mastering_display_metadata.h"
 #include "libavutil/mem.h"
 #include "libavutil/opt.h"
 #include "libavutil/parseutils.h"
@@ -69,8 +68,6 @@ typedef struct DemuxStream {
     int                      autorotate;
     int                      apply_cropping;
     int                      force_display_matrix;
-    int                      force_mastering_display;
-    int                      force_content_light;
     int                      drop_changed;
 
 
@@ -998,7 +995,11 @@ int ist_use(InputStream *ist, int decoding_needed,
 
         ds->dec_opts.flags |= (!!ist->fix_sub_duration * DECODER_FLAG_FIX_SUB_DURATION) |
                               (!!is_unreliable * DECODER_FLAG_TS_UNRELIABLE) |
-                              (!!(d->loop && is_audio) * DECODER_FLAG_SEND_END_TS);
+                              (!!(d->loop && is_audio) * DECODER_FLAG_SEND_END_TS)
+#if FFMPEG_OPT_TOP
+                              | ((ist->top_field_first >= 0) * DECODER_FLAG_TOP_FIELD_FIRST)
+#endif
+                             ;
 
         if (ist->framerate.num) {
             ds->dec_opts.flags     |= DECODER_FLAG_FRAMERATE_FORCED;
@@ -1260,125 +1261,6 @@ static int add_display_matrix_to_stream(const OptionsContext *o,
     return 0;
 }
 
-static int add_mastering_display_to_stream(const OptionsContext *o,
-                                           AVFormatContext *ctx, InputStream *ist)
-{
-    AVStream *st = ist->st;
-    DemuxStream *ds = ds_from_ist(ist);
-    AVMasteringDisplayMetadata *master_display;
-    AVPacketSideData *sd;
-    const char *p = NULL;
-    const int chroma_den = 50000;
-    const int luma_den = 10000;
-    size_t size;
-    int ret;
-
-    opt_match_per_stream_str(ist, &o->mastering_displays, ctx, st, &p);
-
-    if (!p)
-        return 0;
-
-    master_display = av_mastering_display_metadata_alloc_size(&size);
-    if (!master_display)
-        return AVERROR(ENOMEM);
-
-    ret = sscanf(p,
-                 "G(%u,%u)B(%u,%u)R(%u,%u)WP(%u,%u)L(%u,%u)",
-                 (unsigned*)&master_display->display_primaries[1][0].num,
-                 (unsigned*)&master_display->display_primaries[1][1].num,
-                 (unsigned*)&master_display->display_primaries[2][0].num,
-                 (unsigned*)&master_display->display_primaries[2][1].num,
-                 (unsigned*)&master_display->display_primaries[0][0].num,
-                 (unsigned*)&master_display->display_primaries[0][1].num,
-                 (unsigned*)&master_display->white_point[0].num,
-                 (unsigned*)&master_display->white_point[1].num,
-                 (unsigned*)&master_display->max_luminance.num,
-                 (unsigned*)&master_display->min_luminance.num);
-
-    if (ret != 10 ||
-        (unsigned)(master_display->display_primaries[1][0].num | master_display->display_primaries[1][1].num |
-                   master_display->display_primaries[2][0].num | master_display->display_primaries[2][1].num |
-                   master_display->display_primaries[0][0].num | master_display->display_primaries[0][1].num |
-                   master_display->white_point[0].num | master_display->white_point[1].num) > UINT16_MAX ||
-        (unsigned)(master_display->max_luminance.num  | master_display->min_luminance.num) > INT_MAX ||
-                   master_display->min_luminance.num  > master_display->max_luminance.num) {
-        av_freep(&master_display);
-        av_log(ist, AV_LOG_ERROR, "Failed to parse mastering display option\n");
-        return AVERROR(EINVAL);
-    }
-
-    master_display->display_primaries[1][0].den = chroma_den;
-    master_display->display_primaries[1][1].den = chroma_den;
-    master_display->display_primaries[2][0].den = chroma_den;
-    master_display->display_primaries[2][1].den = chroma_den;
-    master_display->display_primaries[0][0].den = chroma_den;
-    master_display->display_primaries[0][1].den = chroma_den;
-    master_display->white_point[0].den = chroma_den;
-    master_display->white_point[1].den = chroma_den;
-    master_display->max_luminance.den = luma_den;
-    master_display->min_luminance.den = luma_den;
-
-    master_display->has_primaries = 1;
-    master_display->has_luminance = 1;
-
-    sd = av_packet_side_data_add(&st->codecpar->coded_side_data,
-                                 &st->codecpar->nb_coded_side_data,
-                                 AV_PKT_DATA_MASTERING_DISPLAY_METADATA,
-                                 (uint8_t *)master_display, size, 0);
-    if (!sd) {
-        av_freep(&master_display);
-        return AVERROR(ENOMEM);
-    }
-
-    ds->force_mastering_display = 1;
-
-    return 0;
-}
-
-static int add_content_light_to_stream(const OptionsContext *o,
-                                       AVFormatContext *ctx, InputStream *ist)
-{
-    AVStream *st = ist->st;
-    DemuxStream *ds = ds_from_ist(ist);
-    AVContentLightMetadata *cll;
-    AVPacketSideData *sd;
-    const char *p = NULL;
-    size_t size;
-    int ret;
-
-    opt_match_per_stream_str(ist, &o->content_lights, ctx, st, &p);
-
-    if (!p)
-        return 0;
-
-    cll = av_content_light_metadata_alloc(&size);
-    if (!cll)
-        return AVERROR(ENOMEM);
-
-    ret = sscanf(p, "%u,%u",
-                 (unsigned*)&cll->MaxCLL,
-                 (unsigned*)&cll->MaxFALL);
-
-    if (ret != 2 || (unsigned)(cll->MaxCLL | cll->MaxFALL) > UINT16_MAX) {
-        av_freep(&cll);
-        av_log(ist, AV_LOG_ERROR, "Failed to parse content light option\n");
-        return AVERROR(EINVAL);
-    }
-
-    sd = av_packet_side_data_add(&st->codecpar->coded_side_data,
-                                 &st->codecpar->nb_coded_side_data,
-                                 AV_PKT_DATA_CONTENT_LIGHT_LEVEL,
-                                 (uint8_t *)cll, size, 0);
-    if (!sd) {
-        av_freep(&cll);
-        return AVERROR(ENOMEM);
-    }
-
-    ds->force_content_light = 1;
-
-    return 0;
-}
-
 static const char *input_stream_item_name(void *obj)
 {
     const DemuxStream *ds = obj;
@@ -1432,7 +1314,6 @@ static int ist_add(const OptionsContext *o, Demuxer *d, AVStream *st, AVDictiona
     const char *bsfs = NULL;
     char *next;
     const char *discard_str = NULL;
-    AVBPrint bp;
     int ret;
 
     ds  = demux_stream_alloc(d, st);
@@ -1498,14 +1379,6 @@ static int ist_add(const OptionsContext *o, Demuxer *d, AVStream *st, AVDictiona
         if (ret < 0)
             return ret;
 
-        ret = add_mastering_display_to_stream(o, ic, ist);
-        if (ret < 0)
-            return ret;
-
-        ret = add_content_light_to_stream(o, ic, ist);
-        if (ret < 0)
-            return ret;
-
         opt_match_per_stream_str(ist, &o->hwaccels, ic, st, &hwaccel);
         opt_match_per_stream_str(ist, &o->hwaccel_output_formats, ic, st,
                                        &hwaccel_output_format);
@@ -1623,26 +1496,15 @@ static int ist_add(const OptionsContext *o, Demuxer *d, AVStream *st, AVDictiona
     av_dict_set_int(&ds->decoder_opts, "apply_cropping",
                     ds->apply_cropping && ds->apply_cropping != CROP_CONTAINER, 0);
 
-    av_bprint_init(&bp, 0, AV_BPRINT_SIZE_AUTOMATIC);
     if (ds->force_display_matrix) {
+        char buf[32];
         if (av_dict_get(ds->decoder_opts, "side_data_prefer_packet", NULL, 0))
-            av_bprintf(&bp, ",");
-        av_bprintf(&bp, "displaymatrix");
+            buf[0] = ',';
+        else
+            buf[0] = '\0';
+        av_strlcat(buf, "displaymatrix", sizeof(buf));
+        av_dict_set(&ds->decoder_opts, "side_data_prefer_packet", buf, AV_DICT_APPEND);
     }
-    if (ds->force_mastering_display) {
-        if (bp.len || av_dict_get(ds->decoder_opts, "side_data_prefer_packet", NULL, 0))
-            av_bprintf(&bp, ",");
-        av_bprintf(&bp, "mastering_display_metadata");
-    }
-    if (ds->force_content_light) {
-        if (bp.len || av_dict_get(ds->decoder_opts, "side_data_prefer_packet", NULL, 0))
-            av_bprintf(&bp, ",");
-        av_bprintf(&bp, "content_light_level");
-    }
-    if (bp.len)
-        av_dict_set(&ds->decoder_opts, "side_data_prefer_packet", bp.str, AV_DICT_APPEND);
-    av_bprint_finalize(&bp, NULL);
-
     /* Attached pics are sparse, therefore we would not want to delay their decoding
      * till EOF. */
     if (ist->st->disposition & AV_DISPOSITION_ATTACHED_PIC)
@@ -1659,6 +1521,12 @@ static int ist_add(const OptionsContext *o, Demuxer *d, AVStream *st, AVDictiona
                 return ret;
             }
         }
+
+#if FFMPEG_OPT_TOP
+        ist->top_field_first = -1;
+        opt_match_per_stream_int(ist, &o->top_field_first, ic, st, &ist->top_field_first);
+#endif
+
         break;
     case AVMEDIA_TYPE_AUDIO: {
         const char *ch_layout_str = NULL;
@@ -1804,11 +1672,6 @@ static int istg_parse_tile_grid(const OptionsContext *o, Demuxer *d, InputStream
 
     if (tg->nb_tiles == 1)
         return 0;
-    if (!tg->nb_tiles) {
-        av_log(istg, AV_LOG_FATAL, "A demuxer exported an invalid tile group stream group. "
-                                   "This is a bug, please report it.\n");
-        return AVERROR_BUG;
-    }
 
     memset(&opts, 0, sizeof(opts));
 

fftools/ffmpeg_enc.c

@@ -227,9 +227,6 @@ int enc_open(void *opaque, const AVFrame *frame)
                    frame->ch_layout.nb_channels > 0);
         enc_ctx->sample_fmt     = frame->format;
         enc_ctx->sample_rate    = frame->sample_rate;
-        if (!enc_ctx->frame_size && (!(enc->capabilities & AV_CODEC_CAP_VARIABLE_FRAME_SIZE) ||
-                                      (enc_ctx->flags2 & AV_CODEC_FLAG2_FIXED_FRAME_SIZE)))
-            enc_ctx->frame_size = frame->nb_samples;
         ret = av_channel_layout_copy(&enc_ctx->ch_layout, &frame->ch_layout);
         if (ret < 0)
             return ret;
@@ -283,8 +280,17 @@ int enc_open(void *opaque, const AVFrame *frame)
         }
 
         if (enc_ctx->flags & (AV_CODEC_FLAG_INTERLACED_DCT | AV_CODEC_FLAG_INTERLACED_ME) ||
-            (frame->flags & AV_FRAME_FLAG_INTERLACED)) {
-            int top_field_first = !!(frame->flags & AV_FRAME_FLAG_TOP_FIELD_FIRST);
+            (frame->flags & AV_FRAME_FLAG_INTERLACED)
+#if FFMPEG_OPT_TOP
+            || ost->top_field_first >= 0
+#endif
+            ) {
+            int top_field_first =
+#if FFMPEG_OPT_TOP
+                ost->top_field_first >= 0 ?
+                ost->top_field_first :
+#endif
+                !!(frame->flags & AV_FRAME_FLAG_TOP_FIELD_FIRST);
 
             if (enc->id == AV_CODEC_ID_MJPEG)
                 enc_ctx->field_order = top_field_first ? AV_FIELD_TT : AV_FIELD_BB;
@@ -796,6 +802,13 @@ static int frame_encode(OutputStream *ost, AVFrame *frame, AVPacket *pkt)
         if (type == AVMEDIA_TYPE_VIDEO) {
             frame->quality   = e->enc_ctx->global_quality;
             frame->pict_type = forced_kf_apply(e, &ost->kf, frame);
+
+#if FFMPEG_OPT_TOP
+            if (ost->top_field_first >= 0) {
+                frame->flags &= ~AV_FRAME_FLAG_TOP_FIELD_FIRST;
+                frame->flags |= AV_FRAME_FLAG_TOP_FIELD_FIRST * (!!ost->top_field_first);
+            }
+#endif
         } else {
             if (!(e->enc_ctx->codec->capabilities & AV_CODEC_CAP_PARAM_CHANGE) &&
                 e->enc_ctx->ch_layout.nb_channels != frame->ch_layout.nb_channels) {

fftools/ffmpeg_filter.c

@@ -27,7 +27,6 @@
 #include "libavfilter/buffersink.h"
 #include "libavfilter/buffersrc.h"
 
-#include "libavutil/attributes.h"
 #include "libavutil/avassert.h"
 #include "libavutil/avstring.h"
 #include "libavutil/bprint.h"
@@ -51,6 +50,8 @@ typedef struct FilterGraphPriv {
     // true when the filtergraph contains only meta filters
     // that do not modify the frame data
     int              is_meta;
+    // source filters are present in the graph
+    int              have_sources;
     int              disable_conversions;
 
     unsigned         nb_outputs_done;
@@ -1037,7 +1038,6 @@ void fg_free(FilterGraph **pfg)
         av_frame_free(&ifp->opts.fallback);
 
         av_buffer_unref(&ifp->hw_frames_ctx);
-        av_channel_layout_uninit(&ifp->ch_layout);
         av_freep(&ifilter->linklabel);
         av_freep(&ifp->opts.name);
         av_frame_side_data_free(&ifp->side_data, &ifp->nb_side_data);
@@ -1143,6 +1143,16 @@ int fg_create(FilterGraph **pfg, char **graph_desc, Scheduler *sch,
     if (ret < 0)
         goto fail;
 
+    for (unsigned i = 0; i < graph->nb_filters; i++) {
+        const AVFilter *f = graph->filters[i]->filter;
+        if ((!avfilter_filter_pad_count(f, 0) &&
+             !(f->flags & AVFILTER_FLAG_DYNAMIC_INPUTS)) ||
+            !strcmp(f->name, "apad")) {
+            fgp->have_sources = 1;
+            break;
+        }
+    }
+
     for (AVFilterInOut *cur = inputs; cur; cur = cur->next) {
         InputFilter *const ifilter = ifilter_alloc(fg);
 
@@ -1671,10 +1681,7 @@ static int configure_output_video_filter(FilterGraphPriv *fgp, AVFilterGraph *gr
         av_frame_side_data_remove(&ofp->side_data, &ofp->nb_side_data, AV_FRAME_DATA_DISPLAYMATRIX);
     }
 
-    if ((ofp->width || ofp->height) && (ofp->flags & OFILTER_FLAG_AUTOSCALE) &&
-        // skip add scale for hardware format
-        !(ofp->format != AV_PIX_FMT_NONE &&
-          av_pix_fmt_desc_get(ofp->format)->flags & AV_PIX_FMT_FLAG_HWACCEL)) {
+    if ((ofp->width || ofp->height) && (ofp->flags & OFILTER_FLAG_AUTOSCALE)) {
         char args[255];
         AVFilterContext *filter;
         const AVDictionaryEntry *e = NULL;
@@ -1799,8 +1806,10 @@ static int configure_output_audio_filter(FilterGraphPriv *fgp, AVFilterGraph *gr
         pad_idx = 0;
     }
 
-    if (ofilter->apad)
+    if (ofilter->apad) {
         AUTO_INSERT_FILTER("-apad", "apad", ofilter->apad);
+        fgp->have_sources = 1;
+    }
 
     snprintf(name, sizeof(name), "trim for output %s", ofilter->output_name);
     ret = insert_trim(fgp, ofp->trim_start_us, ofp->trim_duration_us,
@@ -2257,7 +2266,8 @@ static int ifilter_parameters_from_frame(InputFilter *ifilter, const AVFrame *fr
     for (int i = 0; i < frame->nb_side_data; i++) {
         const AVSideDataDescriptor *desc = av_frame_side_data_desc(frame->side_data[i]->type);
 
-        if (!(desc->props & AV_SIDE_DATA_PROP_GLOBAL))
+        if (!(desc->props & AV_SIDE_DATA_PROP_GLOBAL) ||
+            frame->side_data[i]->type == AV_FRAME_DATA_DISPLAYMATRIX)
             continue;
 
         ret = av_frame_side_data_clone(&ifp->side_data,
@@ -2268,11 +2278,8 @@ static int ifilter_parameters_from_frame(InputFilter *ifilter, const AVFrame *fr
     }
 
     sd = av_frame_get_side_data(frame, AV_FRAME_DATA_DISPLAYMATRIX);
-    if (sd) {
+    if (sd)
         memcpy(ifp->displaymatrix, sd->data, sizeof(ifp->displaymatrix));
-        if (ifp->opts.flags & IFILTER_FLAG_AUTOROTATE)
-            av_frame_side_data_remove(&ifp->side_data, &ifp->nb_side_data, AV_FRAME_DATA_DISPLAYMATRIX);
-    }
     ifp->displaymatrix_present = !!sd;
 
     /* Copy downmix related side data to InputFilterPriv so it may be propagated
@@ -2540,7 +2547,11 @@ static void video_sync_process(OutputFilterPriv *ofp, AVFrame *frame,
 
     if (delta0 < 0 &&
         delta > 0 &&
-        fps->vsync_method != VSYNC_PASSTHROUGH) {
+        fps->vsync_method != VSYNC_PASSTHROUGH
+#if FFMPEG_OPT_VSYNC_DROP
+        && fps->vsync_method != VSYNC_DROP
+#endif
+        ) {
         if (delta0 < -0.6) {
             av_log(ofp, AV_LOG_VERBOSE, "Past duration %f too large\n", -delta0);
         } else
@@ -2558,7 +2569,6 @@ static void video_sync_process(OutputFilterPriv *ofp, AVFrame *frame,
             delta0 = 0;
             ofp->next_pts = llrint(sync_ipts);
         }
-        av_fallthrough;
     case VSYNC_CFR:
         // FIXME set to 0.5 after we fix some dts/pts bugs like in avidec.c
         if (frame_drop_threshold && delta < frame_drop_threshold && fps->frame_number) {
@@ -2579,6 +2589,9 @@ static void video_sync_process(OutputFilterPriv *ofp, AVFrame *frame,
             ofp->next_pts = llrint(sync_ipts);
         frame->duration = llrint(duration);
         break;
+#if FFMPEG_OPT_VSYNC_DROP
+    case VSYNC_DROP:
+#endif
     case VSYNC_PASSTHROUGH:
         ofp->next_pts = llrint(sync_ipts);
         frame->duration = llrint(duration);
@@ -2823,10 +2836,8 @@ static int fg_output_step(OutputFilterPriv *ofp, FilterGraphThread *fgt,
     if (!fgt->got_frame) {
         ret = clone_side_data(&fd->side_data, &fd->nb_side_data,
                               ofp->side_data, ofp->nb_side_data, 0);
-        if (ret < 0) {
-            av_frame_unref(frame);
+        if (ret < 0)
             return ret;
-        }
     }
 
     fd->wallclock[LATENCY_PROBE_FILTER_POST] = av_gettime_relative();
@@ -2860,6 +2871,7 @@ static int read_frames(FilterGraph *fg, FilterGraphThread *fgt,
                        AVFrame *frame)
 {
     FilterGraphPriv *fgp = fgp_from_fg(fg);
+    int did_step = 0;
 
     // graph not configured, just select the input to request
     if (!fgt->graph) {
@@ -2878,7 +2890,7 @@ static int read_frames(FilterGraph *fg, FilterGraphThread *fgt,
         return AVERROR_BUG;
     }
 
-    if (fgp->nb_outputs_done < fg->nb_outputs) {
+    while (fgp->nb_outputs_done < fg->nb_outputs) {
         int ret;
 
         /* Reap all buffers present in the buffer sinks */
@@ -2893,6 +2905,9 @@ static int read_frames(FilterGraph *fg, FilterGraphThread *fgt,
             }
         }
 
+        // return after one iteration, so that scheduler can rate-control us
+        if (did_step && fgp->have_sources)
+            return 0;
 
         ret = avfilter_graph_request_oldest(fgt->graph);
         if (ret == AVERROR(EAGAIN)) {
@@ -2909,8 +2924,7 @@ static int read_frames(FilterGraph *fg, FilterGraphThread *fgt,
         }
         fgt->next_in = fg->nb_inputs;
 
-        // return so that scheduler can rate-control us
-        return 0;
+        did_step = 1;
     }
 
     return AVERROR_EOF;
@@ -3167,7 +3181,7 @@ static int send_frame(FilterGraph *fg, FilterGraphThread *fgt,
                 const char *color_space_name = av_color_space_name(frame->colorspace);
                 const char *color_range_name = av_color_range_name(frame->color_range);
                 const char *alpha_mode = av_alpha_mode_name(frame->alpha_mode);
-                av_bprintf(&reason, "video parameters changed to %s(%s, %s), %dx%d, %s alpha, ",
+                av_bprintf(&reason, "video parameters changed to %s(%s, %s), %dx%d, %s alpha,",
                         unknown_if_null(pixel_format_name), unknown_if_null(color_range_name),
                         unknown_if_null(color_space_name), frame->width, frame->height,
                         unknown_if_null(alpha_mode));
@@ -3323,6 +3337,8 @@ static int filter_thread(void *arg)
 
         o = (intptr_t)fgt.frame->opaque;
 
+        o = (intptr_t)fgt.frame->opaque;
+
         // message on the control stream
         if (input_idx == fg->nb_inputs) {
             FilterCommand *fc;

fftools/ffmpeg_mux.c

@@ -139,6 +139,11 @@ static int mux_fixup_ts(Muxer *mux, MuxStream *ms, AVPacket *pkt)
 {
     OutputStream *ost = &ms->ost;
 
+#if FFMPEG_OPT_VSYNC_DROP
+    if (ost->type == AVMEDIA_TYPE_VIDEO && ms->ts_drop)
+        pkt->pts = pkt->dts = AV_NOPTS_VALUE;
+#endif
+
     // rescale timestamps to the stream timebase
     if (ost->type == AVMEDIA_TYPE_AUDIO && !ost->enc) {
         // use av_rescale_delta() for streamcopying audio, to preserve

fftools/ffmpeg_mux.h

@@ -81,6 +81,9 @@ typedef struct MuxStream {
     int             copy_initial_nonkeyframes;
     int             copy_prior_start;
     int             streamcopy_started;
+#if FFMPEG_OPT_VSYNC_DROP
+    int             ts_drop;
+#endif
 
     AVRational      frame_rate;
     AVRational      max_frame_rate;

fftools/ffmpeg_mux_init.c

@@ -417,19 +417,45 @@ static int ost_get_filters(const OptionsContext *o, AVFormatContext *oc,
                            OutputStream *ost, char **dst)
 {
     const char *filters = NULL;
+#if FFMPEG_OPT_FILTER_SCRIPT
+    const char *filters_script = NULL;
+
+    opt_match_per_stream_str(ost, &o->filter_scripts, oc, ost->st, &filters_script);
+#endif
     opt_match_per_stream_str(ost, &o->filters, oc, ost->st, &filters);
 
     if (!ost->ist) {
-        if (filters) {
+        if (
+#if FFMPEG_OPT_FILTER_SCRIPT
+            filters_script ||
+#endif
+            filters) {
             av_log(ost, AV_LOG_ERROR,
-                   "Filtergraph '%s' was specified for a stream fed from a complex "
+                   "%s '%s' was specified for a stream fed from a complex "
                    "filtergraph. Simple and complex filtering cannot be used "
-                   "together for the same stream.\n", filters);
+                   "together for the same stream.\n",
+#if FFMPEG_OPT_FILTER_SCRIPT
+                   filters ? "Filtergraph" : "Filtergraph script",
+                   filters ? filters : filters_script
+#else
+                   "Filtergraph", filters
+#endif
+                   );
             return AVERROR(EINVAL);
         }
         return 0;
     }
 
+#if FFMPEG_OPT_FILTER_SCRIPT
+    if (filters_script && filters) {
+        av_log(ost, AV_LOG_ERROR, "Both -filter and -filter_script set\n");
+        return AVERROR(EINVAL);
+    }
+
+    if (filters_script)
+        *dst = read_file_to_string(filters_script);
+    else
+#endif
     if (filters)
         *dst = av_strdup(filters);
     else
@@ -735,10 +761,21 @@ static int new_stream_video(Muxer *mux, const OptionsContext *o,
 
         opt_match_per_stream_int(ost, &o->force_fps, oc, st, &ms->force_fps);
 
+#if FFMPEG_OPT_TOP
+        ost->top_field_first = -1;
+        opt_match_per_stream_int(ost, &o->top_field_first, oc, st, &ost->top_field_first);
+        if (ost->top_field_first >= 0)
+            av_log(ost, AV_LOG_WARNING, "-top is deprecated, use the setfield filter instead\n");
+#endif
+
+#if FFMPEG_OPT_VSYNC
+        *vsync_method = video_sync_method;
+#else
         *vsync_method = VSYNC_AUTO;
+#endif
         opt_match_per_stream_str(ost, &o->fps_mode, oc, st, &fps_mode);
         if (fps_mode) {
-            ret = parse_and_set_vsync(fps_mode, vsync_method, ost->file->index, ost->index);
+            ret = parse_and_set_vsync(fps_mode, vsync_method, ost->file->index, ost->index, 0);
             if (ret < 0)
                 return ret;
         }
@@ -773,6 +810,10 @@ static int new_stream_video(Muxer *mux, const OptionsContext *o,
                 *vsync_method = VSYNC_VSCFR;
             }
         }
+#if FFMPEG_OPT_VSYNC_DROP
+        if (*vsync_method == VSYNC_DROP)
+            ms->ts_drop = 1;
+#endif
     }
 
     return 0;
@@ -999,12 +1040,28 @@ static int streamcopy_init(const OptionsContext *o, const Muxer *mux,
     int ret = 0;
 
     const char *filters = NULL;
+#if FFMPEG_OPT_FILTER_SCRIPT
+    const char *filters_script = NULL;
+
+    opt_match_per_stream_str(ost, &o->filter_scripts, mux->fc, ost->st, &filters_script);
+#endif
     opt_match_per_stream_str(ost, &o->filters, mux->fc, ost->st, &filters);
 
-    if (filters) {
+    if (
+#if FFMPEG_OPT_FILTER_SCRIPT
+        filters_script ||
+#endif
+        filters) {
         av_log(ost, AV_LOG_ERROR,
-               "Filtergraph '%s' was specified, but codec copy was selected. "
-               "Filtering and streamcopy cannot be used together.\n", filters);
+               "%s '%s' was specified, but codec copy was selected. "
+               "Filtering and streamcopy cannot be used together.\n",
+#if FFMPEG_OPT_FILTER_SCRIPT
+               filters ? "Filtergraph" : "Filtergraph script",
+               filters ? filters : filters_script
+#else
+               "Filtergraph", filters
+#endif
+               );
         return AVERROR(EINVAL);
     }
 
@@ -1344,11 +1401,20 @@ static int ost_add(Muxer *mux, const OptionsContext *o, enum AVMediaType type,
                 q = (AVRational){ ENC_TIME_BASE_FILTER, 0 };
             } else {
                 ret = av_parse_ratio(&q, enc_time_base, INT_MAX, 0, NULL);
-                if (ret < 0 || q.den <= 0 || q.num < 0) {
+                if (ret < 0 || q.den <= 0
+#if !FFMPEG_OPT_ENC_TIME_BASE_NUM
+                    || q.num < 0
+#endif
+                    ) {
                     av_log(ost, AV_LOG_FATAL, "Invalid time base: %s\n", enc_time_base);
                     ret = ret < 0 ? ret : AVERROR(EINVAL);
                     goto fail;
                 }
+#if FFMPEG_OPT_ENC_TIME_BASE_NUM
+                if (q.num < 0)
+                    av_log(ost, AV_LOG_WARNING, "-enc_time_base -1 is deprecated,"
+                           " use -enc_time_base demux\n");
+#endif
             }
 
             enc_tb = q;
@@ -1465,8 +1531,6 @@ static int ost_add(Muxer *mux, const OptionsContext *o, enum AVMediaType type,
 
     if (oc->oformat->flags & AVFMT_GLOBALHEADER && ost->enc)
         ost->enc->enc_ctx->flags |= AV_CODEC_FLAG_GLOBAL_HEADER;
-    if (oc->oformat->flags & AVFMT_FIXED_FRAMESIZE && ost->enc)
-        ost->enc->enc_ctx->flags2 |= AV_CODEC_FLAG2_FIXED_FRAME_SIZE;
 
     opt_match_per_stream_int(ost, &o->copy_initial_nonkeyframes,
                              oc, st, &ms->copy_initial_nonkeyframes);
@@ -2063,8 +2127,7 @@ static int setup_sync_queues(Muxer *mux, AVFormatContext *oc,
         nb_interleaved += IS_INTERLEAVED(type);
         nb_av_enc      += IS_AV_ENC(ost, type);
         nb_audio_fs    += (ost->enc && type == AVMEDIA_TYPE_AUDIO &&
-                           (!(ost->enc->enc_ctx->codec->capabilities & AV_CODEC_CAP_VARIABLE_FRAME_SIZE) ||
-                            (ost->enc->enc_ctx->flags2 & AV_CODEC_FLAG2_FIXED_FRAME_SIZE)));
+                           !(ost->enc->enc_ctx->codec->capabilities & AV_CODEC_CAP_VARIABLE_FRAME_SIZE));
 
         limit_frames        |=  ms->max_frames < INT64_MAX;
         limit_frames_av_enc |= (ms->max_frames < INT64_MAX) && IS_AV_ENC(ost, type);
@@ -2489,10 +2552,6 @@ static int of_map_group(Muxer *mux, AVDictionary **dict, AVBPrint *bp, const cha
         }
         break;
     }
-    case AV_STREAM_GROUP_PARAMS_LCEVC:
-    case AV_STREAM_GROUP_PARAMS_TREF:
-    case AV_STREAM_GROUP_PARAMS_DOLBY_VISION:
-        break;
     default:
         av_log(mux, AV_LOG_ERROR, "Unsupported mapped group type %d.\n", stg->type);
         ret = AVERROR(EINVAL);
@@ -2515,10 +2574,6 @@ static int of_parse_group_token(Muxer *mux, const char *token, char *ptr)
                 { .i64 = AV_STREAM_GROUP_PARAMS_IAMF_AUDIO_ELEMENT },    .unit = "type" },
             { "iamf_mix_presentation", NULL, 0, AV_OPT_TYPE_CONST,
                 { .i64 = AV_STREAM_GROUP_PARAMS_IAMF_MIX_PRESENTATION }, .unit = "type" },
-            { "lcevc", NULL, 0, AV_OPT_TYPE_CONST,
-                { .i64 = AV_STREAM_GROUP_PARAMS_LCEVC }, .unit = "type" },
-            { "tref", NULL, 0, AV_OPT_TYPE_CONST,
-                { .i64 = AV_STREAM_GROUP_PARAMS_TREF }, .unit = "type" },
         { NULL },
     };
     const AVClass class = {
@@ -2593,10 +2648,6 @@ static int of_parse_group_token(Muxer *mux, const char *token, char *ptr)
         ret = avformat_stream_group_add_stream(stg, oc->streams[idx]);
         if (ret < 0)
             goto end;
-        OutputStream *ost = mux->of.streams[idx];
-        if (ost->enc && (type == AV_STREAM_GROUP_PARAMS_IAMF_AUDIO_ELEMENT ||
-                         type == AV_STREAM_GROUP_PARAMS_IAMF_MIX_PRESENTATION))
-            ost->enc->enc_ctx->flags2 |= AV_CODEC_FLAG2_FIXED_FRAME_SIZE;
     }
     while (e = av_dict_get(dict, "stg", e, 0)) {
         char *endptr;
@@ -2621,6 +2672,8 @@ static int of_parse_group_token(Muxer *mux, const char *token, char *ptr)
         ret = of_parse_iamf_submixes(mux, stg, ptr);
         break;
     default:
+        av_log(mux, AV_LOG_FATAL, "Unknown group type %d.\n", type);
+        ret = AVERROR(EINVAL);
         break;
     }
 
@@ -3227,6 +3280,12 @@ static int process_forced_keyframes(Muxer *mux, const OptionsContext *o)
             // parse it only for static kf timings
         } else if (!strcmp(forced_keyframes, "source")) {
             ost->kf.type = KF_FORCE_SOURCE;
+#if FFMPEG_OPT_FORCE_KF_SOURCE_NO_DROP
+        } else if (!strcmp(forced_keyframes, "source_no_drop")) {
+            av_log(ost, AV_LOG_WARNING, "The 'source_no_drop' value for "
+                   "-force_key_frames is deprecated, use just 'source'\n");
+            ost->kf.type = KF_FORCE_SOURCE;
+#endif
         } else if (!strcmp(forced_keyframes, "scd_metadata")) {
             ost->kf.type = KF_FORCE_SCD_METADATA;
         } else {

fftools/ffmpeg_opt.c

@@ -56,6 +56,9 @@ char *vstats_filename;
 float dts_delta_threshold   = 10;
 float dts_error_threshold   = 3600*30;
 
+#if FFMPEG_OPT_VSYNC
+enum VideoSyncMethod video_sync_method = VSYNC_AUTO;
+#endif
 float frame_drop_threshold = 0;
 int do_benchmark      = 0;
 int do_benchmark_all  = 0;
@@ -356,17 +359,38 @@ int view_specifier_parse(const char **pspec, ViewSpecifier *vs)
     return 0;
 }
 
-int parse_and_set_vsync(const char *arg, enum VideoSyncMethod *vsync_var, int file_idx, int st_idx)
+int parse_and_set_vsync(const char *arg, enum VideoSyncMethod *vsync_var, int file_idx, int st_idx, int is_global)
 {
     if      (!av_strcasecmp(arg, "cfr"))         *vsync_var = VSYNC_CFR;
     else if (!av_strcasecmp(arg, "vfr"))         *vsync_var = VSYNC_VFR;
     else if (!av_strcasecmp(arg, "passthrough")) *vsync_var = VSYNC_PASSTHROUGH;
-    else if (!av_strcasecmp(arg, "auto"))        *vsync_var = VSYNC_AUTO;
-    else {
+#if FFMPEG_OPT_VSYNC_DROP
+    else if (!av_strcasecmp(arg, "drop")) {
+        av_log(NULL, AV_LOG_WARNING, "-vsync/fps_mode drop is deprecated\n");
+        *vsync_var = VSYNC_DROP;
+    }
+#endif
+    else if (!is_global && !av_strcasecmp(arg, "auto"))  *vsync_var = VSYNC_AUTO;
+    else if (!is_global) {
         av_log(NULL, AV_LOG_FATAL, "Invalid value %s specified for fps_mode of #%d:%d.\n", arg, file_idx, st_idx);
         return AVERROR(EINVAL);
     }
 
+#if FFMPEG_OPT_VSYNC
+    if (is_global && *vsync_var == VSYNC_AUTO) {
+        int ret;
+        double num;
+
+        ret = parse_number("vsync", arg, OPT_TYPE_INT, VSYNC_AUTO, VSYNC_VFR, &num);
+        if (ret < 0)
+            return ret;
+
+        video_sync_method = num;
+        av_log(NULL, AV_LOG_WARNING, "Passing a number to -vsync is deprecated,"
+               " use a string argument as described in the manual.\n");
+    }
+#endif
+
     return 0;
 }
 
@@ -621,7 +645,6 @@ static int opt_map(void *optctx, const char *opt, const char *arg)
             for (i = 0; i < o->nb_stream_maps; i++) {
                 m = &o->stream_maps[i];
                 if (file_idx == m->file_index &&
-                    !m->linklabel &&
                     m->stream_index >= 0 &&
                     m->stream_index < input_files[m->file_index]->nb_streams &&
                     stream_specifier_match(&ss,
@@ -1210,6 +1233,14 @@ static int opt_audio_filters(void *optctx, const char *opt, const char *arg)
     return parse_option(o, "filter:a", arg, options);
 }
 
+#if FFMPEG_OPT_VSYNC
+static int opt_vsync(void *optctx, const char *opt, const char *arg)
+{
+    av_log(NULL, AV_LOG_WARNING, "-vsync is deprecated. Use -fps_mode\n");
+    return parse_and_set_vsync(arg, &video_sync_method, -1, -1, 1);
+}
+#endif
+
 static int opt_timecode(void *optctx, const char *opt, const char *arg)
 {
     OptionsContext *o = optctx;
@@ -1250,6 +1281,31 @@ static int opt_filter_complex(void *optctx, const char *opt, const char *arg)
     return 0;
 }
 
+#if FFMPEG_OPT_FILTER_SCRIPT
+static int opt_filter_complex_script(void *optctx, const char *opt, const char *arg)
+{
+    GlobalOptionsContext *go = optctx;
+    char *graph_desc;
+    int ret;
+
+    graph_desc = read_file_to_string(arg);
+    if (!graph_desc)
+        return AVERROR(EINVAL);
+
+    av_log(NULL, AV_LOG_WARNING, "-%s is deprecated, use -/filter_complex %s instead\n",
+           opt, arg);
+
+    ret = GROW_ARRAY(go->filtergraphs, go->nb_filtergraphs);
+    if (ret < 0) {
+        av_freep(&graph_desc);
+        return ret;
+    }
+    go->filtergraphs[go->nb_filtergraphs - 1] = graph_desc;
+
+    return 0;
+}
+#endif
+
 void show_help_default(const char *opt, const char *arg)
 {
     int show_advanced = 0, show_avoptions = 0;
@@ -1540,6 +1596,22 @@ int opt_timelimit(void *optctx, const char *opt, const char *arg)
     return 0;
 }
 
+#if FFMPEG_OPT_QPHIST
+static int opt_qphist(void *optctx, const char *opt, const char *arg)
+{
+    av_log(NULL, AV_LOG_WARNING, "Option -%s is deprecated and has no effect\n", opt);
+    return 0;
+}
+#endif
+
+#if FFMPEG_OPT_ADRIFT_THRESHOLD
+static int opt_adrift_threshold(void *optctx, const char *opt, const char *arg)
+{
+    av_log(NULL, AV_LOG_WARNING, "Option -%s is deprecated and has no effect\n", opt);
+    return 0;
+}
+#endif
+
 static const char *const alt_channel_layout[] = { "ch_layout", NULL};
 static const char *const alt_codec[]          = { "c", "acodec", "vcodec", "scodec", "dcodec", NULL };
 static const char *const alt_filter[]         = { "af", "vf", NULL };
@@ -1746,6 +1818,11 @@ const OptionDef options[] = {
     { "filter_buffered_frames", OPT_TYPE_INT, OPT_EXPERT,
         { &filter_buffered_frames },
         "maximum number of buffered frames in a filter graph" },
+#if FFMPEG_OPT_FILTER_SCRIPT
+    { "filter_script",          OPT_TYPE_STRING, OPT_PERSTREAM | OPT_EXPERT | OPT_OUTPUT,
+        { .off = OFFSET(filter_scripts) },
+        "deprecated, use -/filter", "filename" },
+#endif
     { "reinit_filter",          OPT_TYPE_INT, OPT_PERSTREAM | OPT_INPUT | OPT_EXPERT,
         { .off = OFFSET(reinit_filters) },
         "reinit filtergraph on input parameter changes", "" },
@@ -1761,6 +1838,11 @@ const OptionDef options[] = {
     { "lavfi",               OPT_TYPE_FUNC, OPT_FUNC_ARG | OPT_EXPERT,
         { .func_arg = opt_filter_complex },
         "create a complex filtergraph", "graph_description" },
+#if FFMPEG_OPT_FILTER_SCRIPT
+    { "filter_complex_script", OPT_TYPE_FUNC, OPT_FUNC_ARG | OPT_EXPERT,
+        { .func_arg = opt_filter_complex_script },
+        "deprecated, use -/filter_complex instead", "filename" },
+#endif
     { "print_graphs",   OPT_TYPE_BOOL, 0,
         { &print_graphs },
         "print execution graph data to stderr" },
@@ -1860,12 +1942,6 @@ const OptionDef options[] = {
         { .off = OFFSET(display_vflips) },
         "set display vertical flip for stream(s) "
         "(overrides any display rotation if it is not set)"},
-    { "mastering_display",          OPT_TYPE_STRING, OPT_VIDEO | OPT_PERSTREAM | OPT_INPUT | OPT_EXPERT,
-        { .off = OFFSET(mastering_displays) },
-        "set SMPTE2084 mastering display color volume info" },
-    { "content_light",              OPT_TYPE_STRING, OPT_VIDEO | OPT_PERSTREAM | OPT_INPUT | OPT_EXPERT,
-        { .off = OFFSET(content_lights) },
-        "set SMPTE2084 Max CLL and Max FALL values" },
     { "vn",                         OPT_TYPE_BOOL,   OPT_VIDEO | OPT_OFFSET | OPT_INPUT | OPT_OUTPUT,
         { .off = OFFSET(video_disable) },
         "disable video" },
@@ -1913,7 +1989,7 @@ const OptionDef options[] = {
         .u1.name_canon = "tag", },
     { "fps_mode",                   OPT_TYPE_STRING, OPT_VIDEO | OPT_EXPERT | OPT_PERSTREAM | OPT_OUTPUT,
         { .off = OFFSET(fps_mode) },
-        "set framerate mode for matching video streams" },
+        "set framerate mode for matching video streams; overrides vsync" },
     { "force_fps",                  OPT_TYPE_BOOL,   OPT_VIDEO | OPT_EXPERT  | OPT_PERSTREAM | OPT_OUTPUT,
         { .off = OFFSET(force_fps) },
         "force the selected framerate, disable the best supported framerate selection" },
@@ -2098,5 +2174,27 @@ const OptionDef options[] = {
         { .func_arg = opt_filter_hw_device },
         "set hardware device used when filtering", "device" },
 
+    // deprecated options
+#if FFMPEG_OPT_ADRIFT_THRESHOLD
+    { "adrift_threshold", OPT_TYPE_FUNC, OPT_FUNC_ARG | OPT_EXPERT,
+        { .func_arg = opt_adrift_threshold },
+        "deprecated, does nothing", "threshold" },
+#endif
+#if FFMPEG_OPT_TOP
+    { "top", OPT_TYPE_INT,     OPT_VIDEO | OPT_EXPERT | OPT_PERSTREAM | OPT_INPUT | OPT_OUTPUT,
+        { .off = OFFSET(top_field_first) },
+        "deprecated, use the setfield video filter", "" },
+#endif
+#if FFMPEG_OPT_QPHIST
+    { "qphist", OPT_TYPE_FUNC, OPT_VIDEO | OPT_EXPERT,
+        { .func_arg = opt_qphist },
+        "deprecated, does nothing" },
+#endif
+#if FFMPEG_OPT_VSYNC
+    { "vsync",                  OPT_TYPE_FUNC, OPT_FUNC_ARG | OPT_EXPERT,
+        { .func_arg = opt_vsync },
+        "set video sync method globally; deprecated, use -fps_mode", "" },
+#endif
+
     { NULL, },
 };

fftools/ffmpeg_sched.c

@@ -32,7 +32,6 @@
 #include "libavcodec/packet.h"
 
 #include "libavutil/avassert.h"
-#include "libavutil/container_fifo.h"
 #include "libavutil/error.h"
 #include "libavutil/fifo.h"
 #include "libavutil/frame.h"
@@ -87,7 +86,6 @@ typedef struct SchDec {
     unsigned         nb_outputs;
 
     SchTask             task;
-    SchWaiter           waiter;
     // Queue for receiving input packets, one stream.
     ThreadQueue        *queue;
 
@@ -97,9 +95,6 @@ typedef struct SchDec {
 
     // temporary storage used by sch_dec_send()
     AVFrame            *send_frame;
-
-    // internal queue of undecoded packets used by sch_dec_receive()
-    AVContainerFifo    *overflow;
 } SchDec;
 
 typedef struct SchSyncQueue {
@@ -440,7 +435,7 @@ static void task_init(Scheduler *sch, SchTask *task, enum SchedulerNodeType type
     task->func_arg  = func_arg;
 }
 
-static int64_t trailing_dts(const Scheduler *sch)
+static int64_t trailing_dts(const Scheduler *sch, int count_finished)
 {
     int64_t min_dts = INT64_MAX;
 
@@ -450,7 +445,7 @@ static int64_t trailing_dts(const Scheduler *sch)
         for (unsigned j = 0; j < mux->nb_streams; j++) {
             const SchMuxStream *ms = &mux->streams[j];
 
-            if (ms->source_finished)
+            if (ms->source_finished && !count_finished)
                 continue;
             if (ms->last_dts == AV_NOPTS_VALUE)
                 return AV_NOPTS_VALUE;
@@ -462,26 +457,6 @@ static int64_t trailing_dts(const Scheduler *sch)
     return min_dts == INT64_MAX ? AV_NOPTS_VALUE : min_dts;
 }
 
-static int64_t progressing_dts(const Scheduler *sch, int count_finished)
-{
-    int64_t max_dts = INT64_MIN;
-
-    for (unsigned i = 0; i < sch->nb_mux; i++) {
-        const SchMux *mux = &sch->mux[i];
-
-        for (unsigned j = 0; j < mux->nb_streams; j++) {
-            const SchMuxStream *ms = &mux->streams[j];
-
-            if (ms->source_finished && !count_finished)
-                continue;
-            if (ms->last_dts != AV_NOPTS_VALUE)
-                max_dts = FFMAX(max_dts, ms->last_dts);
-        }
-    }
-
-    return max_dts == INT64_MIN ? AV_NOPTS_VALUE : max_dts;
-}
-
 void sch_remove_filtergraph(Scheduler *sch, int idx)
 {
     SchFilterGraph *fg = &sch->filters[idx];
@@ -553,7 +528,6 @@ void sch_free(Scheduler **psch)
         tq_free(&dec->queue);
 
         av_thread_message_queue_free(&dec->queue_end_ts);
-        av_container_fifo_free(&dec->overflow);
 
         for (unsigned j = 0; j < dec->nb_outputs; j++) {
             SchDecOutput *o = &dec->outputs[j];
@@ -565,8 +539,6 @@ void sch_free(Scheduler **psch)
         av_freep(&dec->outputs);
 
         av_frame_free(&dec->send_frame);
-
-        waiter_uninit(&dec->waiter);
     }
     av_freep(&sch->dec);
 
@@ -817,14 +789,6 @@ int sch_add_dec(Scheduler *sch, SchThreadFunc func, void *ctx, int send_end_ts)
             return ret;
     }
 
-    dec->overflow = av_container_fifo_alloc_avpacket(0);
-    if (!dec->overflow)
-        return AVERROR(ENOMEM);
-
-    ret = waiter_init(&dec->waiter);
-    if (ret < 0)
-        return ret;
-
     return idx;
 }
 
@@ -1318,15 +1282,7 @@ int sch_mux_sub_heartbeat_add(Scheduler *sch, unsigned mux_idx, unsigned stream_
     return 0;
 }
 
-enum {
-    UNCHOKE_DEMUX  = (1 << 0),
-    UNCHOKE_FILTER = (1 << 1),
-    UNCHOKE_DECODE = (1 << 2),
-
-    UNCHOKE_ALL = UNCHOKE_DEMUX | UNCHOKE_FILTER | UNCHOKE_DECODE,
-};
-
-static void unchoke_for_stream(Scheduler *sch, SchedulerNode src, int flags);
+static void unchoke_for_stream(Scheduler *sch, SchedulerNode src);
 
 // Unchoke any filter graphs that are downstream of this node, to prevent it
 // from getting stuck trying to push data to a full queue
@@ -1338,10 +1294,8 @@ static void unchoke_downstream(Scheduler *sch, SchedulerNode *dst)
     switch (dst->type) {
     case SCH_NODE_TYPE_DEC:
         dec = &sch->dec[dst->idx];
-        if (!dec->waiter.choked_next) {
-            for (int i = 0; i < dec->nb_outputs; i++)
-                unchoke_downstream(sch, dec->outputs[i].dst);
-        }
+        for (int i = 0; i < dec->nb_outputs; i++)
+            unchoke_downstream(sch, dec->outputs[i].dst);
         break;
     case SCH_NODE_TYPE_ENC:
         enc = &sch->enc[dst->idx];
@@ -1357,8 +1311,8 @@ static void unchoke_downstream(Scheduler *sch, SchedulerNode *dst)
             fg->waiter.choked_next = 0;
         } else {
             // ensure that this filter graph is not stuck waiting for
-            // input from a different upstream source
-            unchoke_for_stream(sch, fg->inputs[fg->best_input].src, UNCHOKE_ALL);
+            // input from a different upstream demuxer
+            unchoke_for_stream(sch, fg->inputs[fg->best_input].src);
         }
         break;
     default:
@@ -1367,30 +1321,23 @@ static void unchoke_downstream(Scheduler *sch, SchedulerNode *dst)
     }
 }
 
-static void unchoke_for_stream(Scheduler *sch, SchedulerNode src, int flags)
+static void unchoke_for_stream(Scheduler *sch, SchedulerNode src)
 {
     while (1) {
         SchFilterGraph *fg;
         SchDemux *demux;
-        SchDec *dec;
         switch (src.type) {
         case SCH_NODE_TYPE_DEMUX:
             // fed directly by a demuxer (i.e. not through a filtergraph)
             demux = &sch->demux[src.idx];
             if (demux->waiter.choked_next == 0)
                 return; // prevent infinite loop
-            if (flags & UNCHOKE_DEMUX) {
-                demux->waiter.choked_next = 0;
-                for (int i = 0; i < demux->nb_streams; i++)
-                    unchoke_downstream(sch, demux->streams[i].dst);
-            }
+            demux->waiter.choked_next = 0;
+            for (int i = 0; i < demux->nb_streams; i++)
+                unchoke_downstream(sch, demux->streams[i].dst);
             return;
         case SCH_NODE_TYPE_DEC:
-            dec = &sch->dec[src.idx];
-            if (!(flags & UNCHOKE_DECODE))
-                return;
-            dec->waiter.choked_next = 0;
-            src = dec->src;
+            src = sch->dec[src.idx].src;
             continue;
         case SCH_NODE_TYPE_ENC:
             src = sch->enc[src.idx].src;
@@ -1400,8 +1347,7 @@ static void unchoke_for_stream(Scheduler *sch, SchedulerNode src, int flags)
             // the filtergraph contains internal sources and
             // requested to be scheduled directly
             if (fg->best_input == fg->nb_inputs) {
-                if (flags & UNCHOKE_FILTER)
-                    fg->waiter.choked_next = 0;
+                fg->waiter.choked_next = 0;
                 return;
             }
             src = fg->inputs[fg->best_input].src;
@@ -1453,23 +1399,17 @@ static void schedule_update_locked(Scheduler *sch)
     if (atomic_load(&sch->terminate))
         return;
 
-    dts = trailing_dts(sch);
+    dts = trailing_dts(sch, 0);
 
-    atomic_store(&sch->last_dts, progressing_dts(sch, 0));
+    atomic_store(&sch->last_dts, dts);
 
     // initialize our internal state
-#define RESET_WAITER(field)                                                    \
-    do {                                                                       \
-        for (unsigned i = 0; i < sch->nb_##field; i++) {                       \
-            SchWaiter *w   = &sch->field[i].waiter;                            \
-            w->choked_prev = atomic_load(&w->choked);                          \
-            w->choked_next = 1;                                                \
-        }                                                                      \
-    } while (0)
-
-    RESET_WAITER(demux);
-    RESET_WAITER(filters);
-    RESET_WAITER(dec);
+    for (unsigned type = 0; type < 2; type++)
+        for (unsigned i = 0; i < (type ? sch->nb_filters : sch->nb_demux); i++) {
+            SchWaiter *w = type ? &sch->filters[i].waiter : &sch->demux[i].waiter;
+            w->choked_prev = atomic_load(&w->choked);
+            w->choked_next = 1;
+        }
 
     // figure out the sources that are allowed to proceed
     for (unsigned i = 0; i < sch->nb_mux; i++) {
@@ -1480,18 +1420,15 @@ static void schedule_update_locked(Scheduler *sch)
 
             // unblock sources for output streams that are not finished
             // and not too far ahead of the trailing stream
-            if (ms->source_finished) {
-                // still allow decoders to drain
-                unchoke_for_stream(sch, ms->src, UNCHOKE_DECODE);
+            if (ms->source_finished)
                 continue;
-            }
             if (dts == AV_NOPTS_VALUE && ms->last_dts != AV_NOPTS_VALUE)
                 continue;
             if (dts != AV_NOPTS_VALUE && ms->last_dts - dts >= SCHEDULE_TOLERANCE)
                 continue;
 
             // resolve the source to unchoke
-            unchoke_for_stream(sch, ms->src, UNCHOKE_ALL);
+            unchoke_for_stream(sch, ms->src);
             have_unchoked = 1;
         }
     }
@@ -1504,41 +1441,33 @@ static void schedule_update_locked(Scheduler *sch)
         for (unsigned j = 0; j < fg->nb_inputs; j++) {
             SchFilterIn *fi = &fg->inputs[j];
             if (fi->receive_finished && !fi->send_finished)
-                unchoke_for_stream(sch, fi->src, UNCHOKE_ALL);
+                unchoke_for_stream(sch, fi->src);
         }
     }
 
     // make sure to unchoke at least one source, if still available
-#define UNCHOKE_ONCE(field)                                                    \
-    do {                                                                       \
-        for (unsigned i = 0; !have_unchoked && i < sch->nb_##field; i++) {     \
-            SchWaiter *w = &sch->field[i].waiter;                              \
-            if (!sch->field[i].task_exited) {                                  \
-                w->choked_next = 0;                                            \
-                have_unchoked  = 1;                                            \
-                break;                                                         \
-            }                                                                  \
-        }                                                                      \
-    } while (0)
+    for (unsigned type = 0; !have_unchoked && type < 2; type++)
+        for (unsigned i = 0; i < (type ? sch->nb_filters : sch->nb_demux); i++) {
+            int exited = type ? sch->filters[i].task_exited : sch->demux[i].task_exited;
+            SchWaiter *w = type ? &sch->filters[i].waiter : &sch->demux[i].waiter;
+            if (!exited) {
+                w->choked_next = 0;
+                have_unchoked  = 1;
+                break;
+            }
+        }
 
-    UNCHOKE_ONCE(demux);
-    UNCHOKE_ONCE(filters);
+    for (unsigned type = 0; type < 2; type++) {
+        for (unsigned i = 0; i < (type ? sch->nb_filters : sch->nb_demux); i++) {
+            SchWaiter *w = type ? &sch->filters[i].waiter : &sch->demux[i].waiter;
+            if (w->choked_prev != w->choked_next) {
+                waiter_set(w, w->choked_next);
+                if (!type)
+                    choke_demux(sch, i, w->choked_next);
+            }
+        }
+    }
 
-#define UPDATE_WAITER(field)                                                   \
-    do {                                                                       \
-        for (unsigned i = 0; i < sch->nb_##field; i++) {                       \
-            SchWaiter *w = &sch->field[i].waiter;                              \
-            if (w->choked_prev != w->choked_next) {                            \
-                waiter_set(w, w->choked_next);                                 \
-                if (offsetof(Scheduler, field) == offsetof(Scheduler, demux))  \
-                    choke_demux(sch, i, w->choked_next);                       \
-            }                                                                  \
-        }                                                                      \
-    } while (0)
-
-    UPDATE_WAITER(demux);
-    UPDATE_WAITER(filters);
-    UPDATE_WAITER(dec);
 }
 
 enum {
@@ -2257,7 +2186,7 @@ int sch_mux_receive(Scheduler *sch, unsigned mux_idx, AVPacket *pkt)
     av_assert0(mux_idx < sch->nb_mux);
     mux = &sch->mux[mux_idx];
 
-    ret = tq_receive(mux->queue, &stream_idx, pkt, 0);
+    ret = tq_receive(mux->queue, &stream_idx, pkt);
     pkt->stream_index = stream_idx;
     return ret;
 }
@@ -2341,12 +2270,6 @@ int sch_dec_receive(Scheduler *sch, unsigned dec_idx, AVPacket *pkt)
     av_assert0(dec_idx < sch->nb_dec);
     dec = &sch->dec[dec_idx];
 
-retry:
-    // Pull a packet from the overflow FIFO while unchoked or expecting EOF ts
-    if (av_container_fifo_can_read(dec->overflow) &&
-        (!atomic_load(&dec->waiter.choked) || dec->expect_end_ts))
-        return av_container_fifo_read(dec->overflow, pkt, 0);
-
     // the decoder should have given us post-flush end timestamp in pkt
     if (dec->expect_end_ts) {
         Timestamp ts = (Timestamp){ .ts = pkt->pts, .tb = pkt->time_base };
@@ -2357,32 +2280,14 @@ retry:
         dec->expect_end_ts = 0;
     }
 
-    ret = tq_receive(dec->queue, &dummy, pkt, 0);
+    ret = tq_receive(dec->queue, &dummy, pkt);
     av_assert0(dummy <= 0);
 
-    // drain packets from overflow queue before returning EOF
-    if (ret == AVERROR_EOF && av_container_fifo_can_read(dec->overflow)) {
-        int terminate = waiter_wait(sch, &dec->waiter);
-        if (terminate)
-            return ret;
-        return av_container_fifo_read(dec->overflow, pkt, 0);
-    } else if (ret < 0)
-        return ret;
-
     // got a flush packet, on the next call to this function the decoder
-    // should give us post-flush end timestamp (after draining overflow fifo)
-    if (!pkt->data && !pkt->side_data_elems && dec->queue_end_ts)
+    // will give us post-flush end timestamp
+    if (ret >= 0 && !pkt->data && !pkt->side_data_elems && dec->queue_end_ts)
         dec->expect_end_ts = 1;
 
-    // we got a packet, but we're currently choked or have existing overflow
-    // packets; so push it to the FIFO first
-    if (atomic_load(&dec->waiter.choked) || av_container_fifo_can_read(dec->overflow)) {
-        ret = av_container_fifo_write(dec->overflow, pkt, 0);
-        if (ret < 0)
-            return ret;
-        goto retry;
-    }
-
     return ret;
 }
 
@@ -2516,7 +2421,7 @@ int sch_enc_receive(Scheduler *sch, unsigned enc_idx, AVFrame *frame)
     av_assert0(enc_idx < sch->nb_enc);
     enc = &sch->enc[enc_idx];
 
-    ret = tq_receive(enc->queue, &dummy, frame, 0);
+    ret = tq_receive(enc->queue, &dummy, frame);
     av_assert0(dummy <= 0);
 
     return ret;
@@ -2605,7 +2510,6 @@ int sch_filter_receive(Scheduler *sch, unsigned fg_idx,
                        unsigned *in_idx, AVFrame *frame)
 {
     SchFilterGraph *fg;
-    int ret, idx;
 
     av_assert0(fg_idx < sch->nb_filters);
     fg = &sch->filters[fg_idx];
@@ -2626,20 +2530,14 @@ int sch_filter_receive(Scheduler *sch, unsigned fg_idx,
     }
 
     if (*in_idx == fg->nb_inputs) {
-        // drain incoming frames before waiting, to avoid blocking downstream
-        ret = tq_receive(fg->queue, &idx, frame, THREAD_QUEUE_FLAG_NO_BLOCK);
-        if (ret >= 0) {
-            av_assert0(idx >= 0);
-            *in_idx = idx;
-            return 0;
-        }
-
         int terminate = waiter_wait(sch, &fg->waiter);
         return terminate ? AVERROR_EOF : AVERROR(EAGAIN);
     }
 
     while (1) {
-        ret = tq_receive(fg->queue, &idx, frame, 0);
+        int ret, idx;
+
+        ret = tq_receive(fg->queue, &idx, frame);
         if (idx < 0)
             return AVERROR_EOF;
         else if (ret >= 0) {
@@ -2826,10 +2724,6 @@ int sch_stop(Scheduler *sch, int64_t *finish_ts)
 
     atomic_store(&sch->terminate, 1);
 
-    // Ensure no other thread is currently in schedule_update_locked while
-    // we are choking all demuxers
-    pthread_mutex_lock(&sch->schedule_lock);
-
     for (unsigned type = 0; type < 2; type++)
         for (unsigned i = 0; i < (type ? sch->nb_demux : sch->nb_filters); i++) {
             SchWaiter *w = type ? &sch->demux[i].waiter : &sch->filters[i].waiter;
@@ -2838,11 +2732,6 @@ int sch_stop(Scheduler *sch, int64_t *finish_ts)
                 choke_demux(sch, i, 0); // unfreeze to allow draining
         }
 
-    for (unsigned i = 0; i < sch->nb_dec; i++)
-        waiter_set(&sch->dec[i].waiter, 0); // unfreeze to allow draining
-
-    pthread_mutex_unlock(&sch->schedule_lock);
-
     for (unsigned i = 0; i < sch->nb_demux; i++) {
         SchDemux *d = &sch->demux[i];
 
@@ -2879,7 +2768,7 @@ int sch_stop(Scheduler *sch, int64_t *finish_ts)
     }
 
     if (finish_ts)
-        *finish_ts = progressing_dts(sch, 1);
+        *finish_ts = trailing_dts(sch, 1);
 
     sch->state = SCH_STATE_STOPPED;
 

fftools/ffplay.c

@@ -3007,7 +3007,6 @@ static int read_thread(void *arg)
         // initial metadata as update.
         st->event_flags &= ~AVSTREAM_EVENT_FLAG_METADATA_UPDATED;
     }
-    ic->event_flags &= ~AVFMT_EVENT_FLAG_METADATA_UPDATED;
     for (i = 0; i < AVMEDIA_TYPE_NB; i++) {
         if (wanted_stream_spec[i] && st_index[i] == -1) {
             av_log(NULL, AV_LOG_ERROR, "Stream specifier %s does not match any %s stream\n", wanted_stream_spec[i], av_get_media_type_string(i));
@@ -3176,24 +3175,16 @@ static int read_thread(void *arg)
             is->eof = 0;
         }
 
-        if (show_status) {
-            if (ic->event_flags & AVFMT_EVENT_FLAG_METADATA_UPDATED) {
-                fprintf(stderr, "\x1b[2K\r");
-                dump_dictionary(NULL, ic->metadata,
-                                "\r  New metadata", "    ", AV_LOG_INFO);
-            }
-            if (ic->streams[pkt->stream_index]->event_flags &
-                AVSTREAM_EVENT_FLAG_METADATA_UPDATED) {
-                fprintf(stderr, "\x1b[2K\r");
-                snprintf(metadata_description,
-                         sizeof(metadata_description),
-                         "\r  New metadata for stream %d",
-                         pkt->stream_index);
-                dump_dictionary(NULL, ic->streams[pkt->stream_index]->metadata,
-                                   metadata_description, "    ", AV_LOG_INFO);
-            }
+        if (show_status && ic->streams[pkt->stream_index]->event_flags &
+            AVSTREAM_EVENT_FLAG_METADATA_UPDATED) {
+            fprintf(stderr, "\x1b[2K\r");
+            snprintf(metadata_description,
+                     sizeof(metadata_description),
+                     "\r  New metadata for stream %d",
+                     pkt->stream_index);
+            dump_dictionary(NULL, ic->streams[pkt->stream_index]->metadata,
+                               metadata_description, "    ", AV_LOG_INFO);
         }
-        ic->event_flags &= ~AVFMT_EVENT_FLAG_METADATA_UPDATED;
         ic->streams[pkt->stream_index]->event_flags &= ~AVSTREAM_EVENT_FLAG_METADATA_UPDATED;
 
         /* check if packet is in play range specified by user, then queue, otherwise discard */
@@ -3883,6 +3874,12 @@ int main(int argc, char **argv)
     flags = SDL_INIT_VIDEO | SDL_INIT_AUDIO | SDL_INIT_TIMER;
     if (audio_disable)
         flags &= ~SDL_INIT_AUDIO;
+    else {
+        /* Try to work around an occasional ALSA buffer underflow issue when the
+         * period size is NPOT due to ALSA resampling by forcing the buffer size. */
+        if (!SDL_getenv("SDL_AUDIO_ALSA_SET_BUFFER_SIZE"))
+            SDL_setenv("SDL_AUDIO_ALSA_SET_BUFFER_SIZE","1", 1);
+    }
     if (display_disable)
         flags &= ~SDL_INIT_VIDEO;
     if (SDL_Init (flags)) {

fftools/ffplay_renderer.c

@@ -23,7 +23,9 @@
 #include "ffplay_renderer.h"
 
 #if (SDL_VERSION_ATLEAST(2, 0, 6) && CONFIG_LIBPLACEBO)
-#define HAVE_VULKAN_RENDERER 1
+/* Get PL_API_VER */
+#include <libplacebo/config.h>
+#define HAVE_VULKAN_RENDERER (PL_API_VER >= 278)
 #else
 #define HAVE_VULKAN_RENDERER 0
 #endif

fftools/ffprobe.c

@@ -39,7 +39,6 @@
 #include "libavutil/avutil.h"
 #include "libavutil/bprint.h"
 #include "libavutil/channel_layout.h"
-#include "libavutil/downmix_info.h"
 #include "libavutil/display.h"
 #include "libavutil/film_grain_params.h"
 #include "libavutil/hdr_dynamic_metadata.h"
@@ -216,8 +215,6 @@ typedef enum {
     SECTION_ID_STREAM_GROUP_SUBPIECE,
     SECTION_ID_STREAM_GROUP_BLOCKS,
     SECTION_ID_STREAM_GROUP_BLOCK,
-    SECTION_ID_STREAM_GROUP_SIDE_DATA_LIST,
-    SECTION_ID_STREAM_GROUP_SIDE_DATA,
     SECTION_ID_STREAM_GROUP_STREAMS,
     SECTION_ID_STREAM_GROUP_STREAM,
     SECTION_ID_STREAM_GROUP_DISPOSITION,
@@ -301,7 +298,7 @@ static const AVTextFormatSection sections[] = {
     [SECTION_ID_STREAM_GROUP_STREAM_TAGS] =        { SECTION_ID_STREAM_GROUP_STREAM_TAGS, "tags", AV_TEXTFORMAT_SECTION_FLAG_HAS_VARIABLE_FIELDS, { -1 }, .element_name = "tag", .unique_name = "stream_group_stream_tags" },
     [SECTION_ID_STREAM_GROUP] =                    { SECTION_ID_STREAM_GROUP, "stream_group", 0, { SECTION_ID_STREAM_GROUP_TAGS, SECTION_ID_STREAM_GROUP_DISPOSITION, SECTION_ID_STREAM_GROUP_COMPONENTS, SECTION_ID_STREAM_GROUP_STREAMS, -1 } },
     [SECTION_ID_STREAM_GROUP_COMPONENTS] =         { SECTION_ID_STREAM_GROUP_COMPONENTS, "components", AV_TEXTFORMAT_SECTION_FLAG_IS_ARRAY, { SECTION_ID_STREAM_GROUP_COMPONENT, -1 }, .element_name = "component", .unique_name = "stream_group_components" },
-    [SECTION_ID_STREAM_GROUP_COMPONENT] =          { SECTION_ID_STREAM_GROUP_COMPONENT, "component", AV_TEXTFORMAT_SECTION_FLAG_HAS_VARIABLE_FIELDS|AV_TEXTFORMAT_SECTION_FLAG_HAS_TYPE, { SECTION_ID_STREAM_GROUP_SIDE_DATA_LIST, SECTION_ID_STREAM_GROUP_SUBCOMPONENTS, -1 }, .unique_name = "stream_group_component", .element_name = "component_entry", .get_type = get_stream_group_type },
+    [SECTION_ID_STREAM_GROUP_COMPONENT] =          { SECTION_ID_STREAM_GROUP_COMPONENT, "component", AV_TEXTFORMAT_SECTION_FLAG_HAS_VARIABLE_FIELDS|AV_TEXTFORMAT_SECTION_FLAG_HAS_TYPE, { SECTION_ID_STREAM_GROUP_SUBCOMPONENTS, -1 }, .unique_name = "stream_group_component", .element_name = "component_entry", .get_type = get_stream_group_type },
     [SECTION_ID_STREAM_GROUP_SUBCOMPONENTS] =      { SECTION_ID_STREAM_GROUP_SUBCOMPONENTS, "subcomponents", AV_TEXTFORMAT_SECTION_FLAG_IS_ARRAY, { SECTION_ID_STREAM_GROUP_SUBCOMPONENT, -1 }, .element_name = "component" },
     [SECTION_ID_STREAM_GROUP_SUBCOMPONENT] =       { SECTION_ID_STREAM_GROUP_SUBCOMPONENT, "subcomponent", AV_TEXTFORMAT_SECTION_FLAG_HAS_VARIABLE_FIELDS|AV_TEXTFORMAT_SECTION_FLAG_HAS_TYPE, { SECTION_ID_STREAM_GROUP_PIECES, -1 }, .element_name = "subcomponent_entry", .get_type = get_raw_string_type },
     [SECTION_ID_STREAM_GROUP_PIECES] =             { SECTION_ID_STREAM_GROUP_PIECES, "pieces", AV_TEXTFORMAT_SECTION_FLAG_IS_ARRAY, { SECTION_ID_STREAM_GROUP_PIECE, -1 }, .element_name = "piece", .unique_name = "stream_group_pieces" },
@@ -310,8 +307,6 @@ static const AVTextFormatSection sections[] = {
     [SECTION_ID_STREAM_GROUP_SUBPIECE] =           { SECTION_ID_STREAM_GROUP_SUBPIECE, "subpiece", AV_TEXTFORMAT_SECTION_FLAG_HAS_VARIABLE_FIELDS|AV_TEXTFORMAT_SECTION_FLAG_HAS_TYPE, { SECTION_ID_STREAM_GROUP_BLOCKS, -1 }, .element_name = "subpiece_entry", .get_type = get_raw_string_type },
     [SECTION_ID_STREAM_GROUP_BLOCKS] =             { SECTION_ID_STREAM_GROUP_BLOCKS, "blocks", AV_TEXTFORMAT_SECTION_FLAG_IS_ARRAY, { SECTION_ID_STREAM_GROUP_BLOCK, -1 }, .element_name = "block" },
     [SECTION_ID_STREAM_GROUP_BLOCK] =              { SECTION_ID_STREAM_GROUP_BLOCK, "block", AV_TEXTFORMAT_SECTION_FLAG_HAS_VARIABLE_FIELDS|AV_TEXTFORMAT_SECTION_FLAG_HAS_TYPE, { -1 }, .element_name = "block_entry", .get_type = get_raw_string_type },
-    [SECTION_ID_STREAM_GROUP_SIDE_DATA_LIST] =     { SECTION_ID_STREAM_GROUP_SIDE_DATA_LIST, "side_data_list", AV_TEXTFORMAT_SECTION_FLAG_IS_ARRAY, { SECTION_ID_STREAM_GROUP_SIDE_DATA, -1 }, .element_name = "side_data", .unique_name = "stream_group_side_data_list" },
-    [SECTION_ID_STREAM_GROUP_SIDE_DATA] =          { SECTION_ID_STREAM_GROUP_SIDE_DATA, "side_data", AV_TEXTFORMAT_SECTION_FLAG_HAS_TYPE|AV_TEXTFORMAT_SECTION_FLAG_HAS_VARIABLE_FIELDS, { -1 }, .unique_name = "stream_group_side_data", .element_name = "side_datum", .get_type = get_packet_side_data_type },
     [SECTION_ID_STREAM_GROUP_STREAMS] =            { SECTION_ID_STREAM_GROUP_STREAMS, "streams", AV_TEXTFORMAT_SECTION_FLAG_IS_ARRAY, { SECTION_ID_STREAM_GROUP_STREAM, -1 }, .unique_name = "stream_group_streams" },
     [SECTION_ID_STREAM_GROUP_STREAM] =             { SECTION_ID_STREAM_GROUP_STREAM, "stream", 0, { SECTION_ID_STREAM_GROUP_STREAM_DISPOSITION, SECTION_ID_STREAM_GROUP_STREAM_TAGS, -1 }, .unique_name = "stream_group_stream" },
     [SECTION_ID_STREAM_GROUP_DISPOSITION] =        { SECTION_ID_STREAM_GROUP_DISPOSITION, "disposition", 0, { -1 }, .unique_name = "stream_group_disposition" },
@@ -345,11 +340,12 @@ static const char *print_input_filename;
 static const AVInputFormat *iformat = NULL;
 static const char *output_filename = NULL;
 
+static const char unit_second_str[]         = "s"    ;
 static const char unit_hertz_str[]          = "Hz"   ;
 static const char unit_byte_str[]           = "byte" ;
 static const char unit_bit_per_second_str[] = "bit/s";
 
-static unsigned int nb_streams;
+static int nb_streams;
 static uint64_t *nb_streams_packets;
 static uint64_t *nb_streams_frames;
 static int *selected_streams;
@@ -436,8 +432,8 @@ static void log_callback(void *ptr, int level, const char *fmt, va_list vl)
 
 #define print_list_fmt(k, f, n, m, ...) do {    \
     av_bprint_clear(&pbuf);                     \
-    for (unsigned int idx = 0; idx < n; idx++) {         \
-        for (unsigned int idx2 = 0; idx2 < m; idx2++) {  \
+    for (int idx = 0; idx < n; idx++) {         \
+        for (int idx2 = 0; idx2 < m; idx2++) {  \
             if (idx > 0 || idx2 > 0)            \
                 av_bprint_chars(&pbuf, ' ', 1); \
             av_bprintf(&pbuf, f, __VA_ARGS__);  \
@@ -455,9 +451,7 @@ static void log_callback(void *ptr, int level, const char *fmt, va_list vl)
 #define print_ts(k, v)          avtext_print_ts(tfc, k, v, 0)
 #define print_duration_time(k, v, tb) avtext_print_time(tfc, k, v, tb, 1)
 #define print_duration_ts(k, v)       avtext_print_ts(tfc, k, v, 1)
-#define print_val(k, v, u)            avtext_print_unit_integer(tfc, k, v, AV_TEXTFORMAT_VALUE_FMT_INT, u)
-#define print_int_fmt(k, v, f, u)     avtext_print_unit_integer(tfc, k, v, f, u)
-#define print_decibel(k, v)           avtext_print_unit_double(tfc, k, v, AV_TEXTFORMAT_VALUE_FMT_DECIBEL, 0)
+#define print_val(k, v, u)      avtext_print_unit_integer(tfc, k, v, u)
 
 static void print_integers(AVTextFormatContext *tfc, const char *key,
                            const void *data, int size, const char *format,
@@ -509,30 +503,6 @@ static inline int show_tags(AVTextFormatContext *tfc, AVDictionary *tags, int se
     return ret;
 }
 
-static void print_downmix_info(AVTextFormatContext *tfc,
-                                      const AVDownmixInfo *downmix_info)
-{
-    switch (downmix_info->preferred_downmix_type) {
-    case AV_DOWNMIX_TYPE_LORO:
-        print_str("preferred_downmix_type", "loro");
-        break;
-    case AV_DOWNMIX_TYPE_LTRT:
-        print_str("preferred_downmix_type", "ltrt");
-        break;
-    case AV_DOWNMIX_TYPE_DPLII:
-        print_str("preferred_downmix_type", "dplII");
-        break;
-    default:
-        print_str("preferred_downmix_type", "unknown");
-        break;
-    }
-    print_decibel("center_mix_level_db", downmix_info->center_mix_level);
-    print_decibel("center_mix_level_ltrt_db", downmix_info->center_mix_level_ltrt);
-    print_decibel("surround_mix_level_db", downmix_info->surround_mix_level);
-    print_decibel("surround_mix_level_ltrt_db", downmix_info->surround_mix_level_ltrt);
-    print_decibel("lfe_mix_level_db", downmix_info->lfe_mix_level);
-}
-
 static void print_displaymatrix(AVTextFormatContext *tfc, const int32_t matrix[9])
 {
     double rotation = av_display_rotation_get(matrix);
@@ -831,62 +801,6 @@ static void print_dynamic_hdr10_plus(AVTextFormatContext *tfc, const AVDynamicHD
     }
 }
 
-static void print_dynamic_hdr_smpte2094_app5(AVTextFormatContext *tfc, const AVDynamicHDRSmpte2094App5 *metadata)
-{
-    if (!metadata)
-        return;
-    print_int("application_version", metadata->application_version);
-    print_int("minimum_application_version", metadata->minimum_application_version);
-    print_int("has_custom_hdr_reference_white_flag", metadata->has_custom_hdr_reference_white_flag);
-    print_int("has_adaptive_tone_map_flag", metadata->has_adaptive_tone_map_flag);
-
-    if (metadata->has_custom_hdr_reference_white_flag)
-        print_int("hdr_reference_white", metadata->hdr_reference_white);
-
-    if (!metadata->has_adaptive_tone_map_flag)
-        return;
-
-    print_int("baseline_hdr_headroom", metadata->baseline_hdr_headroom);
-    print_int("use_reference_white_tone_mapping_flag", metadata->use_reference_white_tone_mapping_flag);
-
-    if (metadata->use_reference_white_tone_mapping_flag)
-        return;
-
-    print_int("num_alternate_images", metadata->num_alternate_images);
-    print_int("gain_application_space_chromaticities_flag", metadata->gain_application_space_chromaticities_flag);
-    print_int("has_common_component_mix_params_flag", metadata->has_common_component_mix_params_flag);
-    print_int("has_common_curve_params_flag", metadata->has_common_curve_params_flag);
-
-    if (metadata->gain_application_space_chromaticities_flag == 3) {
-        for (int i = 0; i < 8; i++)
-            print_int("gain_application_space_chromaticities", metadata->gain_application_space_chromaticities[i]);
-    }
-
-    for (int a = 0; a < metadata->num_alternate_images; a++) {
-        print_int("alternate_hdr_headroom", metadata->alternate_hdr_headrooms[a]);
-
-        print_int("component_mixing_type", metadata->component_mixing_type[a]);
-        if (metadata->component_mixing_type[a] == 3) {
-            for (int k = 0; k < 6; k++) {
-                print_int("has_component_mixing_coefficient_flag", metadata->has_component_mixing_coefficient_flag[a][k]);
-                if (metadata->has_component_mixing_coefficient_flag[a][k])
-                    print_int("component_mixing_coefficient", metadata->component_mixing_coefficient[a][k]);
-            }
-        }
-
-        print_int("gain_curve_num_control_points_minus_1", metadata->gain_curve_num_control_points_minus_1[a]);
-        print_int("gain_curve_use_pchip_slope_flag", metadata->gain_curve_use_pchip_slope_flag[a]);
-        for (int c = 0; c <= metadata->gain_curve_num_control_points_minus_1[a]; c++)
-            print_int("gain_curve_control_point_x", metadata->gain_curve_control_points_x[a][c]);
-        for (int c = 0; c <= metadata->gain_curve_num_control_points_minus_1[a]; c++)
-            print_int("gain_curve_control_point_y", metadata->gain_curve_control_points_y[a][c]);
-        if (!metadata->gain_curve_use_pchip_slope_flag[a]) {
-            for (int c = 0; c <= metadata->gain_curve_num_control_points_minus_1[a]; c++)
-                print_int("gain_curve_control_point_theta", metadata->gain_curve_control_points_theta[a][c]);
-        }
-    }
-}
-
 static void print_dynamic_hdr_vivid(AVTextFormatContext *tfc, const AVDynamicHDRVivid *metadata)
 {
     if (!metadata)
@@ -1028,7 +942,7 @@ static void print_film_grain_params(AVTextFormatContext *tfc,
             avtext_print_section_footer(tfc);
         }
 
-        for (unsigned uv = 0; uv < 2; uv++) {
+        for (int uv = 0; uv < 2; uv++) {
             if (!aom->num_uv_points[uv] && !aom->chroma_scaling_from_luma)
                 continue;
 
@@ -1094,8 +1008,7 @@ static void print_film_grain_params(AVTextFormatContext *tfc,
 }
 
 static void print_pkt_side_data(AVTextFormatContext *tfc,
-                                int width,
-                                int height,
+                                AVCodecParameters *par,
                                 const AVPacketSideData *sd,
                                 SectionID id_data)
 {
@@ -1121,7 +1034,7 @@ static void print_pkt_side_data(AVTextFormatContext *tfc,
             print_int("padding", spherical->padding);
         } else if (spherical->projection == AV_SPHERICAL_EQUIRECTANGULAR_TILE) {
             size_t l, t, r, b;
-            av_spherical_tile_bounds(spherical, width, height,
+            av_spherical_tile_bounds(spherical, par->width, par->height,
                                      &l, &t, &r, &b);
             print_int("bound_left", l);
             print_int("bound_top", t);
@@ -1368,7 +1281,7 @@ static void show_packet(AVTextFormatContext *tfc, InputFile *ifile, AVPacket *pk
     print_time("dts_time",        pkt->dts, &st->time_base);
     print_duration_ts("duration",        pkt->duration);
     print_duration_time("duration_time", pkt->duration, &st->time_base);
-    print_int_fmt("size",         pkt->size, AV_TEXTFORMAT_VALUE_FMT_BYTE, unit_byte_str);
+    print_val("size",             pkt->size, unit_byte_str);
     if (pkt->pos != -1) print_fmt    ("pos", "%"PRId64, pkt->pos);
     else                print_str_opt("pos", "N/A");
     print_fmt("flags", "%c%c%c",      pkt->flags & AV_PKT_FLAG_KEY ? 'K' : '_',
@@ -1392,7 +1305,7 @@ static void show_packet(AVTextFormatContext *tfc, InputFile *ifile, AVPacket *pk
 
         avtext_print_section_header(tfc, NULL, SECTION_ID_PACKET_SIDE_DATA_LIST);
         for (int i = 0; i < pkt->side_data_elems; i++) {
-            print_pkt_side_data(tfc, st->codecpar->width, st->codecpar->height, &pkt->side_data[i],
+            print_pkt_side_data(tfc, st->codecpar, &pkt->side_data[i],
                                 SECTION_ID_PACKET_SIDE_DATA);
             avtext_print_section_footer(tfc);
         }
@@ -1428,9 +1341,6 @@ static void show_subtitle(AVTextFormatContext *tfc, AVSubtitle *sub, AVStream *s
     fflush(stdout);
 }
 
-static void print_iamf_param_definition(AVTextFormatContext *tfc, const char *name,
-                                        const AVIAMFParamDefinition *param, SectionID section_id);
-
 static void print_frame_side_data(AVTextFormatContext *tfc,
                                   const AVFrame *frame,
                                   const AVStream *stream)
@@ -1444,9 +1354,7 @@ static void print_frame_side_data(AVTextFormatContext *tfc,
         avtext_print_section_header(tfc, sd, SECTION_ID_FRAME_SIDE_DATA);
         name = av_frame_side_data_name(sd->type);
         print_str("side_data_type", name ? name : "unknown");
-        if (sd->type == AV_FRAME_DATA_DOWNMIX_INFO) {
-            print_downmix_info(tfc, (AVDownmixInfo *)sd->data);
-        } else if (sd->type == AV_FRAME_DATA_DISPLAYMATRIX && sd->size >= 9*4) {
+        if (sd->type == AV_FRAME_DATA_DISPLAYMATRIX && sd->size >= 9*4) {
             print_displaymatrix(tfc, (const int32_t*)sd->data);
         } else if (sd->type == AV_FRAME_DATA_AFD && sd->size > 0) {
             print_int("active_format", *sd->data);
@@ -1471,9 +1379,6 @@ static void print_frame_side_data(AVTextFormatContext *tfc,
         } else if (sd->type == AV_FRAME_DATA_DYNAMIC_HDR_PLUS) {
             AVDynamicHDRPlus *metadata = (AVDynamicHDRPlus *)sd->data;
             print_dynamic_hdr10_plus(tfc, metadata);
-        } else if (sd->type == AV_FRAME_DATA_DYNAMIC_HDR_SMPTE_2094_APP5) {
-            AVDynamicHDRSmpte2094App5 *metadata = (AVDynamicHDRSmpte2094App5 *)sd->data;
-            print_dynamic_hdr_smpte2094_app5(tfc, metadata);
         } else if (sd->type == AV_FRAME_DATA_CONTENT_LIGHT_LEVEL) {
             print_context_light_level(tfc, (AVContentLightMetadata *)sd->data);
         } else if (sd->type == AV_FRAME_DATA_ICC_PROFILE) {
@@ -1495,11 +1400,6 @@ static void print_frame_side_data(AVTextFormatContext *tfc,
             print_int("view_id", *(int*)sd->data);
         } else if (sd->type == AV_FRAME_DATA_EXIF) {
             print_int("size", sd->size);
-        } else if (sd->type == AV_FRAME_DATA_IAMF_MIX_GAIN_PARAM ||
-            sd->type == AV_FRAME_DATA_IAMF_DEMIXING_INFO_PARAM ||
-            sd->type == AV_FRAME_DATA_IAMF_RECON_GAIN_INFO_PARAM) {
-            const AVIAMFParamDefinition *param = (AVIAMFParamDefinition *)sd->data;
-            print_iamf_param_definition(tfc, NULL, param, SECTION_ID_FRAME_SIDE_DATA);
         }
         avtext_print_section_footer(tfc);
     }
@@ -1533,7 +1433,7 @@ static void show_frame(AVTextFormatContext *tfc, AVFrame *frame, AVStream *strea
     print_duration_time("duration_time",     frame->duration, &stream->time_base);
     if (fd && fd->pkt_pos != -1)  print_fmt    ("pkt_pos", "%"PRId64, fd->pkt_pos);
     else                          print_str_opt("pkt_pos", "N/A");
-    if (fd && fd->pkt_size != -1) print_int_fmt("pkt_size", fd->pkt_size, AV_TEXTFORMAT_VALUE_FMT_BYTE, unit_byte_str);
+    if (fd && fd->pkt_size != -1) print_val    ("pkt_size", fd->pkt_size, unit_byte_str);
     else                          print_str_opt("pkt_size", "N/A");
 
     switch (stream->codecpar->codec_type) {
@@ -1796,10 +1696,12 @@ static int read_interval_packets(AVTextFormatContext *tfc, InputFile *ifile,
     }
     av_packet_unref(pkt);
     //Flush remaining frames that are cached in the decoder
-    for (int i = 0; i < ifile->nb_streams; i++) {
+    for (i = 0; i < ifile->nb_streams; i++) {
         pkt->stream_index = i;
         if (do_read_frames) {
             while (process_frame(tfc, ifile, frame, pkt, &(int){1}) > 0);
+            if (ifile->streams[i].dec_ctx)
+                avcodec_flush_buffers(ifile->streams[i].dec_ctx);
         }
     }
 
@@ -1813,33 +1715,17 @@ end:
     return ret;
 }
 
-static void flush_buffers(InputFile *ifile)
-{
-    int i;
-
-    if (!do_read_frames)
-        return;
-    for (i = 0; i < ifile->nb_streams; i++) {
-        if (ifile->streams[i].dec_ctx)
-            avcodec_flush_buffers(ifile->streams[i].dec_ctx);
-    }
-}
-
 static int read_packets(AVTextFormatContext *tfc, InputFile *ifile)
 {
     AVFormatContext *fmt_ctx = ifile->fmt_ctx;
-    int ret = 0;
+    int i, ret = 0;
     int64_t cur_ts = fmt_ctx->start_time;
 
     if (read_intervals_nb == 0) {
         ReadInterval interval = (ReadInterval) { .has_start = 0, .has_end = 0 };
         ret = read_interval_packets(tfc, ifile, &interval, &cur_ts);
     } else {
-        for (int i = 0; i < read_intervals_nb; i++) {
-            /* flushing buffers can reset parts of the private context which may be
-             * read by show_streams(), so only flush between each read_interval */
-            if (i)
-                flush_buffers(ifile);
+        for (i = 0; i < read_intervals_nb; i++) {
             ret = read_interval_packets(tfc, ifile, &read_intervals[i], &cur_ts);
             if (ret < 0)
                 break;
@@ -1852,7 +1738,7 @@ static int read_packets(AVTextFormatContext *tfc, InputFile *ifile)
 static void print_dispositions(AVTextFormatContext *tfc, uint32_t disposition, SectionID section_id)
 {
     avtext_print_section_header(tfc, NULL, section_id);
-    for (unsigned i = 0; i < sizeof(disposition) * CHAR_BIT; i++) {
+    for (int i = 0; i < sizeof(disposition) * CHAR_BIT; i++) {
         const char *disposition_str = av_disposition_to_string(1U << i);
 
         if (disposition_str)
@@ -2075,7 +1961,7 @@ static int show_stream(AVTextFormatContext *tfc, AVFormatContext *fmt_ctx, int s
     if (stream->codecpar->nb_coded_side_data) {
         avtext_print_section_header(tfc, NULL, SECTION_ID_STREAM_SIDE_DATA_LIST);
         for (int i = 0; i < stream->codecpar->nb_coded_side_data; i++) {
-            print_pkt_side_data(tfc, stream->codecpar->width, stream->codecpar->height, &stream->codecpar->coded_side_data[i],
+            print_pkt_side_data(tfc, stream->codecpar, &stream->codecpar->coded_side_data[i],
                                 SECTION_ID_STREAM_SIDE_DATA);
             avtext_print_section_footer(tfc);
         }
@@ -2092,10 +1978,10 @@ static int show_stream(AVTextFormatContext *tfc, AVFormatContext *fmt_ctx, int s
 static int show_streams(AVTextFormatContext *tfc, InputFile *ifile)
 {
     AVFormatContext *fmt_ctx = ifile->fmt_ctx;
-    int ret = 0;
+    int i, ret = 0;
 
     avtext_print_section_header(tfc, NULL, SECTION_ID_STREAMS);
-    for (int i = 0; i < ifile->nb_streams; i++)
+    for (i = 0; i < ifile->nb_streams; i++)
         if (selected_streams[i]) {
             ret = show_stream(tfc, fmt_ctx, i, &ifile->streams[i], 0);
             if (ret < 0)
@@ -2109,7 +1995,7 @@ static int show_streams(AVTextFormatContext *tfc, InputFile *ifile)
 static int show_program(AVTextFormatContext *tfc, InputFile *ifile, AVProgram *program)
 {
     AVFormatContext *fmt_ctx = ifile->fmt_ctx;
-    int ret = 0;
+    int i, ret = 0;
 
     avtext_print_section_header(tfc, NULL, SECTION_ID_PROGRAM);
     print_int("program_id", program->id);
@@ -2123,7 +2009,7 @@ static int show_program(AVTextFormatContext *tfc, InputFile *ifile, AVProgram *p
         goto end;
 
     avtext_print_section_header(tfc, NULL, SECTION_ID_PROGRAM_STREAMS);
-    for (unsigned i = 0; i < program->nb_stream_indexes; i++) {
+    for (i = 0; i < program->nb_stream_indexes; i++) {
         if (selected_streams[program->stream_index[i]]) {
             ret = show_stream(tfc, fmt_ctx, program->stream_index[i], &ifile->streams[program->stream_index[i]], IN_PROGRAM);
             if (ret < 0)
@@ -2140,10 +2026,10 @@ end:
 static int show_programs(AVTextFormatContext *tfc, InputFile *ifile)
 {
     AVFormatContext *fmt_ctx = ifile->fmt_ctx;
-    int ret = 0;
+    int i, ret = 0;
 
     avtext_print_section_header(tfc, NULL, SECTION_ID_PROGRAMS);
-    for (unsigned i = 0; i < fmt_ctx->nb_programs; i++) {
+    for (i = 0; i < fmt_ctx->nb_programs; i++) {
         AVProgram *program = fmt_ctx->programs[i];
         if (!program)
             continue;
@@ -2167,7 +2053,7 @@ static void print_tile_grid_params(AVTextFormatContext *tfc, const AVStreamGroup
     print_int("width",             tile_grid->width);
     print_int("height",            tile_grid->height);
     avtext_print_section_header(tfc, NULL, SECTION_ID_STREAM_GROUP_SUBCOMPONENTS);
-    for (unsigned i = 0; i < tile_grid->nb_tiles; i++) {
+    for (int i = 0; i < tile_grid->nb_tiles; i++) {
         avtext_print_section_header(tfc, "tile_offset", SECTION_ID_STREAM_GROUP_SUBCOMPONENT);
         print_int("stream_index",           tile_grid->offsets[i].idx);
         print_int("tile_horizontal_offset", tile_grid->offsets[i].horizontal);
@@ -2175,15 +2061,6 @@ static void print_tile_grid_params(AVTextFormatContext *tfc, const AVStreamGroup
         avtext_print_section_footer(tfc);
     }
     avtext_print_section_footer(tfc);
-    if (tile_grid->nb_coded_side_data) {
-        avtext_print_section_header(tfc, NULL, SECTION_ID_STREAM_GROUP_SIDE_DATA_LIST);
-        for (int i = 0; i < tile_grid->nb_coded_side_data; i++) {
-            print_pkt_side_data(tfc, tile_grid->width, tile_grid->height, &tile_grid->coded_side_data[i],
-                                SECTION_ID_STREAM_GROUP_SIDE_DATA);
-            avtext_print_section_footer(tfc);
-        }
-        avtext_print_section_footer(tfc);
-    }
     avtext_print_section_footer(tfc);
 }
 
@@ -2191,21 +2068,12 @@ static void print_iamf_param_definition(AVTextFormatContext *tfc, const char *na
                                         const AVIAMFParamDefinition *param, SectionID section_id)
 {
     SectionID subsection_id, parameter_section_id;
-    if (section_id == SECTION_ID_FRAME_SIDE_DATA)
-        subsection_id = SECTION_ID_FRAME_SIDE_DATA_COMPONENT_LIST;
-    else {
-        av_assert0(sections[section_id].children_ids[0] != -1);
-        subsection_id = sections[section_id].children_ids[0];
-    }
-    av_assert0(sections[subsection_id].children_ids[0] != -1);
+    subsection_id = sections[section_id].children_ids[0];
+    av_assert0(subsection_id != -1);
     parameter_section_id = sections[subsection_id].children_ids[0];
-
-    // When printing as part of side-data, skip opening a section
-    if (section_id != SECTION_ID_FRAME_SIDE_DATA)
-        avtext_print_section_header(tfc, "IAMF Param Definition", section_id);
-
-    if (name)
-        print_str("name",           name);
+    av_assert0(parameter_section_id != -1);
+    avtext_print_section_header(tfc, "IAMF Param Definition", section_id);
+    print_str("name",           name);
     print_int("nb_subblocks",   param->nb_subblocks);
     print_int("type",           param->type);
     print_int("parameter_id",   param->parameter_id);
@@ -2214,7 +2082,7 @@ static void print_iamf_param_definition(AVTextFormatContext *tfc, const char *na
     print_int("constant_subblock_duration",          param->constant_subblock_duration);
     if (param->nb_subblocks > 0)
         avtext_print_section_header(tfc, NULL, subsection_id);
-    for (unsigned i = 0; i < param->nb_subblocks; i++) {
+    for (int i = 0; i < param->nb_subblocks; i++) {
         const void *subblock = av_iamf_param_definition_get_subblock(param, i);
         switch(param->type) {
         case AV_IAMF_PARAMETER_DEFINITION_MIX_GAIN: {
@@ -2248,9 +2116,7 @@ static void print_iamf_param_definition(AVTextFormatContext *tfc, const char *na
     }
     if (param->nb_subblocks > 0)
         avtext_print_section_footer(tfc); // subsection_id
-
-    if (section_id != SECTION_ID_FRAME_SIDE_DATA)
-        avtext_print_section_footer(tfc); // section_id
+    avtext_print_section_footer(tfc); // section_id
 }
 
 static void print_iamf_audio_element_params(AVTextFormatContext *tfc, const AVStreamGroup *stg,
@@ -2265,7 +2131,7 @@ static void print_iamf_audio_element_params(AVTextFormatContext *tfc, const AVSt
     print_int("audio_element_type", audio_element->audio_element_type);
     print_int("default_w",          audio_element->default_w);
     avtext_print_section_header(tfc, NULL, SECTION_ID_STREAM_GROUP_SUBCOMPONENTS);
-    for (unsigned i = 0; i < audio_element->nb_layers; i++) {
+    for (int i = 0; i < audio_element->nb_layers; i++) {
         const AVIAMFLayer *layer = audio_element->layers[i];
         char val_str[128];
         avtext_print_section_header(tfc, "IAMF Audio Layer", SECTION_ID_STREAM_GROUP_SUBCOMPONENT);
@@ -2301,7 +2167,7 @@ static void print_iamf_submix_params(AVTextFormatContext *tfc, const AVIAMFSubmi
     print_int("nb_layouts",     submix->nb_layouts);
     print_q("default_mix_gain", submix->default_mix_gain, '/');
     avtext_print_section_header(tfc, NULL, SECTION_ID_STREAM_GROUP_PIECES);
-    for (unsigned i = 0; i < submix->nb_elements; i++) {
+    for (int i = 0; i < submix->nb_elements; i++) {
         const AVIAMFSubmixElement *element = submix->elements[i];
         avtext_print_section_header(tfc, "IAMF Submix Element", SECTION_ID_STREAM_GROUP_PIECE);
         print_int("stream_id",                 element->audio_element_id);
@@ -2324,7 +2190,7 @@ static void print_iamf_submix_params(AVTextFormatContext *tfc, const AVIAMFSubmi
     if (submix->output_mix_config)
         print_iamf_param_definition(tfc, "output_mix_config", submix->output_mix_config,
                                     SECTION_ID_STREAM_GROUP_PIECE);
-    for (unsigned i = 0; i < submix->nb_layouts; i++) {
+    for (int i = 0; i < submix->nb_layouts; i++) {
         const AVIAMFSubmixLayout *layout = submix->layouts[i];
         char val_str[128];
         avtext_print_section_header(tfc, "IAMF Submix Layout", SECTION_ID_STREAM_GROUP_PIECE);
@@ -2354,7 +2220,7 @@ static void print_iamf_mix_presentation_params(AVTextFormatContext *tfc, const A
             print_str(annotation->key, annotation->value);
         avtext_print_section_footer(tfc); // SECTION_ID_STREAM_GROUP_SUBCOMPONENT
     }
-    for (unsigned i = 0; i < mix_presentation->nb_submixes; i++)
+    for (int i = 0; i < mix_presentation->nb_submixes; i++)
         print_iamf_submix_params(tfc, mix_presentation->submixes[i]);
     avtext_print_section_footer(tfc); // SECTION_ID_STREAM_GROUP_SUBCOMPONENTS
     avtext_print_section_footer(tfc); // SECTION_ID_STREAM_GROUP_COMPONENT
@@ -2376,7 +2242,7 @@ static int show_stream_group(AVTextFormatContext *tfc, InputFile *ifile, AVStrea
 {
     AVFormatContext *fmt_ctx = ifile->fmt_ctx;
     AVBPrint pbuf;
-    int ret = 0;
+    int i, ret = 0;
 
     av_bprint_init(&pbuf, 1, AV_BPRINT_SIZE_UNLIMITED);
     avtext_print_section_header(tfc, NULL, SECTION_ID_STREAM_GROUP);
@@ -2401,7 +2267,7 @@ static int show_stream_group(AVTextFormatContext *tfc, InputFile *ifile, AVStrea
         goto end;
 
     avtext_print_section_header(tfc, NULL, SECTION_ID_STREAM_GROUP_STREAMS);
-    for (unsigned i = 0; i < stg->nb_streams; i++) {
+    for (i = 0; i < stg->nb_streams; i++) {
         if (selected_streams[stg->streams[i]->index]) {
             ret = show_stream(tfc, fmt_ctx, stg->streams[i]->index, &ifile->streams[stg->streams[i]->index], IN_STREAM_GROUP);
             if (ret < 0)
@@ -2419,10 +2285,10 @@ end:
 static int show_stream_groups(AVTextFormatContext *tfc, InputFile *ifile)
 {
     AVFormatContext *fmt_ctx = ifile->fmt_ctx;
-    int ret = 0;
+    int i, ret = 0;
 
     avtext_print_section_header(tfc, NULL, SECTION_ID_STREAM_GROUPS);
-    for (unsigned i = 0; i < fmt_ctx->nb_stream_groups; i++) {
+    for (i = 0; i < fmt_ctx->nb_stream_groups; i++) {
         AVStreamGroup *stg = fmt_ctx->stream_groups[i];
 
         ret = show_stream_group(tfc, ifile, stg);
@@ -2436,10 +2302,10 @@ static int show_stream_groups(AVTextFormatContext *tfc, InputFile *ifile)
 static int show_chapters(AVTextFormatContext *tfc, InputFile *ifile)
 {
     AVFormatContext *fmt_ctx = ifile->fmt_ctx;
-    int ret = 0;
+    int i, ret = 0;
 
     avtext_print_section_header(tfc, NULL, SECTION_ID_CHAPTERS);
-    for (unsigned i = 0; i < fmt_ctx->nb_chapters; i++) {
+    for (i = 0; i < fmt_ctx->nb_chapters; i++) {
         AVChapter *chapter = fmt_ctx->chapters[i];
 
         avtext_print_section_header(tfc, NULL, SECTION_ID_CHAPTER);
@@ -2476,7 +2342,7 @@ static int show_format(AVTextFormatContext *tfc, InputFile *ifile)
     }
     print_time("start_time",      fmt_ctx->start_time, &AV_TIME_BASE_Q);
     print_time("duration",        fmt_ctx->duration,   &AV_TIME_BASE_Q);
-    if (size >= 0) print_int_fmt("size", size, AV_TEXTFORMAT_VALUE_FMT_BYTE, unit_byte_str);
+    if (size >= 0) print_val    ("size", size, unit_byte_str);
     else           print_str_opt("size", "N/A");
     if (fmt_ctx->bit_rate > 0) print_val    ("bit_rate", fmt_ctx->bit_rate, unit_bit_per_second_str);
     else                       print_str_opt("bit_rate", "N/A");
@@ -2559,7 +2425,7 @@ static const AVCodec *get_decoder_for_stream(AVFormatContext *fmt_ctx, AVStream
 static int open_input_file(InputFile *ifile, const char *filename,
                            const char *print_filename)
 {
-    int err;
+    int err, i;
     AVFormatContext *fmt_ctx = NULL;
     const AVDictionaryEntry *t = NULL;
     int scan_all_pmts_set = 0;
@@ -2600,7 +2466,7 @@ static int open_input_file(InputFile *ifile, const char *filename,
 
         err = avformat_find_stream_info(fmt_ctx, opts);
 
-        for (int i = 0; i < orig_nb_streams; i++)
+        for (i = 0; i < orig_nb_streams; i++)
             av_dict_free(&opts[i]);
         av_freep(&opts);
 
@@ -2618,7 +2484,7 @@ static int open_input_file(InputFile *ifile, const char *filename,
     ifile->nb_streams = fmt_ctx->nb_streams;
 
     /* bind a decoder to each input stream */
-    for (unsigned i = 0; i < fmt_ctx->nb_streams; i++) {
+    for (i = 0; i < fmt_ctx->nb_streams; i++) {
         InputStream *ist = &ifile->streams[i];
         AVStream *stream = fmt_ctx->streams[i];
         const AVCodec *codec;
@@ -2676,9 +2542,10 @@ static int open_input_file(InputFile *ifile, const char *filename,
 
 static void close_input_file(InputFile *ifile)
 {
+    int i;
 
     /* close decoder for each stream */
-    for (int i = 0; i < ifile->nb_streams; i++)
+    for (i = 0; i < ifile->nb_streams; i++)
         avcodec_free_context(&ifile->streams[i].dec_ctx);
 
     av_freep(&ifile->streams);
@@ -2691,7 +2558,7 @@ static int probe_file(AVTextFormatContext *tfc, const char *filename,
                       const char *print_filename)
 {
     InputFile ifile = { 0 };
-    int ret;
+    int ret, i;
     int section_id;
 
     do_analyze_frames = do_analyze_frames && do_show_streams;
@@ -2711,7 +2578,7 @@ static int probe_file(AVTextFormatContext *tfc, const char *filename,
     REALLOCZ_ARRAY_STREAM(streams_with_closed_captions,0,ifile.fmt_ctx->nb_streams);
     REALLOCZ_ARRAY_STREAM(streams_with_film_grain,0,ifile.fmt_ctx->nb_streams);
 
-    for (unsigned i = 0; i < ifile.fmt_ctx->nb_streams; i++) {
+    for (i = 0; i < ifile.fmt_ctx->nb_streams; i++) {
         if (stream_specifier) {
             ret = avformat_match_stream_specifier(ifile.fmt_ctx,
                                                   ifile.fmt_ctx->streams[i],
@@ -2929,9 +2796,9 @@ static inline void mark_section_show_entries(SectionID section_id,
 static int match_section(const char *section_name,
                          int show_all_entries, AVDictionary *entries)
 {
-    int ret = 0;
+    int i, ret = 0;
 
-    for (unsigned i = 0; i < FF_ARRAY_ELEMS(sections); i++) {
+    for (i = 0; i < FF_ARRAY_ELEMS(sections); i++) {
         const struct AVTextFormatSection *section = &sections[i];
         if (!strcmp(section_name, section->name) ||
             (section->unique_name && !strcmp(section_name, section->unique_name))) {

fftools/graph/graphprint.c

@@ -141,7 +141,7 @@ typedef struct GraphPrintContext {
 #define print_q(k, v, s)        avtext_print_rational(tfc, k, v, s)
 #define print_str(k, v)         avtext_print_string(tfc, k, v, 0)
 #define print_str_opt(k, v)     avtext_print_string(tfc, k, v, gpc->opt_flags)
-#define print_val(k, v, u)      avtext_print_unit_integer(tfc, k, v, AV_TEXTFORMAT_VALUE_FMT_INT, u)
+#define print_val(k, v, u)      avtext_print_unit_integer(tfc, k, v, u)
 
 #define print_fmt(k, f, ...) do {              \
     av_bprint_clear(&gpc->pbuf);                    \

fftools/opt_common.c

@@ -68,11 +68,6 @@ enum show_muxdemuxers {
     SHOW_MUXERS,
 };
 
-enum show_codec {
-    SHOW_DECODER,
-    SHOW_ENCODER,
-};
-
 static FILE *report_file;
 static int report_file_level = AV_LOG_DEBUG;
 
@@ -326,12 +321,6 @@ static void print_codec(const AVCodec *c)
         printf("hardware ");
     if (c->capabilities & AV_CODEC_CAP_HYBRID)
         printf("hybrid ");
-    if (c->capabilities & AV_CODEC_CAP_ENCODER_REORDERED_OPAQUE)
-        printf("reorderedopaque ");
-    if (c->capabilities & AV_CODEC_CAP_ENCODER_FLUSH)
-        printf("flush ");
-    if (c->capabilities & AV_CODEC_CAP_ENCODER_RECON_FRAME)
-        printf("reconframe ");
     if (!c->capabilities)
         printf("none");
     printf("\n");
@@ -602,9 +591,9 @@ int show_help(void *optctx, const char *opt, const char *arg)
     if (!*topic) {
         show_help_default(topic, par);
     } else if (!strcmp(topic, "decoder")) {
-        show_help_codec(par, SHOW_DECODER);
+        show_help_codec(par, 0);
     } else if (!strcmp(topic, "encoder")) {
-        show_help_codec(par, SHOW_ENCODER);
+        show_help_codec(par, 1);
     } else if (!strcmp(topic, "demuxer")) {
         show_help_demuxer(par);
     } else if (!strcmp(topic, "muxer")) {
@@ -719,16 +708,16 @@ int show_codecs(void *optctx, const char *opt, const char *arg)
 
         /* print decoders/encoders when there's more than one or their
          * names are different from codec name */
-        while ((codec = next_codec_for_id(desc->id, &iter, SHOW_DECODER))) {
+        while ((codec = next_codec_for_id(desc->id, &iter, 0))) {
             if (strcmp(codec->name, desc->name)) {
-                print_codecs_for_id(desc->id, SHOW_DECODER);
+                print_codecs_for_id(desc->id, 0);
                 break;
             }
         }
         iter = NULL;
-        while ((codec = next_codec_for_id(desc->id, &iter, SHOW_ENCODER))) {
+        while ((codec = next_codec_for_id(desc->id, &iter, 1))) {
             if (strcmp(codec->name, desc->name)) {
-                print_codecs_for_id(desc->id, SHOW_ENCODER);
+                print_codecs_for_id(desc->id, 1);
                 break;
             }
         }
@@ -785,12 +774,12 @@ static int print_codecs(int encoder)
 
 int show_decoders(void *optctx, const char *opt, const char *arg)
 {
-    return print_codecs(SHOW_DECODER);
+    return print_codecs(0);
 }
 
 int show_encoders(void *optctx, const char *opt, const char *arg)
 {
-    return print_codecs(SHOW_ENCODER);
+    return print_codecs(1);
 }
 
 int show_bsfs(void *optctx, const char *opt, const char *arg)
@@ -887,7 +876,7 @@ static int show_formats_devices(void *optctx, const char *opt, const char *arg,
         const char *name      = NULL;
         const char *long_name = NULL;
 
-        if (muxdemuxers != SHOW_DEMUXERS) {
+        if (muxdemuxers !=SHOW_DEMUXERS) {
             ofmt_opaque = NULL;
             while ((ofmt = av_muxer_iterate(&ofmt_opaque))) {
                 is_dev = is_device(ofmt->priv_class);

fftools/textformat/avtextformat.c

@@ -18,21 +18,19 @@
  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  */
 
-#include <inttypes.h>
 #include <limits.h>
-#include <math.h>
 #include <stdarg.h>
+#include <stdint.h>
 #include <stdio.h>
 #include <string.h>
 
 #include "libavutil/mem.h"
 #include "libavutil/avassert.h"
-#include "libavutil/avutil.h"
 #include "libavutil/base64.h"
 #include "libavutil/bprint.h"
-#include "libavutil/common.h"
 #include "libavutil/error.h"
 #include "libavutil/hash.h"
+#include "libavutil/intreadwrite.h"
 #include "libavutil/macros.h"
 #include "libavutil/opt.h"
 #include "avtextformat.h"
@@ -240,7 +238,12 @@ fail:
     return ret;
 }
 
+/* Temporary definitions during refactoring */
 static const char unit_second_str[]         = "s";
+static const char unit_hertz_str[]          = "Hz";
+static const char unit_byte_str[]           = "byte";
+static const char unit_bit_per_second_str[] = "bit/s";
+
 
 void avtext_print_section_header(AVTextFormatContext *tctx, const void *data, int section_id)
 {
@@ -364,30 +367,24 @@ struct unit_value {
         int64_t i;
     } val;
 
-    AVTextFormatValueFormat fmt;
     const char *unit;
 };
 
-static const char float_fmt_full[] = "%f";
-static const char float_fmt_singledigit[] = "%.1f";
 static char *value_string(const AVTextFormatContext *tctx, char *buf, int buf_size, struct unit_value uv)
 {
     double vald;
     int64_t vali = 0;
-    const char *float_fmt = 0;
+    int show_float = 0;
 
-    if (uv.fmt == AV_TEXTFORMAT_VALUE_FMT_DECIBEL) {
-        vald = 20 * log10(uv.val.d);
-        float_fmt = float_fmt_singledigit;
-    } else if (uv.fmt >= AV_TEXTFORMAT_VALUE_FMT_DOUBLE) {
+    if (uv.unit == unit_second_str) {
         vald = uv.val.d;
-        float_fmt = float_fmt_full;
+        show_float = 1;
     } else {
         vald = (double)uv.val.i;
         vali = uv.val.i;
     }
 
-    if (uv.fmt == AV_TEXTFORMAT_VALUE_FMT_SECOND && tctx->opts.use_value_sexagesimal_format) {
+    if (uv.unit == unit_second_str && tctx->opts.use_value_sexagesimal_format) {
         double secs;
         int hours, mins;
         secs  = vald;
@@ -402,7 +399,7 @@ static char *value_string(const AVTextFormatContext *tctx, char *buf, int buf_si
         if (tctx->opts.use_value_prefix && vald > 1) {
             int64_t index;
 
-            if (uv.fmt == AV_TEXTFORMAT_VALUE_FMT_BYTE && tctx->opts.use_byte_value_binary_prefix) {
+            if (uv.unit == unit_byte_str && tctx->opts.use_byte_value_binary_prefix) {
                 index = (int64_t)(log2(vald) / 10);
                 index = av_clip64(index, 0, FF_ARRAY_ELEMS(si_prefixes) - 1);
                 vald /= si_prefixes[index].bin_val;
@@ -416,46 +413,29 @@ static char *value_string(const AVTextFormatContext *tctx, char *buf, int buf_si
             vali = (int64_t)vald;
         }
 
-        if (float_fmt || (tctx->opts.use_value_prefix && vald != (int64_t)vald))
-            snprintf(buf, buf_size, float_fmt ? float_fmt : "%f", vald);
+        if (show_float || (tctx->opts.use_value_prefix && vald != (int64_t)vald))
+            snprintf(buf, buf_size, "%f", vald);
         else
             snprintf(buf, buf_size, "%"PRId64, vali);
 
-        av_strlcatf(buf, buf_size, "%s%s%s", *prefix_string || tctx->opts.show_value_unit && uv.unit && *uv.unit ? " " : "",
-                    prefix_string, tctx->opts.show_value_unit && uv.unit ? uv.unit : "");
+        av_strlcatf(buf, buf_size, "%s%s%s", *prefix_string || tctx->opts.show_value_unit ? " " : "",
+                    prefix_string, tctx->opts.show_value_unit ? uv.unit : "");
     }
 
     return buf;
 }
 
 
-void avtext_print_unit_integer(AVTextFormatContext *tctx, const char *key, int64_t val, AVTextFormatValueFormat fmt, const char *unit)
+void avtext_print_unit_integer(AVTextFormatContext *tctx, const char *key, int64_t val, const char *unit)
 {
     char val_str[128];
     struct unit_value uv;
-
-    av_assert0(fmt < AV_TEXTFORMAT_VALUE_FMT_DOUBLE);
-
     uv.val.i = val;
-    uv.fmt = fmt;
     uv.unit = unit;
     avtext_print_string(tctx, key, value_string(tctx, val_str, sizeof(val_str), uv), 0);
 }
 
 
-void avtext_print_unit_double(AVTextFormatContext *tctx, const char *key, double val, AVTextFormatValueFormat fmt, const char *unit)
-{
-    char val_str[128];
-    struct unit_value uv;
-
-    av_assert0(fmt >= AV_TEXTFORMAT_VALUE_FMT_DOUBLE);
-
-    uv.val.d = val;
-    uv.fmt = fmt;
-    uv.unit = unit;
-    avtext_print_string(tctx, key, value_string(tctx, val_str, sizeof(val_str), uv), 0);
-}
-
 int avtext_print_string(AVTextFormatContext *tctx, const char *key, const char *val, int flags)
 {
     const AVTextFormatSection *section;
@@ -515,7 +495,6 @@ void avtext_print_time(AVTextFormatContext *tctx, const char *key,
         double d = av_q2d(*time_base) * ts;
         struct unit_value uv;
         uv.val.d = d;
-        uv.fmt = AV_TEXTFORMAT_VALUE_FMT_SECOND;
         uv.unit = unit_second_str;
         value_string(tctx, buf, sizeof(buf), uv);
         avtext_print_string(tctx, key, buf, 0);

fftools/textformat/avtextformat.h

@@ -161,14 +161,6 @@ struct AVTextFormatContext {
     unsigned int string_validation_utf8_flags;
 };
 
-typedef enum {
-    AV_TEXTFORMAT_VALUE_FMT_INT,
-    AV_TEXTFORMAT_VALUE_FMT_BYTE,
-    AV_TEXTFORMAT_VALUE_FMT_DOUBLE = 0x100,
-    AV_TEXTFORMAT_VALUE_FMT_SECOND,
-    AV_TEXTFORMAT_VALUE_FMT_DECIBEL,
-} AVTextFormatValueFormat;
-
 #define AV_TEXTFORMAT_PRINT_STRING_OPTIONAL 1
 #define AV_TEXTFORMAT_PRINT_STRING_VALIDATE 2
 
@@ -186,9 +178,7 @@ void avtext_print_integer(AVTextFormatContext *tctx, const char *key, int64_t va
 
 int avtext_print_string(AVTextFormatContext *tctx, const char *key, const char *val, int flags);
 
-void avtext_print_unit_integer(AVTextFormatContext *tctx, const char *key, int64_t val, AVTextFormatValueFormat fmt, const char *unit);
-
-void avtext_print_unit_double(AVTextFormatContext *tctx, const char *key, double val, AVTextFormatValueFormat fmt, const char *unit);
+void avtext_print_unit_integer(AVTextFormatContext *tctx, const char *key, int64_t val, const char *unit);
 
 void avtext_print_rational(AVTextFormatContext *tctx, const char *key, AVRational q, char sep);
 

fftools/textformat/tf_compact.c

@@ -18,13 +18,13 @@
  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  */
 
-#include <inttypes.h>
 #include <limits.h>
 #include <stdarg.h>
+#include <stdint.h>
+#include <stdio.h>
 #include <string.h>
 
 #include "avtextformat.h"
-#include "libavutil/avutil.h"
 #include "libavutil/bprint.h"
 #include "libavutil/error.h"
 #include "libavutil/opt.h"

fftools/textformat/tf_default.c

@@ -18,14 +18,13 @@
  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  */
 
-#include <inttypes.h>
 #include <limits.h>
 #include <stdarg.h>
+#include <stdint.h>
 #include <stdio.h>
 #include <string.h>
 
 #include "avtextformat.h"
-#include "libavutil/avutil.h"
 #include "libavutil/bprint.h"
 #include "libavutil/opt.h"
 #include "tf_internal.h"

fftools/textformat/tf_flat.c

@@ -18,9 +18,10 @@
  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  */
 
-#include <inttypes.h>
 #include <limits.h>
 #include <stdarg.h>
+#include <stdint.h>
+#include <stdio.h>
 #include <string.h>
 
 #include "avtextformat.h"

fftools/textformat/tf_ini.c

@@ -18,13 +18,14 @@
  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  */
 
-#include <inttypes.h>
 #include <limits.h>
 #include <stdarg.h>
+#include <stdint.h>
+#include <stdio.h>
+#include <string.h>
 
 #include "avtextformat.h"
 
-#include "libavutil/attributes.h"
 #include "libavutil/bprint.h"
 #include "libavutil/opt.h"
 #include "tf_internal.h"
@@ -73,7 +74,7 @@ static char *ini_escape_str(AVBPrint *dst, const char *src)
         case '=':
         case ':':
             av_bprint_chars(dst, '\\', 1);
-            av_fallthrough;
+            /* fallthrough */
         default:
             if ((unsigned char)c < 32)
                 av_bprintf(dst, "\\x00%02x", (unsigned char)c);

fftools/textformat/tf_json.c

@@ -18,9 +18,10 @@
  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  */
 
-#include <inttypes.h>
 #include <limits.h>
 #include <stdarg.h>
+#include <stdint.h>
+#include <stdio.h>
 #include <string.h>
 
 #include "avtextformat.h"

fftools/textformat/tf_mermaid.c

@@ -18,9 +18,10 @@
  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  */
 
-#include <inttypes.h>
 #include <limits.h>
 #include <stdarg.h>
+#include <stdint.h>
+#include <stdio.h>
 #include <string.h>
 
 #include "avtextformat.h"

fftools/textformat/tf_xml.c

@@ -18,8 +18,8 @@
  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  */
 
-#include <inttypes.h>
-#include <stddef.h>
+#include <stdint.h>
+#include <string.h>
 
 #include "avtextformat.h"
 #include "libavutil/bprint.h"

fftools/thread_queue.c

@@ -194,7 +194,7 @@ static int receive_locked(ThreadQueue *tq, int *stream_idx,
     return nb_finished == tq->nb_streams ? AVERROR_EOF : AVERROR(EAGAIN);
 }
 
-int tq_receive(ThreadQueue *tq, int *stream_idx, void *data, int flags)
+int tq_receive(ThreadQueue *tq, int *stream_idx, void *data)
 {
     int ret;
 
@@ -211,7 +211,7 @@ int tq_receive(ThreadQueue *tq, int *stream_idx, void *data, int flags)
         if (can_read != av_container_fifo_can_read(tq->fifo))
             pthread_cond_broadcast(&tq->cond);
 
-        if (ret == AVERROR(EAGAIN) && !(flags & THREAD_QUEUE_FLAG_NO_BLOCK)) {
+        if (ret == AVERROR(EAGAIN)) {
             pthread_cond_wait(&tq->cond, &tq->lock);
             continue;
         }

fftools/thread_queue.h

@@ -26,12 +26,6 @@ enum ThreadQueueType {
     THREAD_QUEUE_PACKETS,
 };
 
-enum ThreadQueueFlags {
-    /* When set, tq_receive() will return AVERROR(EAGAIN) instead of blocking
-     * when the queue is empty or choked. */
-    THREAD_QUEUE_FLAG_NO_BLOCK = (1 << 0),
-};
-
 typedef struct ThreadQueue ThreadQueue;
 
 /**
@@ -80,8 +74,6 @@ void tq_choke(ThreadQueue *tq, int choked);
  *                   written here
  * @param data the data item will be written here on success using the
  *             callback provided to tq_alloc()
- * @param flags combination of THREAD_QUEUE_FLAG_*
- *
  * @return
  * - 0 a data item was successfully read; *stream_idx contains a non-negative
  *   stream index
@@ -89,8 +81,7 @@ void tq_choke(ThreadQueue *tq, int choked);
  *   side has marked the given stream as finished. This will happen at most once
  *   for each stream. When *stream_idx is -1, all streams are done.
  */
-int tq_receive(ThreadQueue *tq, int *stream_idx, void *data, int flags);
-
+int tq_receive(ThreadQueue *tq, int *stream_idx, void *data);
 /**
  * Mark the given stream finished from the receiving side.
  */

libavcodec/Makefile

@@ -115,9 +115,10 @@ OBJS-$(CONFIG_H264PARSE)               += h264_parse.o h264_ps.o h264data.o \
                                           h2645data.o h2645_parse.o h2645_vui.o
 OBJS-$(CONFIG_H264PRED)                += h264pred.o
 OBJS-$(CONFIG_H264QPEL)                += h264qpel.o
-OBJS-$(CONFIG_H264_SEI)                += h264_sei.o h2645_sei.o
+OBJS-$(CONFIG_H264_SEI)                += h264_sei.o h2645_sei.o aom_film_grain.o
 OBJS-$(CONFIG_HEVCPARSE)               += h2645data.o h2645_parse.o h2645_vui.o
-OBJS-$(CONFIG_HEVC_SEI)                += h2645_sei.o
+OBJS-$(CONFIG_HEVC_SEI)                += h2645_sei.o aom_film_grain.o \
+                                          dynamic_hdr_vivid.o
 OBJS-$(CONFIG_HPELDSP)                 += hpeldsp.o
 OBJS-$(CONFIG_HUFFMAN)                 += huffman.o
 OBJS-$(CONFIG_HUFFYUVDSP)              += huffyuvdsp.o
@@ -125,12 +126,11 @@ OBJS-$(CONFIG_HUFFYUVENCDSP)           += huffyuvencdsp.o
 OBJS-$(CONFIG_IDCTDSP)                 += idctdsp.o simple_idct.o jrevdct.o
 OBJS-$(CONFIG_INFLATE_WRAPPER)         += zlib_wrapper.o
 OBJS-$(CONFIG_INTRAX8)                 += intrax8.o intrax8dsp.o msmpeg4_vc1_data.o
-OBJS-$(CONFIG_ITUT_T35)                += itut35.o aom_film_grain.o dynamic_hdr_vivid.o
 OBJS-$(CONFIG_IVIDSP)                  += ivi_dsp.o
 OBJS-$(CONFIG_JNI)                     += ffjni.o jni.o
 OBJS-$(CONFIG_JPEGTABLES)              += jpegtables.o
 OBJS-$(CONFIG_LCMS2)                   += fflcms2.o
-OBJS-$(CONFIG_LIBLCEVC_DEC)            += lcevcdec.o lcevctab.o
+OBJS-$(CONFIG_LIBLCEVC_DEC)            += lcevcdec.o
 OBJS-$(CONFIG_LLAUDDSP)                += lossless_audiodsp.o
 OBJS-$(CONFIG_LLVIDDSP)                += lossless_videodsp.o
 OBJS-$(CONFIG_LLVIDENCDSP)             += lossless_videoencdsp.o
@@ -203,7 +203,7 @@ OBJS-$(CONFIG_AAC_FIXED_DECODER)       += aactab.o \
                                           sbrdsp_fixed.o aacpsdsp_fixed.o cbrt_data_fixed.o \
                                           $(if $(!CONFIG_HARDCODED_TABLES), cbrt_tablegen_common.o)
 OBJS-$(CONFIG_AAC_ENCODER)             += aacenc.o aaccoder.o aacenctab.o    \
-                                          aacpsy.o aactab.o aacencdsp.o \
+                                          aacpsy.o aactab.o      \
                                           aacenc_is.o \
                                           aacenc_tns.o \
                                           psymodel.o kbdwin.o \
@@ -252,7 +252,6 @@ OBJS-$(CONFIG_APTX_HD_ENCODER)         += aptxenc.o aptx.o
 OBJS-$(CONFIG_APNG_DECODER)            += png.o pngdec.o pngdsp.o
 OBJS-$(CONFIG_APNG_ENCODER)            += png.o pngenc.o
 OBJS-$(CONFIG_APV_DECODER)             += apv_decode.o apv_entropy.o apv_dsp.o
-OBJS-$(CONFIG_APV_VULKAN_ENCODER)      += apv_encode_vulkan.o
 OBJS-$(CONFIG_ARBC_DECODER)            += arbc.o
 OBJS-$(CONFIG_ARGO_DECODER)            += argo.o
 OBJS-$(CONFIG_SSA_DECODER)             += assdec.o ass.o
@@ -443,6 +442,7 @@ OBJS-$(CONFIG_H264_MMAL_DECODER)       += mmaldec.o
 OBJS-$(CONFIG_H264_NVENC_ENCODER)      += nvenc_h264.o nvenc.o
 OBJS-$(CONFIG_H264_OH_DECODER)         += ohcodec.o ohdec.o
 OBJS-$(CONFIG_H264_OH_ENCODER)         += ohcodec.o ohenc.o
+OBJS-$(CONFIG_H264_OMX_ENCODER)        += omx.o
 OBJS-$(CONFIG_H264_QSV_DECODER)        += qsvdec.o
 OBJS-$(CONFIG_H264_QSV_ENCODER)        += qsvenc_h264.o
 OBJS-$(CONFIG_H264_RKMPP_DECODER)      += rkmppdec.o
@@ -461,7 +461,7 @@ OBJS-$(CONFIG_HCA_DECODER)             += hcadec.o
 OBJS-$(CONFIG_HCOM_DECODER)            += hcom.o
 OBJS-$(CONFIG_HDR_DECODER)             += hdrdec.o
 OBJS-$(CONFIG_HDR_ENCODER)             += hdrenc.o
-OBJS-$(CONFIG_HEVC_DECODER)            += h274.o
+OBJS-$(CONFIG_HEVC_DECODER)            += aom_film_grain.o h274.o
 OBJS-$(CONFIG_HEVC_AMF_ENCODER)        += amfenc_hevc.o
 OBJS-$(CONFIG_HEVC_AMF_DECODER)        += amfdec.o
 OBJS-$(CONFIG_HEVC_CUVID_DECODER)      += cuviddec.o
@@ -587,6 +587,7 @@ OBJS-$(CONFIG_MPEG4_ENCODER)           += mpeg4videoenc.o
 OBJS-$(CONFIG_MPEG4_CUVID_DECODER)     += cuviddec.o
 OBJS-$(CONFIG_MPEG4_MEDIACODEC_DECODER) += mediacodecdec.o
 OBJS-$(CONFIG_MPEG4_MEDIACODEC_ENCODER) += mediacodecenc.o
+OBJS-$(CONFIG_MPEG4_OMX_ENCODER)       += omx.o
 OBJS-$(CONFIG_MPEG4_V4L2M2M_DECODER)   += v4l2_m2m_dec.o
 OBJS-$(CONFIG_MPEG4_V4L2M2M_ENCODER)   += v4l2_m2m_enc.o
 OBJS-$(CONFIG_MPL2_DECODER)            += mpl2dec.o ass.o
@@ -625,7 +626,6 @@ OBJS-$(CONFIG_PBM_ENCODER)             += pnmenc.o
 OBJS-$(CONFIG_PCX_DECODER)             += pcx.o
 OBJS-$(CONFIG_PCX_ENCODER)             += pcxenc.o
 OBJS-$(CONFIG_PDV_DECODER)             += pdvdec.o
-OBJS-$(CONFIG_PDV_ENCODER)             += pdvenc.o
 OBJS-$(CONFIG_PFM_DECODER)             += pnmdec.o pnm.o
 OBJS-$(CONFIG_PFM_ENCODER)             += pnmenc.o
 OBJS-$(CONFIG_PGM_DECODER)             += pnmdec.o pnm.o
@@ -723,6 +723,9 @@ OBJS-$(CONFIG_SNOW_ENCODER)            += snowenc.o snow.o snow_dwt.o
                                           h263data.o ituh263enc.o \
                                           motion_est.o ratecontrol.o
 OBJS-$(CONFIG_SOL_DPCM_DECODER)        += dpcm.o
+OBJS-$(CONFIG_SONIC_DECODER)           += sonic.o
+OBJS-$(CONFIG_SONIC_ENCODER)           += sonic.o
+OBJS-$(CONFIG_SONIC_LS_ENCODER)        += sonic.o
 OBJS-$(CONFIG_SPEEDHQ_DECODER)         += speedhqdec.o speedhq.o mpeg12.o \
                                           mpeg12data.o
 OBJS-$(CONFIG_SPEEDHQ_ENCODER)         += speedhq.o mpeg12data.o mpeg12enc.o speedhqenc.o
@@ -833,7 +836,7 @@ OBJS-$(CONFIG_VP9_V4L2M2M_DECODER)     += v4l2_m2m_dec.o
 OBJS-$(CONFIG_VQA_DECODER)             += vqavideo.o
 OBJS-$(CONFIG_VQC_DECODER)             += vqcdec.o
 OBJS-$(CONFIG_VVC_DECODER)             += executor.o h2645data.o
-OBJS-$(CONFIG_VVC_SEI)                 += h2645_sei.o h274.o
+OBJS-$(CONFIG_VVC_SEI)                 += h2645_sei.o aom_film_grain.o h274.o
 OBJS-$(CONFIG_WADY_DPCM_DECODER)       += dpcm.o
 OBJS-$(CONFIG_WAVARC_DECODER)          += wavarc.o
 OBJS-$(CONFIG_WAVPACK_DECODER)         += wavpack.o wavpackdata.o dsd.o
@@ -842,7 +845,6 @@ OBJS-$(CONFIG_WBMP_DECODER)            += wbmpdec.o
 OBJS-$(CONFIG_WBMP_ENCODER)            += wbmpenc.o
 OBJS-$(CONFIG_WCMV_DECODER)            += wcmv.o
 OBJS-$(CONFIG_WEBP_DECODER)            += webp.o
-OBJS-$(CONFIG_WEBP_ANIM_DECODER)       += webp.o
 OBJS-$(CONFIG_WEBVTT_DECODER)          += webvttdec.o ass.o
 OBJS-$(CONFIG_WEBVTT_ENCODER)          += webvttenc.o ass_split.o
 OBJS-$(CONFIG_WMALOSSLESS_DECODER)     += wmalosslessdec.o wma_common.o
@@ -1044,7 +1046,6 @@ OBJS-$(CONFIG_VIDEOTOOLBOX)               += videotoolbox.o
 OBJS-$(CONFIG_VDPAU)                      += vdpau.o
 OBJS-$(CONFIG_VULKAN)                     += vulkan.o vulkan_video.o
 
-OBJS-$(CONFIG_APV_VULKAN_HWACCEL)         += vulkan_decode.o vulkan_apv.o
 OBJS-$(CONFIG_AV1_D3D11VA_HWACCEL)        += dxva2_av1.o
 OBJS-$(CONFIG_AV1_DXVA2_HWACCEL)          += dxva2_av1.o
 OBJS-$(CONFIG_AV1_D3D12VA_HWACCEL)        += dxva2_av1.o d3d12va_av1.o
@@ -1126,7 +1127,7 @@ OBJS-$(CONFIG_FITS_DEMUXER)            += fits.o
 OBJS-$(CONFIG_TAK_DEMUXER)             += tak.o
 
 # libavformat dependencies for static builds
-STLIBOBJS-$(CONFIG_AVFORMAT)           += h2645_parse.o lcevctab.o to_upper4.o
+STLIBOBJS-$(CONFIG_AVFORMAT)           += to_upper4.o
 STLIBOBJS-$(CONFIG_ISO_MEDIA)          += mpegaudiotabs.o
 STLIBOBJS-$(CONFIG_FLV_MUXER)          += mpeg4audio_sample_rates.o
 STLIBOBJS-$(CONFIG_HLS_DEMUXER)        += ac3_channel_layout_tab.o
@@ -1170,6 +1171,7 @@ OBJS-$(CONFIG_LIBAOM_AV1_DECODER)         += libaomdec.o libaom.o
 OBJS-$(CONFIG_LIBAOM_AV1_ENCODER)         += libaomenc.o libaom.o
 OBJS-$(CONFIG_LIBARIBB24_DECODER)         += libaribb24.o ass.o
 OBJS-$(CONFIG_LIBARIBCAPTION_DECODER)     += libaribcaption.o ass.o
+OBJS-$(CONFIG_LIBCELT_DECODER)            += libcelt_dec.o
 OBJS-$(CONFIG_LIBCODEC2_DECODER)          += libcodec2.o
 OBJS-$(CONFIG_LIBCODEC2_ENCODER)          += libcodec2.o
 OBJS-$(CONFIG_LIBDAV1D_DECODER)           += libdav1d.o av1_parse.o
@@ -1328,7 +1330,6 @@ SKIPHEADERS                            += %_tablegen.h                  \
                                           %_tables.h                    \
                                           tableprint.h                  \
                                           tableprint_vlc.h              \
-                                          aaccoder_nmr.h                \
                                           aaccoder_twoloop.h            \
                                           aaccoder_trellis.h            \
                                           aacenc_quantization.h         \
@@ -1366,7 +1367,6 @@ TESTPROGS = avcodec                                                     \
             bitstream_be                                                \
             bitstream_le                                                \
             codec_desc                                                  \
-            encinfo                                                     \
             htmlsubtitles                                               \
             jpeg2000dwt                                                 \
             mathops                                                    \
@@ -1379,6 +1379,7 @@ TESTPROGS-$(CONFIG_GOLOMB)                += golomb
 TESTPROGS-$(CONFIG_IDCTDSP)               += dct
 TESTPROGS-$(CONFIG_DXV_ENCODER)           += hashtable
 TESTPROGS-$(CONFIG_MJPEG_ENCODER)         += mjpegenc_huffman
+TESTPROGS-$(HAVE_MMX)                     += motion
 TESTPROGS-$(CONFIG_MPEGVIDEO)             += mpeg12framerate
 TESTPROGS-$(CONFIG_H264_METADATA_BSF)     += h264_levels
 TESTPROGS-$(CONFIG_HEVC_METADATA_BSF)     += h265_levels

libavcodec/aac/aacdec.c

@@ -676,7 +676,6 @@ ChannelElement *ff_aac_get_che(AACDecContext *ac, int type, int elem_id)
             ac->tags_mapped++;
             return ac->tag_che_map[type][elem_id] = ac->che[type][elem_id];
         }
-        av_fallthrough;
     case 13:
         if (ac->tags_mapped > 3 && ((type == TYPE_CPE && elem_id < 8) ||
                                     (type == TYPE_SCE && elem_id < 6) ||
@@ -684,20 +683,17 @@ ChannelElement *ff_aac_get_che(AACDecContext *ac, int type, int elem_id)
             ac->tags_mapped++;
             return ac->tag_che_map[type][elem_id] = ac->che[type][elem_id];
         }
-        av_fallthrough;
     case 12:
     case 7:
         if (ac->tags_mapped == 3 && type == TYPE_CPE) {
             ac->tags_mapped++;
             return ac->tag_che_map[TYPE_CPE][elem_id] = ac->che[TYPE_CPE][2];
         }
-        av_fallthrough;
     case 11:
         if (ac->tags_mapped == 3 && type == TYPE_SCE) {
             ac->tags_mapped++;
             return ac->tag_che_map[TYPE_SCE][elem_id] = ac->che[TYPE_SCE][1];
         }
-        av_fallthrough;
     case 6:
         /* Some streams incorrectly code 5.1 audio as
          * SCE[0] CPE[0] CPE[1] SCE[1]
@@ -715,13 +711,11 @@ ChannelElement *ff_aac_get_che(AACDecContext *ac, int type, int elem_id)
             ac->tags_mapped++;
             return ac->tag_che_map[type][elem_id] = ac->che[TYPE_LFE][0];
         }
-        av_fallthrough;
     case 5:
         if (ac->tags_mapped == 2 && type == TYPE_CPE) {
             ac->tags_mapped++;
             return ac->tag_che_map[TYPE_CPE][elem_id] = ac->che[TYPE_CPE][1];
         }
-        av_fallthrough;
     case 4:
         /* Some streams incorrectly code 4.0 audio as
          * SCE[0] CPE[0] LFE[0]
@@ -745,7 +739,6 @@ ChannelElement *ff_aac_get_che(AACDecContext *ac, int type, int elem_id)
             ac->tags_mapped++;
             return ac->tag_che_map[TYPE_SCE][elem_id] = ac->che[TYPE_SCE][1];
         }
-        av_fallthrough;
     case 3:
     case 2:
         if (ac->tags_mapped == (ac->oc[1].m4ac.chan_config != 2) &&
@@ -757,13 +750,11 @@ ChannelElement *ff_aac_get_che(AACDecContext *ac, int type, int elem_id)
             ac->tags_mapped++;
             return ac->tag_che_map[TYPE_SCE][elem_id] = ac->che[TYPE_SCE][1];
         }
-        av_fallthrough;
     case 1:
         if (!ac->tags_mapped && type == TYPE_SCE) {
             ac->tags_mapped++;
             return ac->tag_che_map[TYPE_SCE][elem_id] = ac->che[TYPE_SCE][0];
         }
-        av_fallthrough;
     default:
         return NULL;
     }
@@ -831,8 +822,8 @@ static int decode_pce(AVCodecContext *avctx, MPEG4AudioConfig *m4ac,
     sampling_index = get_bits(gb, 4);
     if (m4ac->sampling_index != sampling_index)
         av_log(avctx, AV_LOG_WARNING,
-               "Sample rate index (%d) in program config element does not "
-               "match the sample rate index (%d) configured by the container.\n", sampling_index, m4ac->sampling_index);
+               "Sample rate index in program config element does not "
+               "match the sample rate index configured by the container.\n");
 
     num_front       = get_bits(gb, 4);
     num_side        = get_bits(gb, 4);
@@ -875,91 +866,6 @@ static int decode_pce(AVCodecContext *avctx, MPEG4AudioConfig *m4ac,
         av_log(avctx, AV_LOG_ERROR, "decode_pce: " overread_err);
         return AVERROR_INVALIDDATA;
     }
-
-    // Height extension
-    int height_ext = 0;
-    if (comment_len >= 16 + (num_front * 2 + num_side * 2 + num_back * 2))
-        height_ext = show_bits(gb, 8) == 0xAC;
-    if (height_ext) {
-        uint8_t height_map[4 /* ChannelPosition */][16 /* Channel */]; // 0 == base, 1 == top, 2 == bottom.
-        uint8_t tag[6 /* ChannelPosition */ ][16 /* Channel */][3];
-        int i, invalid = 0, height_tags = 0;
-
-        skip_bits(gb, 8);
-
-        // Read height extension bits to height_map, which define which layer each element belongs to.
-        // Also make a copy of layout_map that will then be used to rearrange it.
-        for (i = 0; i < num_front; i++) {
-            int height = get_bits(gb, 2);
-            invalid |= height > 2;
-            height_map [AAC_CHANNEL_FRONT][i] = height;
-            memcpy(&tag[AAC_CHANNEL_FRONT][i], layout_map + height_tags, sizeof(*layout_map));
-            height_tags++;
-        }
-        for (i = 0; i < num_side; i++) {
-            int height = get_bits(gb, 2);
-            invalid |= height > 2;
-            height_map [AAC_CHANNEL_SIDE][i] = height;
-            memcpy(&tag[AAC_CHANNEL_SIDE][i], layout_map + height_tags, sizeof(*layout_map));
-            height_tags++;
-        }
-        for (i = 0; i < num_back; i++) {
-            int height = get_bits(gb, 2);
-            invalid |= height > 2;
-            height_map [AAC_CHANNEL_BACK][i] = height;
-            memcpy(&tag[AAC_CHANNEL_BACK][i], layout_map + height_tags, sizeof(*layout_map));
-            height_tags++;
-        }
-        for (i = 0; i < num_lfe; i++) {
-            memcpy(&tag[AAC_CHANNEL_LFE][i], layout_map + height_tags, sizeof(*layout_map));
-            height_tags++;
-        }
-        for (i = 0; i < num_cc; i++) {
-            memcpy(&tag[AAC_CHANNEL_CC][i], layout_map + height_tags, sizeof(*layout_map));
-            height_tags++;
-        }
-        av_assert0(height_tags == tags);
-
-        if (!invalid) {
-            height_tags = 0;
-            // For each height layer, check that an element belongs to it and copy it back to layout_map.
-            // We need to take into account LFE and CC elements that may be present.
-            for (i = 0; i < 3; i++) {
-                for (int j = 0; j < num_front; j++) {
-                    if (height_map[AAC_CHANNEL_FRONT][j] == i) {
-                        memcpy(layout_map + height_tags, &tag[AAC_CHANNEL_FRONT][j], sizeof(*layout_map));
-                        height_tags++;
-                    }
-                }
-                for (int j = 0; j < num_side; j++) {
-                    if (height_map[AAC_CHANNEL_SIDE][j] == i) {
-                        memcpy(layout_map + height_tags, &tag[AAC_CHANNEL_SIDE][j], sizeof(*layout_map));
-                        height_tags++;
-                    }
-                }
-                for (int j = 0; j < num_back; j++) {
-                    if (height_map[AAC_CHANNEL_BACK][j] == i) {
-                        memcpy(layout_map + height_tags, &tag[AAC_CHANNEL_BACK][j], sizeof(*layout_map));
-                        height_tags++;
-                    }
-                }
-                if (i == 0) { // Base height, copy LFE and CC elements before moving to Top and Bottom
-                    for (int j = 0; j < num_lfe; j++) {
-                        memcpy(layout_map + height_tags, &tag[AAC_CHANNEL_LFE][j], sizeof(*layout_map));
-                        height_tags++;
-                    }
-                    for (int j = 0; j < num_cc; j++) {
-                        memcpy(layout_map + height_tags, &tag[AAC_CHANNEL_CC][j], sizeof(*layout_map));
-                        height_tags++;
-                    }
-                }
-            }
-            av_assert0(height_tags == tags);
-        }
-
-        comment_len -= 8 + (num_front * 2 + num_side * 2 + num_back * 2);
-    }
-
     skip_bits_long(gb, comment_len);
     return tags;
 }
@@ -983,6 +889,12 @@ static int decode_ga_specific_config(AACDecContext *ac, AVCodecContext *avctx,
     int tags = 0;
 
     m4ac->frame_length_short = get_bits1(gb);
+    if (m4ac->frame_length_short && m4ac->sbr == 1) {
+      avpriv_report_missing_feature(avctx, "SBR with 960 frame length");
+      if (ac) ac->warned_960_sbr = 1;
+      m4ac->sbr = 0;
+      m4ac->ps = 0;
+    }
 
     if (get_bits1(gb))       // dependsOnCoreCoder
         skip_bits(gb, 14);   // coreCoderDelay
@@ -1334,7 +1246,7 @@ av_cold int ff_aac_decode_init(AVCodecContext *avctx)
         ac->oc[1].m4ac.chan_config = i;
 
         if (ac->oc[1].m4ac.chan_config) {
-            ret = ff_aac_set_default_channel_config(ac, avctx, layout_map,
+            int ret = ff_aac_set_default_channel_config(ac, avctx, layout_map,
                                                         &layout_map_tags,
                                                         ac->oc[1].m4ac.chan_config);
             if (!ret)
@@ -2034,11 +1946,17 @@ static int decode_extension_payload(AACDecContext *ac, GetBitContext *gb, int cn
     switch (type) { // extension type
     case EXT_SBR_DATA_CRC:
         crc_flag++;
-        av_fallthrough;
     case EXT_SBR_DATA:
         if (!che) {
             av_log(ac->avctx, AV_LOG_ERROR, "SBR was found before the first channel element.\n");
             return res;
+        } else if (ac->oc[1].m4ac.frame_length_short) {
+            if (!ac->warned_960_sbr)
+              avpriv_report_missing_feature(ac->avctx,
+                                            "SBR with 960 frame length");
+            ac->warned_960_sbr = 1;
+            skip_bits_long(gb, 8 * cnt - 4);
+            return res;
         } else if (!ac->oc[1].m4ac.sbr) {
             av_log(ac->avctx, AV_LOG_ERROR, "SBR signaled to be not-present but was found in the bitstream.\n");
             skip_bits_long(gb, 8 * cnt - 4);
@@ -2059,8 +1977,7 @@ static int decode_extension_payload(AACDecContext *ac, GetBitContext *gb, int cn
             ac->avctx->profile = AV_PROFILE_AAC_HE;
         }
 
-        ac->proc.sbr_decode_extension(ac, che, gb, crc_flag, cnt, elem_type,
-                                      ac->oc[1].m4ac.frame_length_short);
+        ac->proc.sbr_decode_extension(ac, che, gb, crc_flag, cnt, elem_type);
 
         if (ac->oc[1].m4ac.ps == 1 && !ac->warned_he_aac_mono) {
             av_log(ac->avctx, AV_LOG_VERBOSE, "Treating HE-AAC mono as stereo.\n");
@@ -2170,7 +2087,6 @@ static void spectral_to_sample(AACDecContext *ac, int samples)
                     }
                     if (ac->oc[1].m4ac.sbr > 0) {
                         ac->proc.sbr_apply(ac, che, type,
-                                           ac->oc[1].m4ac.frame_length_short,
                                            che->ch[0].output,
                                            che->ch[1].output);
                     }
@@ -2648,9 +2564,6 @@ static const AVOption options[] = {
       { "coded",    "order in which the channels are coded in the bitstream",
         0, AV_OPT_TYPE_CONST, { .i64 = CHANNEL_ORDER_CODED }, .flags = AACDEC_FLAGS, .unit = "channel_order" },
 
-    { "target_level", "Target output loudness in dBFS for xHE-AAC normalization (0 = disabled)",
-        OFF(target_level), AV_OPT_TYPE_INT, { .i64 = 0 }, -70, 0, AACDEC_FLAGS },
-
     {NULL},
 };
 

libavcodec/aac/aacdec.h

@@ -98,12 +98,6 @@ enum AACUSACLoudnessExt {
     UNIDRCLOUDEXT_EQ = 0x1,
 };
 
-enum AACUSACDRCExt {
-    UNIDRCCONFEXT_TERM = 0x0,
-    UNIDRCCONFEXT_PARAM_DRC = 0x1,
-    UNIDRCCONFEXT_V1 = 0x2,
-};
-
 // Supposed to be equal to AAC_RENAME() in case of USE_FIXED.
 #define RENAME_FIXED(name) name ## _fixed
 
@@ -308,7 +302,6 @@ typedef struct ChannelElement {
 
 typedef struct AACUSACLoudnessInfo {
     uint8_t drc_set_id : 6;
-    uint8_t eq_set_id : 6; /* loudnessInfoV1() only, 0 otherwise */
     uint8_t downmix_id : 7;
     struct {
         uint16_t lvl : 12;
@@ -384,13 +377,6 @@ typedef struct AACUsacElemConfig {
         uint32_t pl_data_offset;
         uint8_t *pl_buf;
     } ext;
-
-    struct {
-        struct {
-            int lower;
-            int upper;
-        } loudness;
-    } drc;
 } AACUsacElemConfig;
 
 typedef struct AACUSACConfig {
@@ -406,13 +392,6 @@ typedef struct AACUSACConfig {
         AACUSACLoudnessInfo album_info[64];
         uint8_t nb_info;
         AACUSACLoudnessInfo info[64];
-
-        /**
-         * Raw bsMethodValue (μ) of the program/anchor-loudness measurement
-         * selected for normalization at config time. -1 == none found.
-         * L_LKFS = -57.75 + 0.25 * input_method_val.
-         */
-        int input_method_val;
     } loudness;
 } AACUSACConfig;
 
@@ -454,9 +433,9 @@ typedef struct AACDecProc {
 
     int (*sbr_ctx_alloc_init)(AACDecContext *ac, ChannelElement **che, int id_aac);
     int (*sbr_decode_extension)(AACDecContext *ac, ChannelElement *che,
-                                GetBitContext *gb, int crc, int cnt, int id_aac, int fl960);
-    void (*sbr_apply)(AACDecContext *ac, ChannelElement *che, int id_aac, int fl960,
-                      void /* INTFLOAT */ *L, void /* INTFLOAT */ *R);
+                                GetBitContext *gb, int crc, int cnt, int id_aac);
+    void (*sbr_apply)(AACDecContext *ac, ChannelElement *che,
+                      int id_aac, void /* INTFLOAT */ *L, void /* INTFLOAT */ *R);
     void (*sbr_ctx_close)(ChannelElement *che);
 } AACDecProc;
 
@@ -576,15 +555,9 @@ struct AACDecContext {
 
     enum AACOutputChannelOrder output_channel_order;
 
-    /**
-     * Target output loudness in dBFS, used for xHE-AAC loudness normalization
-     * based on the parsed loudnessInfoSet() metadata. 0 disables normalization.
-     */
-    int target_level;
-    int warned_loudness_missing;
-
     OutputConfiguration oc[2];
     int warned_num_aac_frames;
+    int warned_960_sbr;
     unsigned warned_71_wide;
     int warned_gain_control;
     int warned_he_aac_mono;

libavcodec/aac/aacdec_tab.c

@@ -75,7 +75,7 @@ const uint8_t ff_aac_channel_layout_map[16][16][3] = {
 const int16_t ff_aac_channel_map[3][4][6] = {
     {
       { AV_CHAN_FRONT_CENTER,        AV_CHAN_FRONT_LEFT_OF_CENTER, AV_CHAN_FRONT_RIGHT_OF_CENTER, AV_CHAN_FRONT_LEFT,        AV_CHAN_FRONT_RIGHT,        AV_CHAN_NONE },
-      { AV_CHAN_UNUSED,              AV_CHAN_SIDE_LEFT,            AV_CHAN_SIDE_RIGHT,            AV_CHAN_NONE,              AV_CHAN_NONE,               AV_CHAN_NONE },
+      { AV_CHAN_UNUSED,              AV_CHAN_NONE,                 AV_CHAN_NONE,                  AV_CHAN_NONE,              AV_CHAN_NONE,               AV_CHAN_NONE },
       { AV_CHAN_UNUSED,              AV_CHAN_SIDE_LEFT,            AV_CHAN_SIDE_RIGHT,            AV_CHAN_BACK_LEFT,         AV_CHAN_BACK_RIGHT,         AV_CHAN_BACK_CENTER },
       { AV_CHAN_LOW_FREQUENCY,       AV_CHAN_LOW_FREQUENCY_2,      AV_CHAN_NONE,                  AV_CHAN_NONE,              AV_CHAN_NONE,               AV_CHAN_NONE },
     },

libavcodec/aac/aacdec_usac.c

@@ -49,7 +49,7 @@ static inline uint32_t get_escaped_value(GetBitContext *gb, int nb1, int nb2, in
     return val;
 }
 
-/* ISO/IEC 23003-3, Table 74: bsOutputChannelPos */
+/* ISO/IEC 23003-3, Table 74 — bsOutputChannelPos */
 static const enum AVChannel usac_ch_pos_to_av[64] = {
     [0] = AV_CHAN_FRONT_LEFT,
     [1] = AV_CHAN_FRONT_RIGHT,
@@ -85,23 +85,10 @@ static const enum AVChannel usac_ch_pos_to_av[64] = {
     [31] = AV_CHAN_TOP_SURROUND_RIGHT, ///< -110 degrees, Rvs, TpRS
 };
 
-/* ISO/IEC 23003-4, Table A.48: bit width of bsMethodValue depends on methodDef. */
-static int methodvalue_width(int method_def)
-{
-    switch (method_def) {
-    case 7: return 5; /* mixing level */
-    case 8: return 2; /* room type */
-    default: return 8; /* loudness (0..6, 9) + reserved */
-    }
-}
-
-/* ISO/IEC 23003-4, Table 58/60: loudnessInfo(), loudnessInfoV1().
- * The only difference in V1 is the added eqSetId field. */
 static int decode_loudness_info(AACDecContext *ac, AACUSACLoudnessInfo *info,
-                                GetBitContext *gb, int v1)
+                                GetBitContext *gb)
 {
     info->drc_set_id = get_bits(gb, 6);
-    info->eq_set_id = v1 ? get_bits(gb, 6) : 0;
     info->downmix_id = get_bits(gb, 7);
 
     if ((info->sample_peak.present = get_bits1(gb))) /* samplePeakLevelPresent */
@@ -116,8 +103,7 @@ static int decode_loudness_info(AACDecContext *ac, AACUSACLoudnessInfo *info,
     info->nb_measurements = get_bits(gb, 4);
     for (int i = 0; i < info->nb_measurements; i++) {
         info->measurements[i].method_def = get_bits(gb, 4);
-        info->measurements[i].method_val =
-            get_bits(gb, methodvalue_width(info->measurements[i].method_def));
+        info->measurements[i].method_val = get_unary(gb, 0, 8);
         info->measurements[i].measurement = get_bits(gb, 4);
         info->measurements[i].reliability = get_bits(gb, 2);
     }
@@ -125,56 +111,6 @@ static int decode_loudness_info(AACDecContext *ac, AACUSACLoudnessInfo *info,
     return 0;
 }
 
-/* ISO/IEC 23003-4, Table 61: loudnessInfoSetExtension(), UNIDRCLOUDEXT_EQ */
-static int decode_loudness_set_v1(AACDecContext *ac, AACUSACConfig *usac,
-                                  GetBitContext *gb)
-{
-    int ret;
-    int nb_album = get_bits(gb, 6); /* loudnessInfoV1AlbumCount */
-    int nb_info = get_bits(gb, 6); /* loudnessInfoV1Count */
-
-    for (int i = 0; i < nb_album; i++) {
-        AACUSACLoudnessInfo tmp;
-        ret = decode_loudness_info(ac, &tmp, gb, 1);
-        if (ret < 0)
-            return ret;
-        if (usac->loudness.nb_album < FF_ARRAY_ELEMS(usac->loudness.album_info))
-            usac->loudness.album_info[usac->loudness.nb_album++] = tmp;
-    }
-
-    for (int i = 0; i < nb_info; i++) {
-        AACUSACLoudnessInfo tmp;
-        ret = decode_loudness_info(ac, &tmp, gb, 1);
-        if (ret < 0)
-            return ret;
-        if (usac->loudness.nb_info < FF_ARRAY_ELEMS(usac->loudness.info))
-            usac->loudness.info[usac->loudness.nb_info++] = tmp;
-    }
-
-    return 0;
-}
-
-/* Pick the bsMethodValue of a program- or anchor-loudness measurement.
- * Per ISO/IEC 23003-4 6.1.2.5, downmixId, drcSetId and eqSetId identify the
- * signal a loudnessInfo() applies to; only downmixId == 0 (base layout)
- * together with drcSetId == 0 and eqSetId == 0 (no DRC/EQ) describes the
- * unprocessed signal we output, so measurements for any other
- * downmix/DRC/EQ set must not be used. */
-static int select_loudness_measurement(const AACUSACConfig *usac)
-{
-    for (int i = 0; i < usac->loudness.nb_info; i++) {
-        const AACUSACLoudnessInfo *info = &usac->loudness.info[i];
-        if (info->downmix_id != 0 || info->drc_set_id != 0 || info->eq_set_id != 0)
-            continue;
-        for (int j = 0; j < info->nb_measurements; j++) {
-            int method = info->measurements[j].method_def;
-            if (method == 1 || method == 2)
-                return info->measurements[j].method_val;
-        }
-    }
-    return -1;
-}
-
 static int decode_loudness_set(AACDecContext *ac, AACUSACConfig *usac,
                                GetBitContext *gb)
 {
@@ -184,13 +120,13 @@ static int decode_loudness_set(AACDecContext *ac, AACUSACConfig *usac,
     usac->loudness.nb_info = get_bits(gb, 6); /* loudnessInfoCount */
 
     for (int i = 0; i < usac->loudness.nb_album; i++) {
-        ret = decode_loudness_info(ac, &usac->loudness.album_info[i], gb, 0);
+        ret = decode_loudness_info(ac, &usac->loudness.album_info[i], gb);
         if (ret < 0)
             return ret;
     }
 
     for (int i = 0; i < usac->loudness.nb_info; i++) {
-        ret = decode_loudness_info(ac, &usac->loudness.info[i], gb, 0);
+        ret = decode_loudness_info(ac, &usac->loudness.info[i], gb);
         if (ret < 0)
             return ret;
     }
@@ -198,25 +134,16 @@ static int decode_loudness_set(AACDecContext *ac, AACUSACConfig *usac,
     if (get_bits1(gb)) { /* loudnessInfoSetExtPresent */
         enum AACUSACLoudnessExt type;
         while ((type = get_bits(gb, 4)) != UNIDRCLOUDEXT_TERM) {
-            uint8_t size_bits = get_bits(gb, 4) + 4; /* bitSizeLen */
-            uint32_t bit_size = get_bits_long(gb, size_bits) + 1; /* bitSize */
-            int start = get_bits_count(gb);
-            int skip;
+            uint8_t size_bits = get_bits(gb, 4) + 4;
+            uint8_t bit_size = get_bits(gb, size_bits) + 1;
             switch (type) {
             case UNIDRCLOUDEXT_EQ:
-                ret = decode_loudness_set_v1(ac, usac, gb);
-                if (ret < 0)
-                    return ret;
-                break;
+                avpriv_report_missing_feature(ac->avctx, "loudnessInfoV1");
+                return AVERROR_PATCHWELCOME;
             default:
-                break;
+                for (int i = 0; i < bit_size; i++)
+                    skip_bits1(gb);
             }
-            /* The extension size is explicit, so unparsed (or unknown)
-             * data can be skipped without desynchronizing. */
-            skip = bit_size - (get_bits_count(gb) - start);
-            if (skip < 0)
-                return AVERROR_INVALIDDATA;
-            skip_bits_long(gb, skip);
         }
     }
 
@@ -300,7 +227,7 @@ static int decode_usac_element_pair(AACDecContext *ac,
         e->mps.phase_coding = get_bits1(gb); /* bsPhaseCoding */
 
         e->mps.otts_bands_phase_present = get_bits1(gb);
-        int otts_bands_phase = ((int[]){0,10,10,7,5,3,2,2})[e->mps.freq_res]; // Table 109:  Default value of bsOttBandsPhase
+        int otts_bands_phase = ((int[]){0,10,10,7,5,3,2,2})[e->mps.freq_res]; // Table 109 — Default value of bsOttBandsPhase
         if (e->mps.otts_bands_phase_present) { /* bsOttBandsPhasePresent */
             otts_bands_phase = get_bits(gb, 5); /* bsOttBandsPhase */
             if (otts_bands_phase > numBands)
@@ -326,114 +253,6 @@ static int decode_usac_element_pair(AACDecContext *ac,
     return 0;
 }
 
-/* ISO/IEC 23003-4, Table 62: channelLayout() */
-static int decode_drc_channel_layout(GetBitContext *gb)
-{
-    int base_channel_count = get_bits(gb, 7); /* baseChannelCount */
-    if (get_bits1(gb)) { /* layoutSignallingPresent */
-        if (get_bits(gb, 8) == 0) /* definedLayout == 0 */
-            for (int i = 0; i < base_channel_count; i++)
-                skip_bits(gb, 7); /* speakerPosition */
-    }
-    return base_channel_count;
-}
-
-/* ISO/IEC 23003-4, Table 63: downmixInstructions() */
-static void skip_drc_downmix_instructions(GetBitContext *gb, int base_channel_count)
-{
-    int target_channel_count;
-    skip_bits(gb, 7); /* downmixId */
-    target_channel_count = get_bits(gb, 7); /* targetChannelCount */
-    skip_bits(gb, 8); /* targetLayout */
-    if (get_bits1(gb)) /* downmixCoefficientsPresent */
-        skip_bits_long(gb, 4 * target_channel_count * base_channel_count);
-}
-
-/* ISO/IEC 23003-4, Table 70: drcInstructionsBasic(), common with the
- * uniDrc variant up to the loudness-target fields. */
-static void decode_drc_instructions_basic(AACUsacElemConfig *e, GetBitContext *gb)
-{
-    int set_effects;
-
-    skip_bits(gb, 6); /* drcSetId */
-    skip_bits(gb, 4); /* drcLocation */
-    skip_bits(gb, 7); /* downmixId */
-    if (get_bits1(gb)) { /* additionalDownmixIdPresent */
-        int add_downmix_cnt = get_bits(gb, 3); /* additionalDownmixIdCount */
-        for (int j = 0; j < add_downmix_cnt; j++)
-            skip_bits(gb, 7); /* additionalDownmixId */
-    }
-
-    set_effects = get_bits(gb, 16); /* drcSetEffect */
-    if ((set_effects & (3 << 10)) == 0) {
-        if (get_bits1(gb)) /* limiterPeakTargetPresent */
-            skip_bits(gb, 8); /* bsLimiterPeakTarget */
-    }
-
-    if (get_bits1(gb)) { /* drcSetTargetLoudnessPresent */
-        e->drc.loudness.upper = get_bits(gb, 6); /* bsDrcSetTargetLoudnessValueUpper */
-        if (get_bits1(gb)) /* drcSetTargetLoudnessValueLowerPresent */
-            e->drc.loudness.lower = get_bits(gb, 6); /* bsDrcSetTargetLoudnessValueLower */
-    }
-}
-
-/* ISO/IEC 23003-4, Table 57: uniDrcConfig() */
-static int decode_drc_config(AACDecContext *ac, AACUsacElemConfig *e,
-                             GetBitContext *gb)
-{
-    int nb_downmix_instr, nb_coeff_basic = 0, nb_instr_basic = 0;
-    int nb_coeff_uni, nb_instr_uni;
-    int base_channel_count;
-
-    e->drc.loudness.lower = -1;
-    e->drc.loudness.upper = -1;
-
-    if (get_bits1(gb)) /* sampleRatePresent */
-        skip_bits(gb, 18); /* bsSampleRate */
-
-    nb_downmix_instr = get_bits(gb, 7); /* downmixInstructionsCount */
-
-    if (get_bits1(gb)) { /* drcDescriptionBasicPresent */
-        nb_coeff_basic = get_bits(gb, 3); /* drcCoefficientsBasicCount */
-        nb_instr_basic = get_bits(gb, 4); /* drcInstructionsBasicCount */
-    }
-
-    nb_coeff_uni = get_bits(gb, 3); /* drcCoefficientsUniDrcCount */
-    nb_instr_uni = get_bits(gb, 6); /* drcInstructionsUniDrcCount */
-
-    if (nb_coeff_uni || nb_instr_uni) {
-        avpriv_report_missing_feature(ac->avctx,
-                                      "AAC USAC uniDrc DRC processing");
-        return AVERROR_PATCHWELCOME;
-    }
-
-    base_channel_count = decode_drc_channel_layout(gb);
-
-    for (int i = 0; i < nb_downmix_instr; i++)
-        skip_drc_downmix_instructions(gb, base_channel_count);
-
-    for (int i = 0; i < nb_coeff_basic; i++)
-        skip_bits(gb, 4 + 7); /* drcLocation, drcCharacteristic */
-
-    for (int i = 0; i < nb_instr_basic; i++)
-        decode_drc_instructions_basic(e, gb);
-
-    if (get_bits1(gb)) { /* uniDrcConfigExtPresent */
-        enum AACUSACDRCExt type;
-        while ((type = get_bits(gb, 4)) != UNIDRCCONFEXT_TERM) {
-            uint8_t size_bits = get_bits(gb, 4) + 4; /* bitSizeLen */
-            uint32_t bit_size = get_bits_long(gb, size_bits) + 1; /* extBitSize */
-            switch (type) {
-            default:
-                skip_bits_long(gb, bit_size);
-                break;
-            }
-        }
-    }
-
-    return 0;
-}
-
 static int decode_usac_extension(AACDecContext *ac, AACUsacElemConfig *e,
                                  GetBitContext *gb)
 {
@@ -457,25 +276,9 @@ static int decode_usac_extension(AACDecContext *ac, AACUsacElemConfig *e,
         break;
     case ID_EXT_ELE_SAOC:
         break;
-#endif
-    case ID_EXT_ELE_UNI_DRC: {
-        int start = get_bits_count(gb);
-        int ret = decode_drc_config(ac, e, gb);
-        int skip = 8*ext_config_len - (get_bits_count(gb) - start);
-        if (ret == AVERROR_PATCHWELCOME) {
-            /* Unsupported uniDrcConfig(): ignore the DRC metadata and treat
-             * the element as fill so the stream stays decodable. */
-            e->ext.type = ID_EXT_ELE_FILL;
-            ret = 0;
-        }
-        if (ret < 0)
-            return ret;
-        if (skip < 0)
-            return AVERROR_INVALIDDATA;
-        /* The config is byte-padded to usacExtElementConfigLength */
-        skip_bits_long(gb, skip);
+    case ID_EXT_ELE_UNI_DRC:
         break;
-    }
+#endif
     case ID_EXT_ELE_FILL:
         break; /* This is what the spec does */
     case ID_EXT_ELE_AUDIOPREROLL:
@@ -570,7 +373,6 @@ int ff_aac_usac_config_decode(AACDecContext *ac, AVCodecContext *avctx,
         return AVERROR_PATCHWELCOME;
 
     memset(usac, 0, sizeof(*usac));
-    usac->loudness.input_method_val = -1;
 
     freq_idx = get_bits(gb, 5); /* usacSamplingFrequencyIndex */
     if (freq_idx == 0x1f) {
@@ -770,13 +572,6 @@ int ff_aac_usac_config_decode(AACDecContext *ac, AVCodecContext *avctx,
 
     ac->avctx->profile = AV_PROFILE_AAC_USAC;
 
-    usac->loudness.input_method_val = select_loudness_measurement(usac);
-    if (usac->loudness.input_method_val >= 0)
-        av_log(avctx, AV_LOG_VERBOSE,
-               "USAC input loudness: %.2f LKFS (bsMethodValue=%d)\n",
-               -57.75f + 0.25f * usac->loudness.input_method_val,
-               usac->loudness.input_method_val);
-
     ret = ff_aac_usac_reset_state(ac, oc);
     if (ret < 0)
         return ret;
@@ -1549,13 +1344,13 @@ static void spectrum_decode(AACDecContext *ac, AACUSACConfig *usac,
 
     /* Save coefficients and alpha values for prediction reasons */
     if (nb_channels > 1) {
-        AACUsacStereo *us2 = &cpe->us;
+        AACUsacStereo *us = &cpe->us;
         for (int ch = 0; ch < nb_channels; ch++) {
             SingleChannelElement *sce = &cpe->ch[ch];
             memcpy(sce->prev_coeffs, sce->coeffs, sizeof(sce->coeffs));
         }
-        memcpy(us2->prev_alpha_q_re, us2->alpha_q_re, sizeof(us2->alpha_q_re));
-        memcpy(us2->prev_alpha_q_im, us2->alpha_q_im, sizeof(us2->alpha_q_im));
+        memcpy(us->prev_alpha_q_re, us->alpha_q_re, sizeof(us->alpha_q_re));
+        memcpy(us->prev_alpha_q_im, us->alpha_q_im, sizeof(us->alpha_q_im));
     }
 
     for (int ch = 0; ch < nb_channels; ch++) {
@@ -1877,7 +1672,7 @@ static int decode_usac_core_coder(AACDecContext *ac, AACUSACConfig *usac,
     spectrum_decode(ac, usac, che, core_nb_channels);
 
     if (ac->oc[1].m4ac.sbr > 0) {
-        ac->proc.sbr_apply(ac, che, nb_channels == 2 ? TYPE_CPE : TYPE_SCE, 0,
+        ac->proc.sbr_apply(ac, che, nb_channels == 2 ? TYPE_CPE : TYPE_SCE,
                            che->ch[0].output,
                            che->ch[1].output);
     }
@@ -1941,8 +1736,8 @@ static int parse_audio_preroll(AACDecContext *ac, GetBitContext *gb)
         }
 
         /* Byte alignment is not guaranteed. */
-        for (int j = 0; j < au_len; j++)
-            tmp_buf[j] = get_bits(gb, 8);
+        for (int i = 0; i < au_len; i++)
+            tmp_buf[i] = get_bits(gb, 8);
 
         ret = init_get_bits8(&gbc, tmp_buf, au_len);
         if (ret < 0)
@@ -2033,10 +1828,6 @@ static int parse_ext_ele(AACDecContext *ac, AACUsacElemConfig *e,
         case ID_EXT_ELE_AUDIOPREROLL:
             ret = parse_audio_preroll(ac, gb2);
             break;
-        case ID_EXT_ELE_UNI_DRC:
-            /* uniDrcGain() payload: DRC is not applied, just consume the
-             * bits via skip_bits_long below. */
-            break;
         default:
             /* This should never happen */
             av_assert0(0);
@@ -2171,28 +1962,6 @@ int ff_aac_usac_decode_frame(AVCodecContext *avctx, AACDecContext *ac,
         *got_frame_ptr = 0;
     }
 
-    if (samples && ac->target_level) {
-        int method_val = usac->loudness.input_method_val;
-        if (method_val < 0) {
-            if (!ac->warned_loudness_missing) {
-                av_log(avctx, AV_LOG_WARNING,
-                       "target_level set but no program/anchor loudness "
-                       "measurement available; normalization skipped\n");
-                ac->warned_loudness_missing = 1;
-            }
-        } else {
-            /* Per ISO/IEC 23003-4 Table A.48: L = -57.75 + 0.25 * μ */
-            float input_loudness = -57.75f + 0.25f * method_val;
-            float gain_dB = (float)ac->target_level - input_loudness;
-            float gain = powf(10.0f, gain_dB / 20.0f);
-
-            for (int ch = 0; ch < frame->ch_layout.nb_channels; ch++)
-                ac->fdsp->vector_fmul_scalar((float *)frame->extended_data[ch],
-                                             (float *)frame->extended_data[ch],
-                                             gain, frame->nb_samples);
-        }
-    }
-
     /* for dual-mono audio (SCE + SCE) */
     is_dmono = ac->dmono_mode && elem_id[0] == 2 &&
                !av_channel_layout_compare(&ac->oc[1].ch_layout,

libavcodec/aac/aacdec_usac_mps212.c

@@ -240,7 +240,7 @@ static void huff_data_2d(GetBitContext *gb, int16_t *part0_data[2], int16_t (*da
                    0, 2*esc_cnt, 0, (2*lav + 1));
         for (i = 0; i < esc_cnt; i++) {
             data[esc_idx[i]][0] = esc_data[0][i] - lav;
-            data[esc_idx[i]][1] = esc_data[1][i] - lav;
+            data[esc_idx[i]][0] = esc_data[0][i] - lav;
         }
     }
 }
@@ -858,7 +858,7 @@ int ff_aac_map_index_data(AACMPSLosslessData *ld,
     for (int i = 0; i < nb_param_sets; i++) {
         if (ld->coarse_quant_no[i] == 1) {
             coarse_to_fine(tmp_idx_data[i], data_type, start_band,
-                           stop_band);
+                           stop_band - start_band);
             ld->coarse_quant_no[i] = 0;
         }
     }

libavcodec/aaccoder.c

@@ -59,7 +59,6 @@
 #define NOISE_LAMBDA_REPLACE 1.948f
 
 #include "libavcodec/aaccoder_trellis.h"
-#include "libavcodec/aaccoder_nmr.h"
 
 typedef float (*quantize_and_encode_band_func)(struct AACEncContext *s, PutBitContext *pb,
                                                const float *in, float *quant, const float *scaled,
@@ -503,12 +502,25 @@ static void search_for_pns(AACEncContext *s, AVCodecContext *avctx, SingleChanne
     const float dist_bias = av_clipf(4.f * 120 / lambda, 0.25f, 4.0f);
     const float pns_transient_energy_r = FFMIN(0.7f, lambda / 140.f);
 
-    int prev = -1000, prev_sf = -1;
+    int refbits = avctx->bit_rate * 1024.0 / avctx->sample_rate
+        / ((avctx->flags & AV_CODEC_FLAG_QSCALE) ? 2.0f : avctx->ch_layout.nb_channels)
+        * (lambda / 120.f);
+
+    /** Keep this in sync with twoloop's cutoff selection */
+    float rate_bandwidth_multiplier = 1.5f;
+    int prev = -1000, prev_sf = -1;
+    int frame_bit_rate = (avctx->flags & AV_CODEC_FLAG_QSCALE)
+        ? (refbits * rate_bandwidth_multiplier * avctx->sample_rate / 1024)
+        : (avctx->bit_rate / avctx->ch_layout.nb_channels);
+
+    frame_bit_rate *= 1.15f;
+
+    if (avctx->cutoff > 0) {
+        bandwidth = avctx->cutoff;
+    } else {
+        bandwidth = FFMAX(3000, AAC_CUTOFF_FROM_BITRATE(frame_bit_rate, 1, avctx->sample_rate));
+    }
 
-    /* PNS candidacy must use the coder's actual coding bandwidth (s->bandwidth,
-     * fixed at init), not a separate heuristic, or it evaluates a different band
-     * range than the coder later codes. */
-    bandwidth = s->bandwidth;
     cutoff = bandwidth * 2 * wlen / avctx->sample_rate;
 
     memcpy(sce->band_alt, sce->band_type, sizeof(sce->band_type));
@@ -627,10 +639,24 @@ static void mark_pns(AACEncContext *s, AVCodecContext *avctx, SingleChannelEleme
     const float spread_threshold = FFMIN(0.75f, NOISE_SPREAD_THRESHOLD*FFMAX(0.5f, lambda/100.f));
     const float pns_transient_energy_r = FFMIN(0.7f, lambda / 140.f);
 
-    /* PNS candidacy must use the coder's actual coding bandwidth (s->bandwidth,
-     * fixed at init), not a separate heuristic, or it evaluates a different band
-     * range than the coder later codes (NMR relies on this output directly). */
-    bandwidth = s->bandwidth;
+    int refbits = avctx->bit_rate * 1024.0 / avctx->sample_rate
+        / ((avctx->flags & AV_CODEC_FLAG_QSCALE) ? 2.0f : avctx->ch_layout.nb_channels)
+        * (lambda / 120.f);
+
+    /** Keep this in sync with twoloop's cutoff selection */
+    float rate_bandwidth_multiplier = 1.5f;
+    int frame_bit_rate = (avctx->flags & AV_CODEC_FLAG_QSCALE)
+        ? (refbits * rate_bandwidth_multiplier * avctx->sample_rate / 1024)
+        : (avctx->bit_rate / avctx->ch_layout.nb_channels);
+
+    frame_bit_rate *= 1.15f;
+
+    if (avctx->cutoff > 0) {
+        bandwidth = avctx->cutoff;
+    } else {
+        bandwidth = FFMAX(3000, AAC_CUTOFF_FROM_BITRATE(frame_bit_rate, 1, avctx->sample_rate));
+    }
+
     cutoff = bandwidth * 2 * wlen / avctx->sample_rate;
 
     memcpy(sce->band_alt, sce->band_type, sizeof(sce->band_type));
@@ -841,17 +867,4 @@ const AACCoefficientsEncoder ff_aac_coders[AAC_CODER_NB] = {
         search_for_ms,
         ff_aac_search_for_is,
     },
-    [AAC_CODER_NMR] = {
-        search_for_quantizers_nmr,
-        codebook_trellis_rate,
-        quantize_and_encode_band,
-        ff_aac_encode_tns_info,
-        ff_aac_apply_tns,
-        set_special_band_scalefactors,
-        NULL,                    /* PNS decided in the trellis (search_for_quantizers_nmr) */
-        mark_pns,
-        ff_aac_search_for_tns,
-        NULL,
-        NULL,
-    },
 };

libavcodec/aaccoder_nmr.h

@@ -1,700 +0,0 @@
-/*
- * AAC encoder NMR (noise-to-mask ratio) scalefactor coder
- * Copyright (c) 2026 Lynne <dev@lynne.ee>
- *
- * This file is part of FFmpeg.
- *
- * FFmpeg is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * FFmpeg is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
- */
-
-/**
- * AAC encoder NMR scalefactor coder.
- *
- * Optimizes the same noise-to-mask objective as the two-loop coder, but with an
- * optimal Viterbi search over scalefactors instead of a heuristic loop. For each
- * coded band the per-scalefactor distortion/bits curve is precomputed, then a
- * trellis over the (window-group, band) coding sequence minimizes
- *   sum_g = dist_g(sf_g)/threshold_g +
- *           lambda * (spectral_bits_g(sf_g) + scalefactor_differential_bits)
- * with |sf_g - sf_{g-1}| <= SCALE_MAX_DIFF as a constraint, and lambda
- * binary-searched so the coded size meets the per-frame bit budget
- *
- * Perceptual noise substitution (PNS) is integrated into the same objective: once
- * the trellis settles on its operating lambda, each noise-like band (flagged by
- * mark_pns) is offered a terminal "code as noise" candidate whose cost is
- * nmr_pns + lambda*NMR_PNS_BITS. Because NMR_PNS_BITS is far below a band's spectral bit
- * count, this candidate only wins when lambda is large, i.e. when the encoder is
- * struggling to hold the bitrate. The bits freed by the chosen PNS bands are
- * then re-spent by a second trellis pass over the remaining bands.
- */
-
-#ifndef AVCODEC_AACCODER_NMR_H
-#define AVCODEC_AACCODER_NMR_H
-
-#include <float.h>
-#include <string.h>
-#include "libavutil/mathematics.h"
-#include "mathops.h"
-#include "avcodec.h"
-#include "put_bits.h"
-#include "aac.h"
-#include "aacenc.h"
-#include "aactab.h"
-#include "aacenctab.h"
-
-/* differential scalefactor coding cost, clamped to the legal delta range */
-#define NMR_SFBITS(d) ff_aac_scalefactor_bits[av_clip((d) + SCALE_DIFF_ZERO, 0, 2*SCALE_MAX_DIFF)]
-
-#define NMR_ITERS  14 /* lambda binary-search iters */
-#define NMR_IFINE    9 /* fine-pass lambda iters */
-#define NMR_CITERS   7 /* coarse-pass lambda iters */
-#define NMR_CWARM    5 /* coarse-pass iters when warm-started off the previous frame's
-                        * lambda: the bracket spans 10 octaves instead of ~43, so fewer
-                        * bisection steps reach the same resolution */
-#define NMR_COARSE   8 /* two-pass coarse->fine grid step, cuts the Viterbi ncand^2 with no
-                        * quality loss, 0 disables it (single full-resolution pass) */
-#define NMR_STEP     1 /* fine-pass scalefactor candidate granularity */
-
-#define NMR_PNS_BITS 9 /* approx cost in bits of signalling PNS */
-
-/* Spectral-hole fill: noise-like bands the trellis left mostly empty are filled with
- * energy-matched noise (PNS); an audible hole sounds worse than matched noise. */
-#define NMR_PNS_HOLE_FRAC   0.5f
-#define NMR_PNS_HOLE_SPREAD 0.5f
-
-/* RC servo gain: scale the corridor centre by exp2(-K*fill/R) each frame to hold
- * the long-run mean rate; without it a bad centre drifts for dozens of frames. */
-#define NMR_RC_K_CBR 0.5f
-
-#define NMR_RC_ITERS 8 /* lambda bisection iters when clamping an over-cap frame */
-/* Corridor: bisect within [lam_rc/NMR_RC_CORR, lam_rc*NMR_RC_CORR] so quality stays
- * smooth while per-frame demand is tracked; 1.5 cuts lambda jitter ~25%. */
-#define NMR_RC_CORR   1.5f
-
-/* Leaky-bucket half-depth (bits/ch); 512 is the sweet spot — tighter rebounds as
- * frames cannot hit the narrow window. Clamped to the 6144 bits/ch decoder buffer. */
-#define NMR_CBR_BUF   512
-#define NMR_RC_CITERS 3 /* corridor coarse-pass iters */
-
-/* Transient bit-burst: an isolated onset (preceded by >= NMR_BURST_GAP long frames)
- * is coded NMR_BURST_GAIN x finer, held uniform across the run, repaid from steady stretches. */
-#define NMR_BURST_GAP   10
-#define NMR_BURST_GAIN  8.0f
-#define NMR_RC_FITERS 4 /* corridor fine-pass iters */
-#define NMR_RC_TRACK  0.1f /* per-frame pull of the corridor centre toward the realized lambda */
-
-/* PNS noise-distortion gate: only bands coded well above the masking floor become noise. */
-#define NMR_PNS_NDGATE 4.0f
-
-/* Energy/threshold cap for PNS: loud bands (energy >> mask) yield clipping random peaks;
- * only near-masked bands are safe substitution targets. */
-#define NMR_PNS_MAX_ET 8.0f
-
-/* Operating-lambda floor for PNS: below it the encoder is not struggling, so
- * substituting real texture for 9 signalling bits is net-negative. */
-#define NMR_PNS_LAM 100.0f
-
-/**
- * Viterbi over the coding sequence act[0..nact-1] (indices into the per-band
- * curves nd/nb), with lambda binary-searched so the coded size ~ destbits.
- * Fills chosen[band] for every band referenced by act. Returns the operating
- * lambda. node cost = dist/threshold + lambda*spectral_bits;
- * edge cost = lambda*sf_differential_bits; |delta sf| <= SCALE_MAX_DIFF hard.
- */
-static float nmr_solve(AACEncContext *s,
-                       const float (*nd)[NMR_NCAND], const int (*nb)[NMR_NCAND],
-                       const int *blo, const int *bnc, int step,
-                       const int *act, int nact, int destbits, int *chosen,
-                       float lo_l, float hi_l, int iters)
-{
-    float dp[NMR_NCAND], dpp[NMR_NCAND], node[NMR_NCAND];
-    float lamsf[2*SCALE_MAX_DIFF + 1];   /* lam*sfdiff bit cost, per lambda */
-    uint8_t bp[128][NMR_NCAND];
-    float lam = 1.0f;
-
-    if (nact <= 0)
-        return lam;
-
-    for (int it = 0; it < iters; it++) {
-        lam = sqrtf(lo_l * hi_l);
-        for (int i = 0; i <= 2*SCALE_MAX_DIFF; i++)
-            lamsf[i] = lam * ff_aac_scalefactor_bits[i];   /* edge cost for this lambda */
-
-        int b0 = act[0];
-        for (int o = 0; o < bnc[b0]; o++)
-            dp[o] = nd[b0][o] + lam * nb[b0][o];   /* anchor band node cost */
-
-        for (int k = 1; k < nact; k++) {
-            int b = act[k], pb = act[k-1];
-            memcpy(dpp, dp, sizeof(dp));
-            for (int o = 0; o < bnc[b]; o++)
-                node[o] = nd[b][o] + lam * nb[b][o];
-            /* dp[o] = node[o] + min_op(dpp[op] + edge cost) */
-            s->aacdsp.nmr_trellis_step(dp, bp[k], dpp, node, lamsf,
-                                       bnc[b], bnc[pb], blo[b] - blo[pb], step,
-                                       SCALE_MAX_DIFF);
-        }
-
-        /* backtrack */
-        int beo = 0, b = act[nact-1];
-        float bec = FLT_MAX;
-        for (int o = 0; o < bnc[b]; o++)
-            if (dp[o] < bec) { bec = dp[o]; beo = o; }
-        chosen[b] = beo;
-        for (int k = nact-1; k > 0; k--)
-            chosen[act[k-1]] = bp[k][chosen[act[k]]];
-
-        /* calc cost */
-        int total = 0;
-        for (int k = 0; k < nact; k++)
-            total += nb[act[k]][chosen[act[k]]];
-        for (int k = 1; k < nact; k++)
-            total += NMR_SFBITS((blo[act[k]]+chosen[act[k]]*step) - (blo[act[k-1]]+chosen[act[k-1]]*step));
-
-        if (it == iters - 1)
-            break;
-
-        /* check if we went over budget, go coarser if we did */
-        if (total > destbits)
-            lo_l = lam;
-        else
-            hi_l = lam;
-    }
-    return lam;
-}
-
-/* Build one coded band's (dist/threshold, bits) cost curve, candidates sf = lo + o*step
- * for o in [0,maxn), stopping when the band would drop (cb <= 0). Returns the bit count. */
-static int nmr_band_curve(AACEncContext *s, SingleChannelElement *sce, int w, int g,
-                          int start, int lo, int step, int maxn, float invthr,
-                          float maxval, float *nd_row, int *nb_row)
-{
-    int ncand = 0;
-    for (int o = 0; o < maxn && lo + o*step <= SCALE_MAX_POS; o++) {
-        int sf = lo + o*step, btot = 0, cb = find_min_book(maxval, sf);
-        float dist = 0.0f;
-        if (cb <= 0)
-            break;
-        for (int w2 = 0; w2 < sce->ics.group_len[w]; w2++) {
-            int bb;
-            dist += quantize_band_cost_cached(s, w + w2, g, sce->coeffs + start + w2*128,
-                                              s->scoefs + start + w2*128, sce->ics.swb_sizes[g],
-                                              sf, cb, 1.0f, INFINITY, &bb, NULL, 0);
-            btot += bb;
-        }
-        nd_row[ncand] = (dist - btot) * invthr;
-        nb_row[ncand] = btot;
-        ncand++;
-    }
-    return ncand;
-}
-
-static void search_for_quantizers_nmr(AVCodecContext *avctx,
-                                      AACEncContext *s,
-                                      SingleChannelElement *sce,
-                                      const float lambda)
-{
-    int bch = ((avctx->flags & AV_CODEC_FLAG_QSCALE) ? 2.0f : avctx->ch_layout.nb_channels);
-    int destbits = avctx->bit_rate * 1024.0 / avctx->sample_rate / bch * (lambda / 120.f);
-    int allz = 0, cutoff = 1024, nbnd = 0;
-
-    float thr[128];                 /* allocation-law effective threshold (drives the trellis) */
-    float thr_real[128];            /* real masking threshold (perceptual gates: PNS) */
-    float pener[128];               /* band energy (for PNS noise target)  */
-    float pspread[128];             /* band tonality spread (1 = noise)     */
-    int   minsf[128];
-    float maxvals[128];
-
-    /* coded-band trellis state (indexed 0..nbnd-1) */
-    int bidx[128];                  /* sce band index (w*16+g) */
-    int bw[128], bg[128], bst[128]; /* window group, swb, coef start per coded band */
-    int blo[128];                   /* finest candidate scalefactor */
-    int bnc[128];                   /* number of candidates */
-    int chosen[128];
-    int act[128];                   /* active (non-PNS) band coding order */
-    uint8_t is_pns[128];            /* trellis band coded as noise */
-
-    float (*nd)[NMR_NCAND] = s->nmr->nd; /* dist / threshold per candidate (heap) */
-    int   (*nb)[NMR_NCAND] = s->nmr->nb; /* spectral bits per candidate (heap)    */
-
-    /* two-pass coarse->fine grid step (see NMR_COARSE), the lambda search runs on
-     * the cheap coarse grid, PASS 2 refines the winner at NMR_STEP granularity */
-    const int cstep = NMR_COARSE > 0 ? NMR_COARSE : NMR_STEP;
-
-    s->nmr->counted[s->cur_channel] = 0;
-
-    /* Global-lambda RC: one solve per frame at a servoed centre lambda; the reservoir
-     * holds the long-run mean rate. Bypassed for VBR (-q:a) and the bootstrap frame. */
-    int rc_eligible = !(avctx->flags & AV_CODEC_FLAG_QSCALE) && avctx->bit_rate > 0 &&
-                      avctx->bit_rate_tolerance != 0;
-    /* Leaky-bucket reservoir: rc_fill (signed +-rc_bmax); the spend-floor/cap below force
-     * lambda so no frame banks past +rc_bmax or borrows past -rc_bmax. */
-    int rc_rate_frame = avctx->bit_rate * 1024.0 / avctx->sample_rate;
-    int rc_bmax = FFMIN(FFMAX(6144 * s->channels - rc_rate_frame, 256), NMR_CBR_BUF * s->channels);
-    if (rc_eligible && avctx->frame_num != s->nmr->rc_frame_num) {
-        if (s->nmr->rc_frame_num > 0 && s->nmr->lam_rc > 0.0f)
-            s->nmr->rc_fill = av_clip(s->nmr->rc_fill + rc_rate_frame - s->last_frame_pb_count,
-                                      -rc_bmax, rc_bmax);
-        s->nmr->rc_frame_num = avctx->frame_num;
-
-        /* Transient burst run state: set at run start and held across the run so
-         * coding stays uniform; repaid from the reservoir's steady stretches. */
-        int is_short = sce->ics.window_sequence[0] == EIGHT_SHORT_SEQUENCE;
-        if (is_short) {
-            if (!s->nmr->prev_was_short)        /* run start */
-                s->nmr->run_burst = s->nmr->frames_since_short >= NMR_BURST_GAP
-                                  ? NMR_BURST_GAIN : 1.0f;
-            s->nmr->frames_since_short = 0;
-        } else {
-            s->nmr->run_burst = 1.0f;
-            s->nmr->frames_since_short++;
-        }
-        s->nmr->prev_was_short = is_short;
-    }
-    int rc_global = rc_eligible && s->nmr->lam_rc > 0.0f;
-
-    if (s->psy.bitres.alloc >= 0)
-        destbits = s->psy.bitres.alloc *
-                   (lambda / (avctx->global_quality ? avctx->global_quality : 120));
-    if (rc_global && s->psy.bitres.alloc >= 0)
-        /* uniform CBR target: nominal rate plus fast reservoir repayment */
-        destbits = (avctx->bit_rate * 1024.0 / avctx->sample_rate
-                    + s->nmr->rc_fill / 2.0) / s->channels;
-    destbits = FFMIN(destbits, 5800);
-    /* honest budget: subtract the measured non-trellis overhead (section data, ICS,
-     * sf/PNS signalling), which is rate-dependent hence adaptive. */
-    if (s->nmr->side_inited)
-        destbits = av_clip(destbits - (int)(s->nmr->side_ema / s->channels), 64, 5800);
-
-    /* Apply the held transient burst factor (set in the run-state machine above). */
-    if (sce->ics.window_sequence[0] == EIGHT_SHORT_SEQUENCE && s->nmr->run_burst > 1.0f)
-        destbits = av_clip((int)(destbits * s->nmr->run_burst), 64, 6800);
-
-    /* band cutoff index for this frame's window size; the bandwidth is fixed
-     * at init and shared with the psy model */
-    cutoff = s->bandwidth * 2 * (1024 / sce->ics.num_windows) / avctx->sample_rate;
-
-    /* Short-block transient noise shaping (pairs with short-block TNS): temporal
-     * premasking clamps each window's threshold toward the preceding windows'
-     * (Apple's preEchoReduction), and flat-residual flattens each window's thresholds
-     * to their per-window mean so TNS synthesis has a white floor to concentrate. */
-    if (sce->ics.window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
-        const float pm_p1 = 0.1f, pm_p2 = 2.0f, pm_p3 = 4.0f;
-        for (int g = 0; g < sce->ics.num_swb; g++) {
-            float t1 = FLT_MAX, t2 = FLT_MAX;   /* original thr of w-1, w-2 */
-            for (int w = 0; w < sce->ics.num_windows; w++) {
-                FFPsyBand *b = &s->psy.ch[s->cur_channel].psy_bands[w*16+g];
-                float t = b->threshold;
-                float c = FFMIN(t, FFMIN(t1*pm_p2, t2*pm_p3));
-                b->threshold = FFMAX(c, t*pm_p1);
-                t2 = t1; t1 = t;
-            }
-        }
-        {
-            for (int w = 0; w < sce->ics.num_windows; w++) {
-                float sum = 0.0f; int n = 0;
-                for (int g = 0; g < sce->ics.num_swb; g++) {
-                    FFPsyBand *b = &s->psy.ch[s->cur_channel].psy_bands[w*16+g];
-                    if (b->energy > b->threshold && b->threshold > 0.0f) { sum += b->threshold; n++; }
-                }
-                if (n > 0) {
-                    float mean = sum / n;
-                    for (int g = 0; g < sce->ics.num_swb; g++) {
-                        FFPsyBand *b = &s->psy.ch[s->cur_channel].psy_bands[w*16+g];
-                        if (b->energy > b->threshold && b->threshold > 0.0f)
-                            b->threshold = mean;
-                    }
-                }
-            }
-        }
-    }
-
-    /* Allocation curve to favour high frequencies */
-    const float a_ae = 0.443f, a_at = 0.111f;
-    for (int w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
-        int start = 0;
-        for (int g = 0; g < sce->ics.num_swb; start += sce->ics.swb_sizes[g++]) {
-            float uplim = 0.0f, ener = 0.0f, spread = 2.0f;
-            int nz = 0;
-            if (sce->band_type[w*16+g] == INTENSITY_BT ||
-                sce->band_type[w*16+g] == INTENSITY_BT2) {
-                /* pre-decided intensity band (right channel): keep its
-                 * signalling, it is not trellis-coded */
-                for (int w2 = 0; w2 < sce->ics.group_len[w]; w2++)
-                    sce->zeroes[(w+w2)*16+g] = 0;
-                continue;
-            }
-            for (int w2 = 0; w2 < sce->ics.group_len[w]; w2++) {
-                FFPsyBand *band = &s->psy.ch[s->cur_channel].psy_bands[(w+w2)*16+g];
-                ener   += band->energy;
-                spread  = FFMIN(spread, band->spread);
-                if (start >= cutoff || band->energy <= band->threshold || band->threshold == 0.0f) {
-                    sce->zeroes[(w+w2)*16+g] = 1;
-                    continue;
-                }
-                uplim += band->threshold;
-                nz = 1;
-            }
-            sce->zeroes[w*16+g] = !nz;
-            thr_real[w*16+g] = uplim;       /* real mask, before the allocation law (PNS gate) */
-            if (nz && ener > 0.0f && uplim > 0.0f)
-                uplim = expf(a_ae * logf(ener) + a_at * logf(uplim));
-            thr[w*16+g]     = uplim;
-            pener[w*16+g]   = ener;
-            pspread[w*16+g] = spread;
-            allz |= nz;
-        }
-    }
-    if (!allz)
-        goto bail;
-
-    s->aacdsp.abs_pow34(s->scoefs, sce->coeffs, 1024);
-    ff_quantize_band_cost_cache_init(s);
-
-    /* finest codeable scalefactor and max value per band */
-    for (int w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
-        int start = w*128;
-        for (int g = 0; g < sce->ics.num_swb; g++) {
-            maxvals[w*16+g] = find_max_val(sce->ics.group_len[w], sce->ics.swb_sizes[g], s->scoefs + start);
-            minsf[w*16+g]   = maxvals[w*16+g] > 0 ? coef2minsf(maxvals[w*16+g]) : 0;
-            start += sce->ics.swb_sizes[g];
-        }
-    }
-
-    /* PASS 1:
-     * precompute each coded band's cost curve at the coarse candidate step
-     * (the lambda search runs on this cheap grid, PASS 2 refines the winner) */
-    {
-        for (int w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
-            int start = w*128;
-            for (int g = 0; g < sce->ics.num_swb; g++) {
-                if (!sce->zeroes[w*16+g] && maxvals[w*16+g] > 0 && nbnd < 128) {
-                    int lo = av_clip(minsf[w*16+g], 0, SCALE_MAX_POS);
-                    float invthr = 1.0f / FFMAX(thr[w*16+g], 1e-9f);
-                    int ncand = nmr_band_curve(s, sce, w, g, start, lo, cstep, NMR_NCAND,
-                                               invthr, maxvals[w*16+g], nd[nbnd], nb[nbnd]);
-                    if (ncand == 0) {
-                        /* nothing codeable -> drop the whole group band. The
-                         * subwindow flags must be cleared too: the encoder later
-                         * re-derives the group flag by ANDing them, which would
-                         * resurrect the band with a never-assigned scalefactor. */
-                        for (int w2 = 0; w2 < sce->ics.group_len[w]; w2++)
-                            sce->zeroes[(w+w2)*16+g] = 1;
-                    } else {
-                        bidx[nbnd] = w*16+g;
-                        bw[nbnd] = w;
-                        bg[nbnd] = g;
-                        bst[nbnd] = start;
-                        blo[nbnd] = lo;
-                        bnc[nbnd] = ncand;
-                        nbnd++;
-                    }
-                }
-                start += sce->ics.swb_sizes[g];
-            }
-        }
-    }
-    if (!nbnd)
-        goto bail;
-
-    /* solve the trellis over all coded bands, then offer PNS at the operating
-     * lambda and re-solve over the survivors with the freed budget */
-    {
-        int nact = nbnd, pns_count = 0;
-        float lam0 = s->nmr->lam[s->cur_channel];
-        float lam;
-
-        for (int b = 0; b < nbnd; b++) {
-            act[b] = b;
-            is_pns[b] = 0;
-        }
-        if (rc_global) {
-            /* bisect to this frame's bit demand within the corridor around the
-             * servoed lambda: per-frame psy demand is tracked, but lambda cannot
-             * jump, which keeps quality smooth across frames */
-            float lo = s->nmr->lam_rc / NMR_RC_CORR;
-            /* Transient burst: widen the lower lambda bound so the bisection can actually
-             * pour the boosted destbits into an onset frame (finer coding kills the
-             * pre-echo); reservoir servo repays it from the steady frames. run_burst==1 on
-             * non-onset frames leaves the corridor unchanged. */
-            if (sce->ics.window_sequence[0] == EIGHT_SHORT_SEQUENCE && s->nmr->run_burst > 1.0f)
-                lo /= s->nmr->run_burst;
-            lam = nmr_solve(s, nd, nb, blo, bnc, cstep, act, nact, destbits, chosen,
-                            lo, s->nmr->lam_rc * NMR_RC_CORR,
-                            NMR_RC_CITERS);
-
-            int tot = 0;
-            for (int k = 0; k < nact; k++)
-                tot += nb[act[k]][chosen[act[k]]];
-            for (int k = 1; k < nact; k++)
-                tot += NMR_SFBITS((blo[act[k]]+chosen[act[k]]*cstep) - (blo[act[k-1]]+chosen[act[k-1]]*cstep));
-            int hardcap = av_clip((int)(5800.f * FFMIN(1.f, lambda / 120.f)), 256, 5800);
-            /* leaky-bucket window: don't borrow past -rc_bmax (cap) or bank past +rc_bmax (floor) */
-            int rc_cap   = FFMIN(hardcap, (s->nmr->rc_fill + rc_rate_frame + rc_bmax) / s->channels);
-            int rc_floor = FFMAX(0, (s->nmr->rc_fill + rc_rate_frame - rc_bmax) / s->channels);
-            if (tot > rc_cap)
-                lam = nmr_solve(s, nd, nb, blo, bnc, cstep, act, nact, rc_cap, chosen,
-                                lam, 1e4f, NMR_CITERS);
-            else if (tot < rc_floor)
-                lam = nmr_solve(s, nd, nb, blo, bnc, cstep, act, nact, rc_floor, chosen,
-                                1e-9f, lam, NMR_CITERS);
-        } else if (NMR_COARSE > 0 && lam0 > 0.0f) {
-            /* per-frame bisection; lambda is strongly frame-correlated, so when a
-             * previous frame's operating lambda exists, bisect a narrow bracket
-             * around it. A result near the bracket edge means the budget crossing
-             * lies outside (hard content transition) == redo the full search. */
-            lam = nmr_solve(s, nd, nb, blo, bnc, cstep, act, nact, destbits, chosen,
-                            lam0/32.0f, lam0*32.0f, NMR_CWARM);
-            if (lam < lam0/16.0f || lam > lam0*16.0f)
-                lam0 = 0.0f;
-        }
-        if (!rc_global && lam0 <= 0.0f)
-            lam = nmr_solve(s, nd, nb, blo, bnc, cstep, act, nact, destbits, chosen,
-                            1e-9f, 1e4f, NMR_COARSE > 0 ? NMR_CITERS : NMR_ITERS);
-
-        /* PASS 2:
-         * refine each band at full granularity (NMR_STEP) in a +/-cstep window
-         * around the coarse pick, then re-solve. Recovers single-pass quality while the
-         * lambda search stayed cheap on the coarse grid. */
-        if (NMR_COARSE > 0) {
-            /* nmr_speed, 0 = slowest/best, higher = faster. It narrows the fine
-             * refine +/-window (scalefactors) below NMR_COARSE: at speed 0 the window
-             * spans the whole coarse-grid gap, so the two-pass result matches the
-             * exhaustive single-pass search.
-             * Each speed level shaves one sf off the window.
-             * At @64k mono (Zim / xRT): speed 0 -> 0.00095/15x,
-             * 2 -> 0.00096/18x, 3 -> 0.00100/20x, 4 -> 0.00103/22x */
-            int win = NMR_COARSE - av_clip(s->options.nmr_speed, 0, 4);
-            for (int b = 0; b < nbnd; b++) {
-                int center = blo[b] + chosen[b]*cstep;
-                int flo    = av_clip(center - win, av_clip(minsf[bidx[b]], 0, SCALE_MAX_POS), SCALE_MAX_POS);
-                int maxn   = FFMIN(NMR_NCAND, 2*win/NMR_STEP + 1);
-                float invthr = 1.0f / FFMAX(thr[bidx[b]], 1e-9f);
-                int ncand  = nmr_band_curve(s, sce, bw[b], bg[b], bst[b], flo, NMR_STEP, maxn,
-                                            invthr, maxvals[bidx[b]], nd[b], nb[b]);
-                blo[b] = flo;
-                bnc[b] = FFMAX(1, ncand);
-            }
-            /* fine pass: narrow corridor around the coarse solve */
-            if (rc_global)
-                lam = nmr_solve(s, nd, nb, blo, bnc, NMR_STEP, act, nact, destbits, chosen,
-                                lam/2.0f, lam*2.0f, NMR_RC_FITERS);
-            else
-                lam = nmr_solve(s, nd, nb, blo, bnc, NMR_STEP, act, nact, destbits, chosen,
-                                lam/16.0f, lam*16.0f, NMR_IFINE);
-        }
-
-        if (rc_global) {
-            /* leaky-bucket clamp: keep the frame within [rc_floor, rc_cap] so the reservoir
-             * stays in +-rc_bmax -- clamp lambda UP if it would borrow past the cap, DOWN if it
-             * would bank past the floor (spend-floor). The hard cap follows the encoder's outer
-             * lambda so the (rare) hard-overflow re-encode -- which shrinks that lambda -- always
-             * converges; on the first pass lambda is nominal and this is 5800. */
-            int hardcap = av_clip((int)(5800.f * FFMIN(1.f, lambda / 120.f)), 256, 5800);
-            int tot = 0;
-            for (int k = 0; k < nact; k++)
-                tot += nb[act[k]][chosen[act[k]]];
-            for (int k = 1; k < nact; k++)
-                tot += NMR_SFBITS((blo[act[k]]+chosen[act[k]]*NMR_STEP) - (blo[act[k-1]]+chosen[act[k-1]]*NMR_STEP));
-            int rc_cap   = FFMIN(hardcap, (s->nmr->rc_fill + rc_rate_frame + rc_bmax) / s->channels);
-            int rc_floor = FFMAX(0, (s->nmr->rc_fill + rc_rate_frame - rc_bmax) / s->channels);
-            if (tot > rc_cap)
-                lam = nmr_solve(s, nd, nb, blo, bnc, NMR_STEP, act, nact, rc_cap, chosen,
-                                lam, 1e4f, NMR_RC_ITERS);
-            else if (tot < rc_floor)
-                lam = nmr_solve(s, nd, nb, blo, bnc, NMR_STEP, act, nact, rc_floor, chosen,
-                                1e-9f, lam, NMR_RC_ITERS);
-        }
-
-        s->nmr->lam[s->cur_channel] = lam;   /* warm start for the next frame */
-        if (rc_global) {
-            /* drag the corridor centre toward the realized lambda so it follows
-             * content drift faster than the reservoir term alone */
-            float c = s->nmr->lam_rc * powf(lam / s->nmr->lam_rc, NMR_RC_TRACK);
-            /* then servo the centre off the reservoir error so the long-run rate
-             * returns to nominal. rc_fill>0 = bits banked (undershooting) -> lower
-             * lambda to spend them; <0 -> raise it. This is what holds the mean;
-             * the corridor tracking alone has no rate authority and a bad centre
-             * would otherwise drift for dozens of frames, starving each one. */
-            float R = avctx->bit_rate * 1024.0 / avctx->sample_rate;
-            c *= exp2f(-NMR_RC_K_CBR * s->nmr->rc_fill / R);
-            s->nmr->lam_rc = av_clipf(c, 1e-6f, 1e4f);
-        } else if (rc_eligible && nbnd >= 8) {
-            /* bootstrap the servo off the first substantive frame; near-silent
-             * lead-in frames have degenerate budgets that rail the bisection to
-             * a nonsense lambda and would poison the whole stream */
-            s->nmr->lam_rc = av_clipf(lam, 1e-4f, 10.0f);
-        }
-
-        {   /* PNS */
-            const float pns_lam = NMR_PNS_LAM;
-            /* band 0 (lowest freq) is kept as the global-gain / sf-chain anchor */
-            for (int b = 1; b < nbnd; b++) {
-                int bi = bidx[b];
-                float spread = pspread[bi];
-                float nmr_pns, cost_keep, cost_pns, frac;
-                if (!sce->can_pns[bi])
-                    continue;
-
-                /* Loud-band guard: never substitute a band whose energy is far above the
-                 * masking threshold -- energy-matched noise on a dominant band clips/pops
-                 * (and is audibly wrong). PNS is for near-masked noise only. */
-                if (pener[bi] > NMR_PNS_MAX_ET * thr_real[bi])
-                    continue;
-
-                /* Struggle gate: no PNS at all unless the encoder is genuinely under bit
-                 * pressure (high operating lambda). */
-                if (lam <= pns_lam)
-                    continue;
-
-                /* Spectral-hole fill: a noise-like band the trellis left mostly empty */
-                frac = nd[b][chosen[b]] * thr[bi] / FFMAX(pener[bi], 1e-9f);
-                if (spread > NMR_PNS_HOLE_SPREAD && frac > NMR_PNS_HOLE_FRAC) {
-                    is_pns[b] = 1;
-                    pns_count++;
-                    continue;
-                }
-
-                /* Only replace a band that is being coded audibly badly */
-                if (nd[b][chosen[b]] * thr[bi] <= NMR_PNS_NDGATE * thr_real[bi])
-                    continue;
-
-                /* perceptual cost of replacing the band with energy-matched noise:
-                 * the non-noise-like fraction of its energy, in dist/threshold units */
-                nmr_pns = FFMAX(0.0f, pener[bi] * (1.0f - spread*spread))
-                          / FFMAX(thr[bi], 1e-9f);
-                cost_keep = nd[b][chosen[b]] + lam * nb[b][chosen[b]];
-                cost_pns  = nmr_pns + lam * NMR_PNS_BITS;
-                if (cost_pns < cost_keep) {
-                    is_pns[b] = 1;
-                    pns_count++;
-                }
-            }
-            if (pns_count) {
-                int budget2 = destbits - pns_count * NMR_PNS_BITS;
-                nact = 0;
-                for (int b = 0; b < nbnd; b++)
-                    if (!is_pns[b])
-                        act[nact++] = b;
-                /* re-solve over the survivors: at fixed lambda the allocation is
-                 * the same except for the repaired sf-delta chain; in bisection
-                 * mode re-spend the freed budget */
-                if (rc_global)
-                    nmr_solve(s, nd, nb, blo, bnc, NMR_STEP, act, nact, budget2, chosen,
-                              lam, lam, 1);
-                else
-                    nmr_solve(s, nd, nb, blo, bnc, NMR_STEP, act, nact, budget2, chosen,
-                              1e-9f, 1e4f, NMR_ITERS);
-            }
-        }
-        for (int b = 0; b < nbnd; b++) {
-            int bi = bidx[b];
-            if (is_pns[b]) {
-                sce->band_type[bi] = NOISE_BT;
-                sce->zeroes[bi]    = 0;
-                sce->pns_ener[bi]  = pener[bi] * FFMIN(1.0f, pspread[bi]*pspread[bi]);
-            } else {
-                sce->sf_idx[bi] = av_clip(blo[b] + chosen[b]*NMR_STEP, 0, SCALE_MAX_POS);
-            }
-        }
-
-        {   /* record the bits this solve accounted for; the encoder compares them
-             * against the channel's real output to keep the budget honest */
-            int tot = 0, prevb = -1;
-            for (int b = 0; b < nbnd; b++) {
-                if (is_pns[b])
-                    continue;
-                tot += nb[b][chosen[b]];
-                if (prevb >= 0)
-                    tot += NMR_SFBITS((blo[b]+chosen[b]*NMR_STEP) - (blo[prevb]+chosen[prevb]*NMR_STEP));
-                prevb = b;
-            }
-            s->nmr->counted[s->cur_channel] = tot;
-        }
-    }
-
-    /* SCALE_MAX_DIFF condition:
-     * re-clamp, codebook fixup, drop uncodeable, set global gain
-     * NOISE_BT bands keep their own scalefactor chain via set_special_band_scalefactors) */
-    {
-        uint8_t nextband[128];
-        int prev = -1;
-        ff_init_nextband_map(sce, nextband);
-        for (int w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
-            for (int g = 0; g < sce->ics.num_swb; g++) {
-                if (sce->band_type[w*16+g] == NOISE_BT ||
-                    sce->band_type[w*16+g] == INTENSITY_BT ||
-                    sce->band_type[w*16+g] == INTENSITY_BT2)
-                    continue;
-                if (sce->zeroes[w*16+g]) {
-                    sce->band_type[w*16+g] = 0;
-                    continue;
-                }
-
-                if (prev != -1)
-                    sce->sf_idx[w*16+g] = av_clip(sce->sf_idx[w*16+g], prev - SCALE_MAX_DIFF, prev + SCALE_MAX_DIFF);
-                sce->band_type[w*16+g] = find_min_book(maxvals[w*16+g], sce->sf_idx[w*16+g]);
-                if (sce->band_type[w*16+g] <= 0) {
-                    if (!ff_sfdelta_can_remove_band(sce, nextband, prev, w*16+g)) {
-                        sce->band_type[w*16+g] = 1;
-                    } else {
-                        /* drop subwindow flags too, see the PASS 1 drop above */
-                        for (int w2 = 0; w2 < sce->ics.group_len[w]; w2++)
-                            sce->zeroes[(w+w2)*16+g] = 1;
-                        sce->band_type[w*16+g] = 0;
-                        continue;
-                    }
-                }
-                if (prev == -1)
-                    sce->sf_idx[0] = sce->sf_idx[w*16+g];   /* global gain */
-                prev = sce->sf_idx[w*16+g];
-            }
-        }
-
-        /* Every band, coded or not, must carry a chain-legal scalefactor: the
-         * codebook trellis (encode_window_bands_info) may later absorb a dropped
-         * band into a nonzero section, resurrecting it, and its sf then gets
-         * coded. Forward-fill with the previous coded sf (delta 0, cheapest);
-         * leading bands get the global gain. */
-        if (prev != -1) {
-            int last = sce->sf_idx[0];
-            for (int w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
-                for (int g = 0; g < sce->ics.num_swb; g++) {
-                    if (!sce->zeroes[w*16+g] && sce->band_type[w*16+g] != NOISE_BT &&
-                        sce->band_type[w*16+g] < RESERVED_BT)
-                        last = sce->sf_idx[w*16+g];
-                    else if (sce->band_type[w*16+g] < RESERVED_BT && (w*16+g) > 0)
-                        sce->sf_idx[w*16+g] = last;
-                }
-            }
-        }
-    }
-    return;
-
-bail:
-    /* Nothing codeable in this channel. Leave a fully consistent state: any
-     * stale nonzero band_type acts as a codebook lower bound in the encoder's
-     * section trellis (encode_window_bands_info), which would forbid the zero
-     * section and resurrect the band with a stale, chain-illegal scalefactor.
-     * Pre-decided intensity bands keep their signalling. */
-    for (int i = 0; i < 128; i++) {
-        if (sce->band_type[i] == INTENSITY_BT || sce->band_type[i] == INTENSITY_BT2)
-            continue;
-        sce->zeroes[i]    = 1;
-        sce->band_type[i] = 0;
-    }
-}
-
-#endif /* AVCODEC_AACCODER_NMR_H */

libavcodec/aaccoder_twoloop.h

@@ -71,6 +71,7 @@ static void search_for_quantizers_twoloop(AVCodecContext *avctx,
     int destbits = avctx->bit_rate * 1024.0 / avctx->sample_rate
         / ((avctx->flags & AV_CODEC_FLAG_QSCALE) ? 2.0f : avctx->ch_layout.nb_channels)
         * (lambda / 120.f);
+    int refbits = destbits;
     int toomanybits, toofewbits;
     char nzs[128];
     uint8_t nextband[128];
@@ -171,8 +172,32 @@ static void search_for_quantizers_twoloop(AVCodecContext *avctx,
     /** and zero out above cutoff frequency */
     {
         int wlen = 1024 / sce->ics.num_windows;
-        /* the bandwidth is fixed at init and shared with the psy model */
-        cutoff = s->bandwidth * 2 * wlen / avctx->sample_rate;
+        int bandwidth;
+
+        /**
+         * Scale, psy gives us constant quality, this LP only scales
+         * bitrate by lambda, so we save bits on subjectively unimportant HF
+         * rather than increase quantization noise. Adjust nominal bitrate
+         * to effective bitrate according to encoding parameters,
+         * AAC_CUTOFF_FROM_BITRATE is calibrated for effective bitrate.
+         */
+        float rate_bandwidth_multiplier = 1.5f;
+        int frame_bit_rate = (avctx->flags & AV_CODEC_FLAG_QSCALE)
+            ? (refbits * rate_bandwidth_multiplier * avctx->sample_rate / 1024)
+            : (avctx->bit_rate / avctx->ch_layout.nb_channels);
+
+        /** Compensate for extensions that increase efficiency */
+        if (s->options.pns || s->options.intensity_stereo)
+            frame_bit_rate *= 1.15f;
+
+        if (avctx->cutoff > 0) {
+            bandwidth = avctx->cutoff;
+        } else {
+            bandwidth = FFMAX(3000, AAC_CUTOFF_FROM_BITRATE(frame_bit_rate, 1, avctx->sample_rate));
+            s->psy.cutoff = bandwidth;
+        }
+
+        cutoff = bandwidth * 2 * wlen / avctx->sample_rate;
         pns_start_pos = NOISE_LOW_LIMIT * 2 * wlen / avctx->sample_rate;
     }
 

libavcodec/aacenc.c

@@ -123,40 +123,40 @@ static const AACPCEInfo aac_pce_configs[] = {
     {
         .layout = AV_CHANNEL_LAYOUT_SURROUND,
         .num_ele = { 2, 0, 0, 0 },
-        .pairing = { { 0, 1 }, },
+        .pairing = { { 1, 0 }, },
         .index = { { 0, 0 }, },
-        .config_map = { 2, TYPE_SCE, TYPE_CPE },
-        .reorder_map = { 2, 0, 1 },
+        .config_map = { 2, TYPE_CPE, TYPE_SCE, },
+        .reorder_map = { 0, 1, 2 },
     },
     {
         .layout = AV_CHANNEL_LAYOUT_3POINT1,
         .num_ele = { 2, 0, 0, 1 },
-        .pairing = { { 0, 1 }, },
+        .pairing = { { 1, 0 }, },
         .index = { { 0, 0 }, { 0 }, { 0 }, { 0 }, },
-        .config_map = { 3, TYPE_SCE, TYPE_CPE, TYPE_LFE },
-        .reorder_map = { 2, 0, 1, 3 },
+        .config_map = { 3, TYPE_CPE, TYPE_SCE, TYPE_LFE },
+        .reorder_map = { 0, 1, 2, 3 },
     },
     {
         .layout = AV_CHANNEL_LAYOUT_4POINT0,
         .num_ele = { 2, 0, 1, 0 },
-        .pairing = { { 0, 1 }, { 0 }, { 0 }, },
+        .pairing = { { 1, 0 }, { 0 }, { 0 }, },
         .index = { { 0, 0 }, { 0 }, { 1 } },
-        .config_map = { 3, TYPE_SCE, TYPE_CPE, TYPE_SCE },
-        .reorder_map = { 2, 0, 1, 3 },
+        .config_map = { 3, TYPE_CPE, TYPE_SCE, TYPE_SCE },
+        .reorder_map = {  0, 1, 2, 3 },
     },
     {
         .layout = AV_CHANNEL_LAYOUT_4POINT1,
-        .num_ele = { 2, 0, 1, 1 },
-        .pairing = { { 0, 1 }, { 0 }, { 0 }, },
-        .index = { { 0, 0 }, { 0 }, { 1 }, { 0 } },
-        .config_map = { 4, TYPE_SCE, TYPE_CPE, TYPE_SCE, TYPE_LFE },
-        .reorder_map = { 2, 0, 1, 4, 3 },
+        .num_ele = { 2, 1, 1, 0 },
+        .pairing = { { 1, 0 }, { 0 }, { 0 }, },
+        .index = { { 0, 0 }, { 1 }, { 2 }, { 0 } },
+        .config_map = { 4, TYPE_CPE, TYPE_SCE, TYPE_SCE, TYPE_SCE },
+        .reorder_map = { 0, 1, 2, 3, 4 },
     },
     {
         .layout = AV_CHANNEL_LAYOUT_2_2,
-        .num_ele = { 1, 0, 1, 0 },
-        .pairing = { { 1 }, { 0 }, { 1 }, },
-        .index = { { 0 }, { 0 }, { 1 } },
+        .num_ele = { 1, 1, 0, 0 },
+        .pairing = { { 1 }, { 1 }, },
+        .index = { { 0 }, { 1 }, },
         .config_map = { 2, TYPE_CPE, TYPE_CPE },
         .reorder_map = { 0, 1, 2, 3 },
     },
@@ -170,131 +170,158 @@ static const AACPCEInfo aac_pce_configs[] = {
     },
     {
         .layout = AV_CHANNEL_LAYOUT_5POINT0,
-        .num_ele = { 2, 0, 1, 0 },
-        .pairing = { { 0, 1 }, { 0 }, { 1 } },
-        .index = { { 0, 0 }, { 0 }, { 1 } },
-        .config_map = { 3, TYPE_SCE, TYPE_CPE, TYPE_CPE },
-        .reorder_map = { 2, 0, 1, 3, 4 },
+        .num_ele = { 2, 1, 0, 0 },
+        .pairing = { { 1, 0 }, { 1 }, },
+        .index = { { 0, 0 }, { 1 } },
+        .config_map = { 3, TYPE_CPE, TYPE_SCE, TYPE_CPE },
+        .reorder_map = { 0, 1, 2, 3, 4 },
     },
     {
         .layout = AV_CHANNEL_LAYOUT_5POINT1,
-        .num_ele = { 2, 0, 1, 1 },
-        .pairing = { { 0, 1 }, { 0 }, { 1 }, },
-        .index = { { 0, 0 }, { 0 }, { 1 }, { 0 } },
-        .config_map = { 4, TYPE_SCE, TYPE_CPE, TYPE_CPE, TYPE_LFE },
-        .reorder_map = { 2, 0, 1, 4, 5, 3 },
+        .num_ele = { 2, 1, 1, 0 },
+        .pairing = { { 1, 0 }, { 0 }, { 1 }, },
+        .index = { { 0, 0 }, { 1 }, { 1 } },
+        .config_map = { 4, TYPE_CPE, TYPE_SCE, TYPE_SCE, TYPE_CPE },
+        .reorder_map = { 0, 1, 2, 3, 4, 5 },
     },
     {
         .layout = AV_CHANNEL_LAYOUT_5POINT0_BACK,
         .num_ele = { 2, 0, 1, 0 },
-        .pairing = { { 0, 1 }, { 0 }, { 1 } },
+        .pairing = { { 1, 0 }, { 0 }, { 1 } },
         .index = { { 0, 0 }, { 0 }, { 1 } },
-        .config_map = { 3, TYPE_SCE, TYPE_CPE, TYPE_CPE },
-        .reorder_map = { 2, 0, 1, 3, 4 },
+        .config_map = { 3, TYPE_CPE, TYPE_SCE, TYPE_CPE },
+        .reorder_map = { 0, 1, 2, 3, 4 },
     },
     {
         .layout = AV_CHANNEL_LAYOUT_5POINT1_BACK,
-        .num_ele = { 2, 0, 1, 1 },
-        .pairing = { { 0, 1 }, { 0 }, { 1 }, },
-        .index = { { 0, 0 }, { 0 }, { 1 }, { 0 } },
-        .config_map = { 4, TYPE_SCE, TYPE_CPE, TYPE_CPE, TYPE_LFE },
-        .reorder_map = { 2, 0, 1, 4, 5, 3 },
+        .num_ele = { 2, 1, 1, 0 },
+        .pairing = { { 1, 0 }, { 0 }, { 1 }, },
+        .index = { { 0, 0 }, { 1 }, { 1 } },
+        .config_map = { 4, TYPE_CPE, TYPE_SCE, TYPE_SCE, TYPE_CPE },
+        .reorder_map = { 0, 1, 2, 3, 4, 5 },
     },
     {
         .layout = AV_CHANNEL_LAYOUT_6POINT0,
-        .num_ele = { 2, 0, 2, 0 },
-        .pairing = { { 0, 1 }, { 0 }, { 1, 0 } },
-        .index = { { 0, 0 }, { 0 }, { 1, 1 } },
-        .config_map = { 4, TYPE_SCE, TYPE_CPE, TYPE_CPE, TYPE_SCE },
-        .reorder_map = { 2, 0, 1, 4, 5, 3 },
+        .num_ele = { 2, 1, 1, 0 },
+        .pairing = { { 1, 0 }, { 1 }, { 0 }, },
+        .index = { { 0, 0 }, { 1 }, { 1 } },
+        .config_map = { 4, TYPE_CPE, TYPE_SCE, TYPE_CPE, TYPE_SCE },
+        .reorder_map = { 0, 1, 2, 3, 4, 5 },
     },
     {
         .layout = AV_CHANNEL_LAYOUT_6POINT0_FRONT,
-        .num_ele = { 2, 0, 1, 0 },
-        .pairing = { { 1, 1 }, { 0 }, { 1 } },
-        .index = { { 0, 1 }, { 0 }, { 2 }, },
+        .num_ele = { 2, 1, 0, 0 },
+        .pairing = { { 1, 1 }, { 1 } },
+        .index = { { 1, 0 }, { 2 }, },
         .config_map = { 3, TYPE_CPE, TYPE_CPE, TYPE_CPE, },
-        .reorder_map = { 2, 3, 0, 1, 4, 5 },
+        .reorder_map = { 0, 1, 2, 3, 4, 5 },
     },
     {
         .layout = AV_CHANNEL_LAYOUT_HEXAGONAL,
         .num_ele = { 2, 0, 2, 0 },
-        .pairing = { { 0, 1 }, { 0 }, { 1, 0 } },
-        .index = { { 0, 0 }, { 0 }, { 1, 1 } },
-        .config_map = { 4, TYPE_SCE, TYPE_CPE, TYPE_CPE, TYPE_SCE },
-        .reorder_map = { 2, 0, 1, 3, 4, 5 },
+        .pairing = { { 1, 0 },{ 0 },{ 1, 0 }, },
+        .index = { { 0, 0 },{ 0 },{ 1, 1 } },
+        .config_map = { 4, TYPE_CPE, TYPE_SCE, TYPE_CPE, TYPE_SCE, },
+        .reorder_map = { 0, 1, 2, 3, 4, 5 },
     },
     {
         .layout = AV_CHANNEL_LAYOUT_6POINT1,
-        .num_ele = { 2, 0, 2, 1 },
-        .pairing = { { 0, 1 }, { 0 }, { 1, 0 }, },
-        .index = { { 0, 0 }, { 0 }, { 1, 1 }, { 0 } },
-        .config_map = { 5, TYPE_SCE, TYPE_CPE, TYPE_CPE, TYPE_SCE, TYPE_LFE },
-        .reorder_map = { 2, 0, 1, 5, 6, 4, 3 },
+        .num_ele = { 2, 1, 2, 0 },
+        .pairing = { { 1, 0 },{ 0 },{ 1, 0 }, },
+        .index = { { 0, 0 },{ 1 },{ 1, 2 } },
+        .config_map = { 5, TYPE_CPE, TYPE_SCE, TYPE_SCE, TYPE_CPE, TYPE_SCE },
+        .reorder_map = { 0, 1, 2, 3, 4, 5, 6 },
     },
     {
         .layout = AV_CHANNEL_LAYOUT_6POINT1_BACK,
-        .num_ele = { 2, 0, 2, 1 },
-        .pairing = { { 0, 1 },{ 0 },{ 1, 0 }, },
-        .index = { { 0, 0 },{ 0 },{ 1, 1 },{ 0 } },
-        .config_map = { 5, TYPE_SCE, TYPE_CPE, TYPE_CPE, TYPE_SCE, TYPE_LFE },
-        .reorder_map = { 2, 0, 1, 4, 5, 6, 3 },
+        .num_ele = { 2, 1, 2, 0 },
+        .pairing = { { 1, 0 }, { 0 }, { 1, 0 }, },
+        .index = { { 0, 0 }, { 1 }, { 1, 2 } },
+        .config_map = { 5, TYPE_CPE, TYPE_SCE, TYPE_SCE, TYPE_CPE, TYPE_SCE },
+        .reorder_map = { 0, 1, 2, 3, 4, 5, 6 },
     },
     {
         .layout = AV_CHANNEL_LAYOUT_6POINT1_FRONT,
-        .num_ele = { 2, 0, 1, 1 },
-        .pairing = { { 1, 1 }, { 0 }, { 1 }, },
-        .index = { { 0, 1 }, { 0 }, { 2 }, { 0 }, },
-        .config_map = { 4, TYPE_CPE, TYPE_CPE, TYPE_CPE, TYPE_LFE, },
-        .reorder_map = { 3, 4, 0, 1, 5, 6, 2 },
+        .num_ele = { 2, 1, 2, 0 },
+        .pairing = { { 1, 0 }, { 0 }, { 1, 0 }, },
+        .index = { { 0, 0 }, { 1 }, { 1, 2 } },
+        .config_map = { 5, TYPE_CPE, TYPE_SCE, TYPE_SCE, TYPE_CPE, TYPE_SCE },
+        .reorder_map = { 0, 1, 2, 3, 4, 5, 6 },
     },
     {
         .layout = AV_CHANNEL_LAYOUT_7POINT0,
-        .num_ele = { 2, 0, 2, 0 },
-        .pairing = { { 0, 1 }, { 0 }, { 1, 1 }, },
-        .index = { { 0, 0 }, { 0 }, { 2, 1 }, },
-        .config_map = { 4, TYPE_SCE, TYPE_CPE, TYPE_CPE, TYPE_CPE },
-        .reorder_map = { 2, 0, 1, 3, 4, 5, 6 },
+        .num_ele = { 2, 1, 1, 0 },
+        .pairing = { { 1, 0 }, { 1 }, { 1 }, },
+        .index = { { 0, 0 }, { 1 }, { 2 }, },
+        .config_map = { 4, TYPE_CPE, TYPE_SCE, TYPE_CPE, TYPE_CPE },
+        .reorder_map = { 0, 1, 2, 3, 4, 5, 6 },
     },
     {
         .layout = AV_CHANNEL_LAYOUT_7POINT0_FRONT,
-        .num_ele = { 3, 0, 1, 0 },
-        .pairing = { { 0, 1, 1 }, { 0 }, { 1 }, },
-        .index = { { 0, 0, 1 }, { 0 }, { 2 }, },
-        .config_map = { 4, TYPE_SCE, TYPE_CPE, TYPE_CPE, TYPE_CPE },
-        .reorder_map = { 2, 3, 4, 0, 1, 5, 6 },
+        .num_ele = { 2, 1, 1, 0 },
+        .pairing = { { 1, 0 }, { 1 }, { 1 }, },
+        .index = { { 0, 0 }, { 1 }, { 2 }, },
+        .config_map = { 4, TYPE_CPE, TYPE_SCE, TYPE_CPE, TYPE_CPE },
+        .reorder_map = { 0, 1, 2, 3, 4, 5, 6 },
     },
     {
         .layout = AV_CHANNEL_LAYOUT_7POINT1,
-        .num_ele = { 2, 0, 2, 1 },
-        .pairing = { { 0, 1 }, { 0 }, { 1, 1 }, },
-        .index = { { 0, 0 }, { 0 }, { 2, 1 }, { 0 } },
-        .config_map = { 5, TYPE_SCE, TYPE_CPE, TYPE_CPE, TYPE_CPE, TYPE_LFE },
-        .reorder_map = { 2, 0, 1, 4, 5, 6, 7, 3 },
+        .num_ele = { 2, 1, 2, 0 },
+        .pairing = { { 1, 0 }, { 0 }, { 1, 1 }, },
+        .index = { { 0, 0 }, { 1 }, { 1, 2 }, { 0 } },
+        .config_map = { 5, TYPE_CPE, TYPE_SCE,  TYPE_SCE, TYPE_CPE, TYPE_CPE },
+        .reorder_map = { 0, 1, 2, 3, 4, 5, 6, 7 },
     },
     {
         .layout = AV_CHANNEL_LAYOUT_7POINT1_WIDE,
-        .num_ele = { 3, 0, 1, 1 },
-        .pairing = { { 0, 1, 1 }, { 0 }, { 1 }, },
-        .index = { { 0, 0, 1 }, { 0 }, { 2 }, { 0 }, },
-        .config_map = { 5, TYPE_SCE, TYPE_CPE, TYPE_CPE, TYPE_CPE, TYPE_LFE },
-        .reorder_map = { 2, 4, 5, 0, 1, 6, 7, 3 },
+        .num_ele = { 2, 1, 2, 0 },
+        .pairing = { { 1, 0 }, { 0 },{  1, 1 }, },
+        .index = { { 0, 0 }, { 1 }, { 1, 2 }, { 0 } },
+        .config_map = { 5, TYPE_CPE, TYPE_SCE, TYPE_SCE, TYPE_CPE, TYPE_CPE },
+        .reorder_map = { 0, 1, 2, 3, 4, 5, 6, 7 },
     },
     {
         .layout = AV_CHANNEL_LAYOUT_7POINT1_WIDE_BACK,
-        .num_ele = { 3, 0, 1, 1 },
-        .pairing = { { 0, 1, 1 }, { 0 }, { 1 } },
-        .index = { { 0, 0, 1 }, { 0 }, { 2 }, { 0 } },
-        .config_map = { 5, TYPE_SCE, TYPE_CPE, TYPE_CPE, TYPE_CPE, TYPE_LFE },
-        .reorder_map = { 2, 6, 7, 0, 1, 4, 5, 3 },
+        .num_ele = { 2, 1, 2, 0 },
+        .pairing = { { 1, 0 }, { 0 }, { 1, 1 }, },
+        .index = { { 0, 0 }, { 1 }, { 1, 2 }, { 0 } },
+        .config_map = { 5, TYPE_CPE, TYPE_SCE, TYPE_SCE, TYPE_CPE, TYPE_CPE },
+        .reorder_map = { 0, 1, 2, 3, 4, 5, 6, 7 },
     },
     {
         .layout = AV_CHANNEL_LAYOUT_OCTAGONAL,
-        .num_ele = { 2, 0, 3, 0 },
-        .pairing = { { 0, 1 }, { 0 }, { 1, 1, 0 }, },
-        .index = { { 0, 0 }, { 0 }, { 1, 2, 1 }, },
-        .config_map = { 5, TYPE_SCE, TYPE_CPE, TYPE_CPE, TYPE_CPE, TYPE_SCE },
-        .reorder_map = { 2, 0, 1, 6, 7, 3, 4, 5 },
+        .num_ele = { 2, 1, 2, 0 },
+        .pairing = { { 1, 0 }, { 1 }, { 1, 0 }, },
+        .index = { { 0, 0 }, { 1 }, { 2, 1 } },
+        .config_map = { 5, TYPE_CPE, TYPE_SCE, TYPE_CPE, TYPE_CPE, TYPE_SCE },
+        .reorder_map = { 0, 1, 2, 3, 4, 5, 6, 7 },
+    },
+    {   /* Meant for order 2/mixed ambisonics */
+        .layout = { .order = AV_CHANNEL_ORDER_NATIVE, .nb_channels = 9,
+                    .u.mask = AV_CH_LAYOUT_OCTAGONAL | AV_CH_TOP_CENTER },
+        .num_ele = { 2, 2, 2, 0 },
+        .pairing = { { 1, 0 }, { 1, 0 }, { 1, 0 }, },
+        .index = { { 0, 0 }, { 1, 1 }, { 2, 2 } },
+        .config_map = { 6, TYPE_CPE, TYPE_SCE, TYPE_CPE, TYPE_SCE, TYPE_CPE, TYPE_SCE },
+        .reorder_map = { 0, 1, 2, 3, 4, 5, 6, 7, 8 },
+    },
+    {   /* Meant for order 2/mixed ambisonics */
+        .layout = { .order = AV_CHANNEL_ORDER_NATIVE, .nb_channels = 10,
+                    .u.mask = AV_CH_LAYOUT_6POINT0_FRONT | AV_CH_BACK_CENTER |
+                              AV_CH_BACK_LEFT | AV_CH_BACK_RIGHT | AV_CH_TOP_CENTER },
+        .num_ele = { 2, 2, 2, 0 },
+        .pairing = { { 1, 1 }, { 1, 0 }, { 1, 0 }, },
+        .index = { { 0, 1 }, { 2, 0 }, { 3, 1 } },
+        .config_map = { 6, TYPE_CPE, TYPE_CPE, TYPE_CPE, TYPE_SCE, TYPE_CPE, TYPE_SCE },
+        .reorder_map = { 0, 1, 2, 3, 4, 5, 6, 7, 8, 9 },
+    },
+    {
+        .layout = AV_CHANNEL_LAYOUT_HEXADECAGONAL,
+        .num_ele = { 4, 2, 4, 0 },
+        .pairing = { { 1, 0, 1, 0 }, { 1, 1 }, { 1, 0, 1, 0 }, },
+        .index = { { 0, 0, 1, 1 }, { 2, 3 }, { 4, 2, 5, 3 } },
+        .config_map = { 10, TYPE_CPE, TYPE_SCE, TYPE_CPE, TYPE_SCE, TYPE_CPE, TYPE_CPE, TYPE_CPE, TYPE_SCE, TYPE_CPE, TYPE_SCE },
+        .reorder_map = { 0, 1, 2, 3, 4, 5, 6, 7, 8, 9, 10, 11, 12, 13, 14, 15 },
     },
 };
 
@@ -339,10 +366,11 @@ static void put_pce(PutBitContext *pb, AVCodecContext *avctx)
  * Make AAC audio config object.
  * @see 1.6.2.1 "Syntax - AudioSpecificConfig"
  */
-static int put_audio_specific_config(AVCodecContext *avctx, int chcfg)
+static int put_audio_specific_config(AVCodecContext *avctx)
 {
     PutBitContext pb;
     AACEncContext *s = avctx->priv_data;
+    int channels = (!s->needs_pce)*(s->channels - (s->channels == 8 ? 1 : 0));
     const int max_size = 32;
 
     avctx->extradata = av_mallocz(max_size);
@@ -352,7 +380,7 @@ static int put_audio_specific_config(AVCodecContext *avctx, int chcfg)
     init_put_bits(&pb, avctx->extradata, max_size);
     put_bits(&pb, 5, s->profile+1); //profile
     put_bits(&pb, 4, s->samplerate_index); //sample rate index
-    put_bits(&pb, 4, chcfg);
+    put_bits(&pb, 4, channels);
     //GASpecificConfig
     put_bits(&pb, 1, 0); //frame length - 1024 samples
     put_bits(&pb, 1, 0); //does not depend on core coder
@@ -577,222 +605,6 @@ static void apply_intensity_stereo(ChannelElement *cpe)
     }
 }
 
-/* Intensity stereo is only allowed when its irreducible image error */
-#define NMR_IS_IMG_GATE 0.5f
-
-/* Frequency in Hz for the lower limit of intensity stereo */
-#define NMR_IS_LOW_LIMIT 6100
-
-/* Rate ceiling (bits/sample/channel) above which intensity is skipped, ~145kbps */
-#define NMR_IS_MAXBPS 1.52f
-
-/* The rate ceiling is lifted on hard-to-code frames. The signal is the bit
- * reservoir going into deficit: a negative fill means the trellis is spending
- * more than the nominal rate to hold quality (operating lambda has climbed). */
-#define NMR_IS_FILLGAIN 0.27f
-#define NMR_IS_FILLMAX  0.40f
-
-/* M/S thresholds: a band is recoded as mid+side when the side is negligible */
-#define NMR_MS_EQUIV 0.01f
-#define NMR_MS_MASK  0.0f
-
-/* PNS-stereo decorrelation gate: a band may be noise-substituted in a CPE only if its
- * side energy is at least this fraction of its mid energy, i.e. the image is genuinely
- * wide (channels decorrelated). PNS renders uncorrelated noise per channel, so it only
- * preserves the image on already-wide bands; a much stricter bar than I/S (which can
- * collapse correlated bands). Lower = more PNS / more imaging risk. */
-#define NMR_PNS_STEREO_DECORR 0.6f
-
-/* Recode one band's window group as mid+side in place, updating the psy band
- * energies/thresholds to the M/S spectra. The threshold is halved as a coarse guard
- * against L/R unmasking of the independently-quantized M/S noise (M/S is a lossless
- * rotation but lossy coding). Used for the M/S decision and the intensity fallback. */
-static void nmr_apply_ms_band(AACEncContext *s, ChannelElement *cpe,
-                              int w, int g, int start, int len, int gl)
-{
-    SingleChannelElement *sce0 = &cpe->ch[0];
-    SingleChannelElement *sce1 = &cpe->ch[1];
-    cpe->ms_mask[w*16+g] = 1;
-    for (int w2 = 0; w2 < gl; w2++) {
-        FFPsyBand *b0 = &s->psy.ch[s->cur_channel+0].psy_bands[(w+w2)*16+g];
-        FFPsyBand *b1 = &s->psy.ch[s->cur_channel+1].psy_bands[(w+w2)*16+g];
-        float *L = sce0->coeffs + start + (w+w2)*128;
-        float *R = sce1->coeffs + start + (w+w2)*128;
-        float em = 0.0f, es = 0.0f;
-        for (int i = 0; i < len; i++) {
-            float m = (L[i] + R[i]) * 0.5f;
-            R[i] = m - R[i]; L[i] = m;
-            em += L[i]*L[i]; es += R[i]*R[i];
-        }
-        b0->threshold = b1->threshold = FFMIN(b0->threshold, b1->threshold) * 0.5f;
-        b0->energy = em; b1->energy = es;
-    }
-}
-
-/* Intensity-stereo perceptual test for one band's window group: collapse the pair
- * to a single carrier (L + p*R)*scale that the decoder rescales per channel, and
- * check that the irreducible image error, which no bit budget can reduce, is
- * masked in both channels. On success returns 1 and fills the carrier scale, the
- * decoder's R/carrier ratio sr_, and the phase p. The caller restricts this to HF
- * bands with energy in both channels. */
-static int nmr_is_image_masked(AACEncContext *s, ChannelElement *cpe,
-                               int w, int g, int start, int len, int gl,
-                               float ener0, float ener1, float dot,
-                               float minthr0, float minthr1,
-                               float *scale_out, float *sr_out, int *p_out)
-{
-    int p = dot >= 0.0f ? 1 : -1;
-    float ener01 = ener0 + ener1 + 2*p*dot;     /* energy of L + p*R */
-    if (ener01 <= FLT_MIN)
-        return 0;
-    float scale = sqrtf(ener0 / ener01);        /* carrier = (L + p*R)*scale */
-    float sr_   = sqrtf(ener1 / ener0);         /* decoder: R = p*sr_*carrier */
-    float img0 = 0.0f, img1 = 0.0f;
-    for (int w2 = 0; w2 < gl; w2++) {
-        const float *L = cpe->ch[0].coeffs + start + (w+w2)*128;
-        const float *R = cpe->ch[1].coeffs + start + (w+w2)*128;
-        for (int i = 0; i < len; i++) {
-            float c  = (L[i] + p*R[i]) * scale;
-            float dl = L[i] - c, dr = R[i] - p*sr_*c;
-            img0 += dl*dl; img1 += dr*dr;
-        }
-    }
-    if (img0 >= NMR_IS_IMG_GATE * minthr0 * gl ||
-        img1 >= NMR_IS_IMG_GATE * minthr1 * gl)
-        return 0;
-    *scale_out = scale; *sr_out = sr_; *p_out = p;
-    return 1;
-}
-
-/* Recode one band's window group as intensity stereo in place: replace L with the
- * carrier, zero R, signal the phase via the side channel's band type, and fold the
- * pair's masking into the surviving (carrier) channel. */
-static void nmr_apply_is_band(AACEncContext *s, ChannelElement *cpe,
-                              int w, int g, int start, int len, int gl,
-                              float scale, float sr_, int p,
-                              float ener0, float ener1)
-{
-    cpe->is_mask[w*16+g] = 1;
-    cpe->ch[0].is_ener[w*16+g] = scale;
-    cpe->ch[1].is_ener[w*16+g] = ener0 / ener1;
-    cpe->ch[1].band_type[w*16+g] = p > 0 ? INTENSITY_BT : INTENSITY_BT2;
-    for (int w2 = 0; w2 < gl; w2++) {
-        FFPsyBand *b0 = &s->psy.ch[s->cur_channel+0].psy_bands[(w+w2)*16+g];
-        FFPsyBand *b1 = &s->psy.ch[s->cur_channel+1].psy_bands[(w+w2)*16+g];
-        float *L = cpe->ch[0].coeffs + start + (w+w2)*128;
-        float *R = cpe->ch[1].coeffs + start + (w+w2)*128;
-        float ec = 0.0f;
-        for (int i = 0; i < len; i++) {
-            L[i] = (L[i] + p*R[i]) * scale;
-            R[i] = 0.0f;
-            ec += L[i]*L[i];
-        }
-        b0->threshold = FFMIN(b0->threshold, b1->threshold / FFMAX(sr_*sr_, 1e-9f));
-        b0->energy = ec; b1->energy = 0.0f;
-    }
-}
-
-/*
- * Per-band stereo-mode decision (L/R vs M/S vs intensity) for the NMR coder,
- * made before quantization from the psychoacoustic model alone, so the
- * quantizer search allocates natively on the spectra that are actually coded.
- */
-static void nmr_decide_stereo(AACEncContext *s, ChannelElement *cpe)
-{
-    SingleChannelElement *sce0 = &cpe->ch[0];
-    SingleChannelElement *sce1 = &cpe->ch[1];
-    IndividualChannelStream *ics = &sce0->ics;
-    const AVCodecContext *avctx = s->psy.avctx;
-    const float freq_mult = avctx->sample_rate / (1024.0f / ics->num_windows) / 2.0f;
-    const float bps = avctx->bit_rate > 0 ?
-                      (float)avctx->bit_rate / avctx->sample_rate / avctx->ch_layout.nb_channels : 0.0f;
-    int is_count = 0;
-
-    /* Stereo decision, with no bitrate dependence. Start from full L/R and depart from
-     * it only where the change is inaudible. M/S and I/S differ in what they trade:
-     *   M/S  recodes the pair as mid+side -- an invertible rotation, but the M and S
-     *        are quantized independently, so it is lossy coding whose noise un-mixes
-     *        back to L/R. Used where it barely changes the result (the side is
-     *        negligible vs the mid, so it is ~equivalent to L/R at the same rate) --
-     *        OR where the doubled side energy is masked.
-     *   I/S  drops the side phase and keeps its energy, where the residual image error
-     *        is masked. Used for the decorrelated HF that M/S cannot help.
-     * Both tests are content/perceptual and frame-stable, so the image holds. */
-
-    /* I/S rate gate: eligible at/below ~128 kbps, with the ceiling lifted on hard
-     * frames (bit reservoir in deficit) so a starved high-rate passage can still
-     * call on intensity. Where an I/S candidate is found but IS is not eligible, fall
-     * back to M/S: not free, but ~equivalent to L/R there and it lets the energy
-     * compact into the mid. */
-    const float rate_frame = avctx->bit_rate * 1024.0f / FFMAX(avctx->sample_rate, 1);
-    const float deficit = (s->nmr && rate_frame > 0.0f)
-                          ? FFMAX(0.0f, -(float)s->nmr->rc_fill / rate_frame) : 0.0f;
-    const float is_bonus = FFMIN(NMR_IS_FILLMAX, NMR_IS_FILLGAIN * deficit);
-    const int allow_is = s->options.intensity_stereo && bps < NMR_IS_MAXBPS + is_bonus;
-
-    for (int w = 0; w < ics->num_windows; w += ics->group_len[w]) {
-        int start = 0;
-        for (int g = 0; g < ics->num_swb; start += ics->swb_sizes[g++]) {
-            int len = ics->swb_sizes[g], gl = ics->group_len[w];
-            float ener0 = 0.0f, ener1 = 0.0f, dot = 0.0f, es_tot = 0.0f, em_tot = 0.0f;
-            float minthr0 = FLT_MAX, minthr1 = FLT_MAX;
-
-            cpe->is_mask[w*16+g] = 0;
-            cpe->ms_mask[w*16+g] = 0;
-
-            for (int w2 = 0; w2 < gl; w2++) {
-                FFPsyBand *b0 = &s->psy.ch[s->cur_channel+0].psy_bands[(w+w2)*16+g];
-                FFPsyBand *b1 = &s->psy.ch[s->cur_channel+1].psy_bands[(w+w2)*16+g];
-                const float *L = sce0->coeffs + start + (w+w2)*128;
-                const float *R = sce1->coeffs + start + (w+w2)*128;
-                float el = 0.0f, er = 0.0f, em = 0.0f, es = 0.0f, d = 0.0f;
-                for (int i = 0; i < len; i++) {
-                    float m  = (L[i] + R[i]) * 0.5f;
-                    float sv = m - R[i];
-                    el += L[i]*L[i]; er += R[i]*R[i];
-                    em += m*m; es += sv*sv; d += L[i]*R[i];
-                }
-                ener0 += el; ener1 += er; dot += d; es_tot += es; em_tot += em;
-                minthr0 = FFMIN(minthr0, b0->threshold);
-                minthr1 = FFMIN(minthr1, b1->threshold);
-            }
-            float thr_g = FFMIN(minthr0, minthr1) * gl;   /* group masking budget */
-
-            /* PNS-stereo reservation. Reserve a band for noise substitution only if it
-             * is noise-like in both channels (intersected can_pns) and clearly
-             * decorrelated (wide image). */
-            if (cpe->ch[0].can_pns[w*16+g] && cpe->ch[1].can_pns[w*16+g] &&
-                es_tot > NMR_PNS_STEREO_DECORR * em_tot)
-                continue;
-            cpe->ch[0].can_pns[w*16+g] = cpe->ch[1].can_pns[w*16+g] = 0;
-
-            int ms_ok = s->options.mid_side &&
-                        (s->options.mid_side == 1 ||
-                         es_tot < NMR_MS_EQUIV * em_tot ||
-                         es_tot < NMR_MS_MASK  * thr_g);
-            float scale, sr_; int p;
-            int is_ok = !ms_ok &&
-                        start * freq_mult > NMR_IS_LOW_LIMIT &&
-                        ener0 > FLT_MIN && ener1 > FLT_MIN &&
-                        nmr_is_image_masked(s, cpe, w, g, start, len, gl,
-                                            ener0, ener1, dot, minthr0, minthr1,
-                                            &scale, &sr_, &p);
-
-            if (ms_ok) {
-                nmr_apply_ms_band(s, cpe, w, g, start, len, gl);
-            } else if (is_ok && allow_is) {
-                nmr_apply_is_band(s, cpe, w, g, start, len, gl,
-                                  scale, sr_, p, ener0, ener1);
-                is_count++;
-            } else if (is_ok && s->options.mid_side) {
-                nmr_apply_ms_band(s, cpe, w, g, start, len, gl);
-            }
-            /* else: keep full L/R stereo */
-        }
-    }
-    cpe->is_mode = !!is_count;
-}
-
 static void apply_mid_side_stereo(ChannelElement *cpe)
 {
     int w, w2, g, i;
@@ -1166,6 +978,12 @@ static int aac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
                 s->psy.bitres.alloc /= chans;
             }
             s->cur_type = tag;
+            for (ch = 0; ch < chans; ch++) {
+                s->cur_channel = start_ch + ch;
+                if (s->options.pns && s->coder->mark_pns)
+                    s->coder->mark_pns(s, avctx, &cpe->ch[ch]);
+                s->coder->search_for_quantizers(avctx, s, &cpe->ch[ch], s->lambda);
+            }
             if (chans > 1
                 && wi[0].window_type[0] == wi[1].window_type[0]
                 && wi[0].window_shape   == wi[1].window_shape) {
@@ -1178,75 +996,26 @@ static int aac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
                     }
                 }
             }
-
-            const int use_tns = s->options.tns && s->coder->search_for_tns &&
-                                s->coder->apply_tns_filt;
-
-            /* The NMR coder rate-controls itself and never re-quantizes, so TNS must run
-             * before the quantizer */
-            const int tns_first = s->options.coder == AAC_CODER_NMR;
-            if (tns_first && use_tns) {
-                for (ch = 0; ch < chans; ch++) {
-                    sce = &cpe->ch[ch];
-                    s->cur_channel = start_ch + ch;
-                    /* mono: mark_pns before TNS so the region cap sees PNS bands. Stereo
-                     * PNS is marked in its own block (below) after the stereo decision. */
-                    if (chans == 1 && s->options.pns && s->coder->mark_pns)
-                        s->coder->mark_pns(s, avctx, sce);
-                    s->coder->search_for_tns(s, sce);
-                    s->coder->apply_tns_filt(s, sce);
-                    if (sce->tns.present)
-                        tns_mode = 1;
-                }
-            }
-
-            /* NMR stereo PNS (imaging-safe). Mark each channel's noise-like bands on the
-             * original L/R psy, then keep PNS only where BOTH channels are noise-like. */
-            if (chans == 2 && cpe->common_window && tns_first &&
-                s->options.pns && s->coder->mark_pns) {
-                s->cur_channel = start_ch;     s->coder->mark_pns(s, avctx, &cpe->ch[0]);
-                s->cur_channel = start_ch + 1; s->coder->mark_pns(s, avctx, &cpe->ch[1]);
-                for (int b = 0; b < 128; b++)
-                    if (!cpe->ch[0].can_pns[b] || !cpe->ch[1].can_pns[b])
-                        cpe->ch[0].can_pns[b] = cpe->ch[1].can_pns[b] = 0;
-            }
-
-            /* The NMR coder decides I/S and M/S BEFORE quantization, from the psy model,
-             * and the trellis then allocates natively on the coeffs actually coded. */
-            if (chans == 2 && cpe->common_window && s->options.coder == AAC_CODER_NMR &&
-                (s->options.mid_side || s->options.intensity_stereo)) {
-                s->cur_channel = start_ch;
-                nmr_decide_stereo(s, cpe);
-            }
-            for (ch = 0; ch < chans; ch++) {
-                s->cur_channel = start_ch + ch;
-                /* NMR PNS is mono-only */
-                if (s->options.pns && s->coder->mark_pns && !tns_first)
-                    s->coder->mark_pns(s, avctx, &cpe->ch[ch]);
-                s->coder->search_for_quantizers(avctx, s, &cpe->ch[ch], s->lambda);
-            }
-            for (ch = 0; ch < chans; ch++) { /* TNS (non-NMR) and PNS */
+            for (ch = 0; ch < chans; ch++) { /* TNS and PNS */
                 sce = &cpe->ch[ch];
                 s->cur_channel = start_ch + ch;
-                if (!tns_first && use_tns) {
+                if (s->options.tns && s->coder->search_for_tns)
                     s->coder->search_for_tns(s, sce);
+                if (s->options.tns && s->coder->apply_tns_filt)
                     s->coder->apply_tns_filt(s, sce);
-                    if (sce->tns.present)
-                        tns_mode = 1;
-                }
+                if (sce->tns.present)
+                    tns_mode = 1;
                 if (s->options.pns && s->coder->search_for_pns)
                     s->coder->search_for_pns(s, avctx, sce);
             }
             s->cur_channel = start_ch;
             if (s->options.intensity_stereo) { /* Intensity Stereo */
-                if (s->options.coder != AAC_CODER_NMR) { /* NMR: decided pre-search */
-                    if (s->coder->search_for_is)
-                        s->coder->search_for_is(s, avctx, cpe);
-                    apply_intensity_stereo(cpe);
-                }
+                if (s->coder->search_for_is)
+                    s->coder->search_for_is(s, avctx, cpe);
                 if (cpe->is_mode) is_mode = 1;
+                apply_intensity_stereo(cpe);
             }
-            if (s->options.mid_side && s->options.coder != AAC_CODER_NMR) { /* Mid/Side stereo */
+            if (s->options.mid_side) { /* Mid/Side stereo */
                 if (s->options.mid_side == -1 && s->coder->search_for_ms)
                     s->coder->search_for_ms(s, cpe);
                 else if (cpe->common_window)
@@ -1274,19 +1043,11 @@ static int aac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
             break;
         }
 
-        frame_bits = put_bits_count(&s->pb);
-
-        /* The NMR coder rate-controls itself (global-lambda reservoir servo):
-         * per-frame bits intentionally float around the nominal rate, so skip
-         * the lambda rate loop and only intervene on a hard overflow. */
-        if (s->options.coder == AAC_CODER_NMR && avctx->bit_rate_tolerance != 0 &&
-            frame_bits < 6144 * s->channels - 3)
-            break;
-
         /* rate control stuff
          * allow between the nominal bitrate, and what psy's bit reservoir says to target
          * but drift towards the nominal bitrate always
          */
+        frame_bits = put_bits_count(&s->pb);
         rate_bits = avctx->bit_rate * 1024 / avctx->sample_rate;
         rate_bits = FFMIN(rate_bits, 6144 * s->channels - 3);
         too_many_bits = FFMAX(target_bits, rate_bits);
@@ -1350,80 +1111,18 @@ static int aac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
         }
     } while (1);
 
-    /* tool-usage stats over the final per-band decisions of this frame */
-    for (i = 0; i < s->chan_map[0]; i++) {
-        int etag = s->chan_map[i + 1], echans = etag == TYPE_CPE ? 2 : 1;
-        ChannelElement *ce = &s->cpe[i];
-        IndividualChannelStream *ics = &ce->ch[0].ics;
-        for (ch = 0; ch < echans; ch++) {   /* per-channel frame stats */
-            int is_short = ce->ch[ch].ics.window_sequence[0] == EIGHT_SHORT_SEQUENCE;
-            s->stat_chans++;
-            if (is_short)
-                s->stat_short++;
-            if (ce->ch[ch].tns.present) {
-                if (is_short) s->stat_tns_short++;
-                else          s->stat_tns_long++;
-            }
-        }
-        for (w = 0; w < ics->num_windows; w += ics->group_len[w]) {
-            for (int g = 0; g < ics->num_swb; g++) {
-                int idx = w*16 + g, coded = 0;
-                for (ch = 0; ch < echans; ch++) {
-                    SingleChannelElement *sce = &ce->ch[ch];
-                    if (sce->zeroes[idx] && sce->band_type[idx] == 0)
-                        continue;
-                    s->stat_ch_bands++;
-                    if (sce->band_type[idx] == NOISE_BT)
-                        s->stat_pns++;
-                    coded = 1;
-                }
-                if (etag == TYPE_CPE && coded) {
-                    s->stat_cpe_bands++;
-                    if (ce->ms_mask[idx]) s->stat_ms++;
-                    if (ce->is_mask[idx]) s->stat_is++;
-                }
-            }
-        }
-    }
-
     put_bits(&s->pb, 3, TYPE_END);
     flush_put_bits(&s->pb);
 
     s->last_frame_pb_count = put_bits_count(&s->pb);
-
-    /* NMR rate accounting: how many bits the frame really took beyond what the
-     * trellis counted; feeds the next frame's budget correction */
-    if (s->nmr) {
-        int counted = 0;
-        for (i = 0; i < s->channels; i++)
-            counted += s->nmr->counted[i];
-        if (counted > 0) {
-            float side = (float)s->last_frame_pb_count - counted;
-            if (s->nmr->side_inited) {
-                s->nmr->side_ema += 0.125f * (side - s->nmr->side_ema);
-            } else {
-                s->nmr->side_ema    = side;
-                s->nmr->side_inited = 1;
-            }
-        }
-    }
     avpkt->size            = put_bytes_output(&s->pb);
 
-    s->lambda_sum += (s->nmr && s->nmr->lam_rc > 0.0f) ? s->nmr->lam_rc : s->lambda;
+    s->lambda_sum += s->lambda;
     s->lambda_count++;
 
     ff_af_queue_remove(&s->afq, avctx->frame_size, &avpkt->pts,
                        &avpkt->duration);
 
-    int discard_padding = avctx->frame_size - ff_samples_from_time_base(avctx, avpkt->duration);
-    if (discard_padding > 0) {
-        uint8_t *side_data =
-            av_packet_new_side_data(avpkt, AV_PKT_DATA_SKIP_SAMPLES, 10);
-        if (!side_data)
-            return AVERROR(ENOMEM);
-        AV_WL32(side_data + 4, discard_padding);
-    }
-
     avpkt->flags |= AV_PKT_FLAG_KEY;
 
     *got_packet_ptr = 1;
@@ -1434,15 +1133,7 @@ static av_cold int aac_encode_end(AVCodecContext *avctx)
 {
     AACEncContext *s = avctx->priv_data;
 
-    av_log(avctx, AV_LOG_INFO,
-           "Qavg: %.3f  Tr: %.1f%%  TNS(L): %.1f%%  TNS(S): %.1f%%  M/S: %.1f%%  I/S: %.1f%%  PNS: %.1f%%\n",
-           s->lambda_count ? s->lambda_sum / s->lambda_count : NAN,
-           s->stat_chans     ? 100.0 * s->stat_short      / s->stat_chans               : 0.0,
-           s->stat_chans - s->stat_short ? 100.0 * s->stat_tns_long  / (s->stat_chans - s->stat_short) : 0.0,
-           s->stat_short     ? 100.0 * s->stat_tns_short  / s->stat_short               : 0.0,
-           s->stat_cpe_bands ? 100.0 * s->stat_ms         / s->stat_cpe_bands           : 0.0,
-           s->stat_cpe_bands ? 100.0 * s->stat_is         / s->stat_cpe_bands           : 0.0,
-           s->stat_ch_bands  ? 100.0 * s->stat_pns        / s->stat_ch_bands            : 0.0);
+    av_log(avctx, AV_LOG_INFO, "Qavg: %.3f\n", s->lambda_count ? s->lambda_sum / s->lambda_count : NAN);
 
     av_tx_uninit(&s->mdct1024);
     av_tx_uninit(&s->mdct128);
@@ -1451,7 +1142,6 @@ static av_cold int aac_encode_end(AVCodecContext *avctx)
     av_freep(&s->buffer.samples);
     av_freep(&s->cpe);
     av_freep(&s->fdsp);
-    av_freep(&s->nmr);
     ff_af_queue_close(&s->afq);
     return 0;
 }
@@ -1485,12 +1175,6 @@ static av_cold int alloc_buffers(AVCodecContext *avctx, AACEncContext *s)
     for(ch = 0; ch < s->channels; ch++)
         s->planar_samples[ch] = s->buffer.samples + 3 * 1024 * ch;
 
-    if (s->options.coder == AAC_CODER_NMR) {
-        s->nmr = av_mallocz(sizeof(*s->nmr));
-        if (!s->nmr)
-            return AVERROR(ENOMEM);
-    }
-
     return 0;
 }
 
@@ -1498,7 +1182,6 @@ static av_cold int aac_encode_init(AVCodecContext *avctx)
 {
     AACEncContext *s = avctx->priv_data;
     int i, ret = 0;
-    int chcfg;
     const uint8_t *sizes[2];
     uint8_t grouping[AAC_MAX_CHANNELS];
     int lengths[2];
@@ -1513,8 +1196,8 @@ static av_cold int aac_encode_init(AVCodecContext *avctx)
     s->channels = avctx->ch_layout.nb_channels;
 
     s->needs_pce = 1;
-    for (chcfg = 1; chcfg < FF_ARRAY_ELEMS(aac_normal_chan_layouts); chcfg++) {
-        if (!av_channel_layout_compare(&avctx->ch_layout, &aac_normal_chan_layouts[chcfg])) {
+    for (i = 0; i < FF_ARRAY_ELEMS(aac_normal_chan_layouts); i++) {
+        if (!av_channel_layout_compare(&avctx->ch_layout, &aac_normal_chan_layouts[i])) {
             s->needs_pce = s->options.pce;
             break;
         }
@@ -1534,10 +1217,9 @@ static av_cold int aac_encode_init(AVCodecContext *avctx)
         s->pce = aac_pce_configs[i];
         s->reorder_map = s->pce.reorder_map;
         s->chan_map = s->pce.config_map;
-        chcfg = 0;
     } else {
-        s->reorder_map = aac_chan_maps[chcfg - 1];
-        s->chan_map = aac_chan_configs[chcfg - 1];
+        s->reorder_map = aac_chan_maps[s->channels - 1];
+        s->chan_map = aac_chan_configs[s->channels - 1];
     }
 
     if (!avctx->bit_rate) {
@@ -1587,34 +1269,6 @@ static av_cold int aac_encode_init(AVCodecContext *avctx)
     if (s->channels > 3)
         s->options.mid_side = 0;
 
-    /* Coding bandwidth, fixed at init time */
-    if (avctx->cutoff > 0) {
-        s->bandwidth = avctx->cutoff;
-    } else {
-        int frame_br = (avctx->flags & AV_CODEC_FLAG_QSCALE) ?
-                       (avctx->bit_rate / 2.0f * (s->lambda / 120.f) * 1.5f) :
-                       (avctx->bit_rate / avctx->ch_layout.nb_channels);
-
-        /* For NMR, the rate to bandwidth conversion was tuned to maximize metrics
-         * over a variable cutoff x bitrate combo */
-        if (s->options.coder == AAC_CODER_NMR && frame_br >= 32000) {
-            static const int rates[] = { 32000, 48000, 64000, 96000, 192000 };
-            static const int bws[]   = { 14000, 15000, 16000, 18000, 20000 };
-            int bw_i = 0;
-            for (; bw_i < FF_ARRAY_ELEMS(rates) - 2 && frame_br > rates[bw_i + 1]; bw_i++);
-            s->bandwidth = bws[bw_i] + (int)((int64_t)(bws[bw_i + 1] - bws[bw_i]) *
-                                             (frame_br - rates[bw_i]) / (rates[bw_i + 1] - rates[bw_i]));
-            s->bandwidth = FFMIN3(s->bandwidth, 22000, avctx->sample_rate / 2);
-        } else {
-            if (s->options.pns || s->options.intensity_stereo)
-                frame_br *= 1.15f;
-            s->bandwidth = FFMAX(3000, AAC_CUTOFF_FROM_BITRATE(frame_br, 1,
-                                                               avctx->sample_rate));
-        }
-
-        s->bandwidth = FFMIN(FFMAX(s->bandwidth, 8000), avctx->sample_rate / 2);
-    }
-
     // Initialize static tables
     ff_aac_float_common_init();
 
@@ -1624,7 +1278,7 @@ static av_cold int aac_encode_init(AVCodecContext *avctx)
     if ((ret = alloc_buffers(avctx, s)) < 0)
         return ret;
 
-    if ((ret = put_audio_specific_config(avctx, chcfg)))
+    if ((ret = put_audio_specific_config(avctx)))
         return ret;
 
     sizes[0]   = ff_aac_swb_size_1024[s->samplerate_index];
@@ -1634,7 +1288,7 @@ static av_cold int aac_encode_init(AVCodecContext *avctx)
     for (i = 0; i < s->chan_map[0]; i++)
         grouping[i] = s->chan_map[i + 1] == TYPE_CPE;
     if ((ret = ff_psy_init(&s->psy, avctx, 2, sizes, lengths,
-                           s->chan_map[0], grouping, s->bandwidth)) < 0)
+                           s->chan_map[0], grouping)) < 0)
         return ret;
     ff_lpc_init(&s->lpc, 2*avctx->frame_size, TNS_MAX_ORDER, FF_LPC_TYPE_LEVINSON);
     s->random_state = 0x1f2e3d4c;
@@ -1648,16 +1302,14 @@ static av_cold int aac_encode_init(AVCodecContext *avctx)
 
 #define AACENC_FLAGS AV_OPT_FLAG_ENCODING_PARAM | AV_OPT_FLAG_AUDIO_PARAM
 static const AVOption aacenc_options[] = {
-    {"aac_coder", "Coding algorithm", offsetof(AACEncContext, options.coder), AV_OPT_TYPE_INT, {.i64 = AAC_CODER_NMR}, 0, AAC_CODER_NB-1, AACENC_FLAGS, .unit = "coder"},
+    {"aac_coder", "Coding algorithm", offsetof(AACEncContext, options.coder), AV_OPT_TYPE_INT, {.i64 = AAC_CODER_TWOLOOP}, 0, AAC_CODER_NB-1, AACENC_FLAGS, .unit = "coder"},
         {"twoloop",  "Two loop searching method", 0, AV_OPT_TYPE_CONST, {.i64 = AAC_CODER_TWOLOOP}, INT_MIN, INT_MAX, AACENC_FLAGS, .unit = "coder"},
         {"fast",     "Fast search",               0, AV_OPT_TYPE_CONST, {.i64 = AAC_CODER_FAST},    INT_MIN, INT_MAX, AACENC_FLAGS, .unit = "coder"},
-        {"nmr",      "Noise-to-mask ratio scalefactor trellis", 0, AV_OPT_TYPE_CONST, {.i64 = AAC_CODER_NMR}, INT_MIN, INT_MAX, AACENC_FLAGS, .unit = "coder"},
     {"aac_ms", "Force M/S stereo coding", offsetof(AACEncContext, options.mid_side), AV_OPT_TYPE_BOOL, {.i64 = -1}, -1, 1, AACENC_FLAGS},
     {"aac_is", "Intensity stereo coding", offsetof(AACEncContext, options.intensity_stereo), AV_OPT_TYPE_BOOL, {.i64 = 1}, -1, 1, AACENC_FLAGS},
     {"aac_pns", "Perceptual noise substitution", offsetof(AACEncContext, options.pns), AV_OPT_TYPE_BOOL, {.i64 = 1}, -1, 1, AACENC_FLAGS},
     {"aac_tns", "Temporal noise shaping", offsetof(AACEncContext, options.tns), AV_OPT_TYPE_BOOL, {.i64 = 1}, -1, 1, AACENC_FLAGS},
     {"aac_pce", "Forces the use of PCEs", offsetof(AACEncContext, options.pce), AV_OPT_TYPE_BOOL, {.i64 = 0}, -1, 1, AACENC_FLAGS},
-    {"aac_nmr_speed", "NMR coder speed level: 0 = slowest/best, higher trades quality for speed", offsetof(AACEncContext, options.nmr_speed), AV_OPT_TYPE_INT, {.i64 = 0}, 0, 4, AACENC_FLAGS},
     FF_AAC_PROFILE_OPTS
     {NULL}
 };

libavcodec/aacenc.h

@@ -44,7 +44,6 @@
 typedef enum AACCoder {
     AAC_CODER_TWOLOOP,
     AAC_CODER_FAST,
-    AAC_CODER_NMR,
 
     AAC_CODER_NB,
 }AACCoder;
@@ -70,7 +69,6 @@ typedef struct AACEncOptions {
     int pce;
     int mid_side;
     int intensity_stereo;
-    int nmr_speed;          ///< NMR coder speed level: 0 = slowest/best, higher is faster
 } AACEncOptions;
 
 /**
@@ -167,33 +165,11 @@ typedef struct AACQuantizeBandCostCacheEntry {
     uint16_t generation;
 } AACQuantizeBandCostCacheEntry;
 
-/** per-band scalefactor candidates above the finest codeable sf (NMR coder) */
-#define NMR_NCAND 96
-
-/**
- * NMR coder per-band candidate cost curves (~96 KiB) and rate-control carry-over
- */
-typedef struct AACNMRCurves {
-    float nd[128][NMR_NCAND];                    ///< dist / threshold per candidate
-    int   nb[128][NMR_NCAND];                    ///< spectral bits per candidate
-    float lam[16];                               ///< per-channel operating lambda of the previous frame, 0 = none yet
-    int   counted[16];                           ///< per-channel bits the trellis accounted for in the last solve
-    float side_ema;                              ///< running estimate of real-minus-counted bits per frame
-    int   side_inited;                           ///< side_ema holds a measurement
-
-    int64_t rc_frame_num;                        ///< frame the reservoir was last advanced for
-    float   lam_rc;                              ///< global-lambda rate control: operating lambda, 0 until bootstrapped
-    int     rc_fill;                             ///< virtual bit reservoir fill, + = bits saved vs nominal
-    int     frames_since_short;                  ///< long-block frames since the last short run (the "gap"): large = isolated transient
-    int     prev_was_short;                      ///< previous frame was a short block (for run-start detection)
-    float   run_burst;                           ///< transient bit-burst factor, set at run start and held across the short run
-} AACNMRCurves;
-
 typedef struct AACPCEInfo {
     AVChannelLayout layout;
-    uint8_t num_ele[4];                          ///< front, side, back, lfe
-    uint8_t pairing[3][8];                       ///< front, side, back
-    uint8_t index[4][8];                         ///< front, side, back, lfe
+    int num_ele[4];                              ///< front, side, back, lfe
+    int pairing[3][8];                           ///< front, side, back
+    int index[4][8];                             ///< front, side, back, lfe
     uint8_t config_map[16];                      ///< configs the encoder's channel specific settings
     uint8_t reorder_map[16];                     ///< maps channels from lavc to aac order
 } AACPCEInfo;
@@ -218,7 +194,6 @@ typedef struct AACEncContext {
     LPCContext lpc;                              ///< used by TNS
     int samplerate_index;                        ///< MPEG-4 samplerate index
     int channels;                                ///< channel count
-    int bandwidth;                               ///< coding bandwidth in Hz, fixed at init; the psy model and the coders' band cutoff agree on it
     const uint8_t *reorder_map;                  ///< lavc to aac reorder map
     const uint8_t *chan_map;                     ///< channel configuration map
 
@@ -231,12 +206,6 @@ typedef struct AACEncContext {
     int last_frame_pb_count;                     ///< number of bits for the previous frame
     float lambda_sum;                            ///< sum(lambda), for Qvg reporting
     int lambda_count;                            ///< count(lambda), for Qvg reporting
-    /* tool-usage stats, reported at close: per-coded-band for PNS (channel bands),
-     * per-coded-pair-band for M/S and I/S (CPE bands) */
-    uint64_t stat_ch_bands, stat_pns;            ///< coded channel-bands, of which PNS
-    uint64_t stat_cpe_bands, stat_ms, stat_is;   ///< coded CPE pair-bands, of which M/S, I/S
-    uint64_t stat_chans, stat_short;             ///< coded channels, of which short-block (transient)
-    uint64_t stat_tns_long, stat_tns_short;      ///< TNS-active channels among long / short blocks
     enum RawDataBlockType cur_type;              ///< channel group type cur_channel belongs to
 
     AudioFrameQueue afq;
@@ -247,7 +216,6 @@ typedef struct AACEncContext {
     AACQuantizeBandCostCacheEntry quantize_band_cost_cache[256][128]; ///< memoization area for quantize_band_cost
 
     AACEncDSPContext aacdsp;
-    AACNMRCurves *nmr;                            ///< NMR coder scratch (NULL unless coder == nmr)
 
     struct {
         float *samples;

libavcodec/aacenc_tns.c

@@ -41,13 +41,9 @@
 /* We really need the bits we save here elsewhere */
 #define TNS_ENABLE_COEF_COMPRESSION
 
-/* Apple-derived TNS: weighted-spectrum predictor, accepted only if the measured
- * post-quantization prediction gain clears a block-type-dependent bar (Apple RE). */
-#define TNS_PREDGAIN_GATE   1.4f    /* first gate: predicted LPC gain */
-#define TNS_PG_C1_LONG      1.4f    /* min measured gain, long blocks */
-#define TNS_PG_C1_SHORT     3.2f    /* min measured gain, short blocks */
-#define TNS_PG_CLAMP        6.0f    /* upper bound: poles near unit circle → noise blowup */
-#define TNS_WEIGHT_FLOOR    0.01f   /* per-bin masking floor for the weighted spectrum */
+/* TNS will only be used if the LPC gain is within these margins */
+#define TNS_GAIN_THRESHOLD_LOW      1.4f
+#define TNS_GAIN_THRESHOLD_HIGH     1.16f*TNS_GAIN_THRESHOLD_LOW
 
 static inline int compress_coeffs(int *coef, int order, int c_bits)
 {
@@ -66,7 +62,11 @@ static inline int compress_coeffs(int *coef, int order, int c_bits)
     return 1;
 }
 
-/** Encode TNS data. */
+/**
+ * Encode TNS data.
+ * Coefficient compression is simply not lossless as it should be
+ * on any decoder tested and as such is not active.
+ */
 void ff_aac_encode_tns_info(AACEncContext *s, SingleChannelElement *sce)
 {
     TemporalNoiseShaping *tns = &sce->tns;
@@ -98,28 +98,15 @@ void ff_aac_encode_tns_info(AACEncContext *s, SingleChannelElement *sce)
     }
 }
 
-/* Cap the TNS band range at the first PNS band to avoid TNS+PNS conflicts. */
-static int tns_max_nonpns(const SingleChannelElement *sce, int mmm)
-{
-    for (int w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w])
-        for (int g = 0; g < mmm; g++)
-            if (sce->band_type[w*16+g] == NOISE_BT) { mmm = g; break; }
-    return mmm;
-}
-
 /* Apply TNS filter */
 void ff_aac_apply_tns(AACEncContext *s, SingleChannelElement *sce)
 {
     TemporalNoiseShaping *tns = &sce->tns;
     IndividualChannelStream *ics = &sce->ics;
     int w, filt, m, i, top, order, bottom, start, end, size, inc;
-    const int mmm = tns_max_nonpns(sce, FFMIN(ics->tns_max_bands, ics->max_sfb));
+    const int mmm = FFMIN(ics->tns_max_bands, ics->max_sfb);
     float lpc[TNS_MAX_ORDER];
 
-    /* TNS predicts from the post-M/S and post-I/S coefficients. */
-    float hist[1024];
-    memcpy(hist, sce->coeffs, sizeof(hist));
-
     for (w = 0; w < ics->num_windows; w++) {
         bottom = ics->num_swb;
         for (filt = 0; filt < tns->n_filt[w]; filt++) {
@@ -147,7 +134,7 @@ void ff_aac_apply_tns(AACEncContext *s, SingleChannelElement *sce)
             /* AR filter */
             for (m = 0; m < size; m++, start += inc) {
                 for (i = 1; i <= FFMIN(m, order); i++) {
-                    sce->coeffs[start] += lpc[i-1]*hist[start - i*inc];
+                    sce->coeffs[start] += lpc[i-1]*sce->pcoeffs[start - i*inc];
                 }
             }
         }
@@ -174,8 +161,9 @@ static inline void quantize_coefs(double *coef, int *idx, float *lpc, int order,
 void ff_aac_search_for_tns(AACEncContext *s, SingleChannelElement *sce)
 {
     TemporalNoiseShaping *tns = &sce->tns;
-    int w, count = 0;
-    const int mmm = tns_max_nonpns(sce, FFMIN(sce->ics.tns_max_bands, sce->ics.max_sfb));
+    int w, g, count = 0;
+    double gain, coefs[MAX_LPC_ORDER];
+    const int mmm = FFMIN(sce->ics.tns_max_bands, sce->ics.max_sfb);
     const int is8 = sce->ics.window_sequence[0] == EIGHT_SHORT_SEQUENCE;
     const int c_bits = is8 ? TNS_Q_BITS_IS8 == 4 : TNS_Q_BITS == 4;
     const int sfb_start = av_clip(tns_min_sfb[is8][s->samplerate_index], 0, mmm);
@@ -186,127 +174,56 @@ void ff_aac_search_for_tns(AACEncContext *s, SingleChannelElement *sce)
     const int sfb_len = sfb_end - sfb_start;
     const int coef_len = sce->ics.swb_offset[sfb_end] - sce->ics.swb_offset[sfb_start];
     const int n_filt = is8 ? 1 : order != TNS_MAX_ORDER ? 2 : 3;
-    const int ord_g  = order / n_filt;
-
-    /* Apple's accept bar (minimum measured prediction gain): higher on short blocks,
-     * where a weak filter's shaped-noise tail spreads across the 50% overlap. */
-    const float c1 = is8 ? TNS_PG_C1_SHORT : TNS_PG_C1_LONG;
-    FFPsyBand *const psy_bands = &s->psy.ch[s->cur_channel].psy_bands[0];
 
     if (coef_len <= 0 || sfb_len <= 0) {
         sce->tns.present = 0;
         return;
     }
 
-    /* time-domain window length backing one coding window: a long MDCT block is
-     * fed 2048 windowed samples (current 1024 + overlap), each short block 256. */
-    const int tlen = is8 ? 256 : 2048;
-
     for (w = 0; w < sce->ics.num_windows; w++) {
-        int filt, any = 0;
+        float en[4] = {0.0f, 0.0f, 0.0f, 0.0f};
+        int oc_start = 0;
+        int coef_start = sce->ics.swb_offset[sfb_start];
 
-        /* The filter gets ran in the direction of the signal's *temporal* energy,
-         * so the quantization noise stays in the loud masked part rather than spilling
-         * into the quiet part. */
-        const float *tw = sce->ret_buf + w*tlen;
-        float e_early = 0.0f, e_late = 0.0f;
-        int ti;
-        for (ti = 0; ti < tlen/2; ti++)
-            e_early += tw[ti]*tw[ti];
-        for (; ti < tlen; ti++)
-            e_late += tw[ti]*tw[ti];
-        const int tdir = e_early > e_late;
-
-        /* Walk the frequency regions exactly as the decoder does: filter 0 is the
-         * topmost band region, each subsequent filter covers the next region down,
-         * clamped to mmm. Each filter gets its own LPC over its own region. */
-        int top_sfb = sce->ics.num_swb;
-        for (filt = 0; filt < n_filt; filt++) {
-            double coefs[MAX_LPC_ORDER];
-            float wspec[1024], tmp[1024], lpc_q[TNS_MAX_ORDER];
-            int len_sfb = (filt == n_filt - 1) ? sfb_len - filt*(sfb_len/n_filt)
-                                               : sfb_len/n_filt;
-            int bot_sfb = FFMAX(0, top_sfb - len_sfb);
-            int g_lo = FFMIN(bot_sfb, mmm), g_hi = FFMIN(top_sfb, mmm);
-            int c_lo = sce->ics.swb_offset[g_lo];
-            int c_hi = sce->ics.swb_offset[g_hi];
-            int clen = c_hi - c_lo;
-            const int dir = slant != 2 ? slant : tdir;
-            float gain, orig_e = 0.0f, filt_e = 0.0f;
-            int m, i, g, inc, st;
-
-            tns->length[w][filt] = len_sfb;
-            tns->order[w][filt]  = 0;     /* default: region carries no filter */
-            top_sfb = bot_sfb;
-
-            if (clen <= 2*ord_g)          /* too short for a stable order-ord_g LPC */
-                continue;
-
-            /* Fit LPC on the perceptually-weighted spectrum X/sqrt(thr), floored
-             * to avoid a near-zero threshold blowing up a single bin (Apple). */
-            {
-                float maxrms = 0.0f, floorrms;
-                int k;
-                for (g = g_lo; g < g_hi; g++) {
-                    int s0 = sce->ics.swb_offset[g], s1 = sce->ics.swb_offset[g+1];
-                    float rms = sqrtf(FFMAX(psy_bands[w*16 + g].threshold, 0.0f) /
-                                      FFMAX(s1 - s0, 1));
-                    maxrms = FFMAX(maxrms, rms);
-                }
-                floorrms = FFMAX(maxrms * TNS_WEIGHT_FLOOR, 1e-9f);
-                for (g = g_lo; g < g_hi; g++) {
-                    int s0 = sce->ics.swb_offset[g], s1 = sce->ics.swb_offset[g+1];
-                    float rms = sqrtf(FFMAX(psy_bands[w*16 + g].threshold, 0.0f) /
-                                      FFMAX(s1 - s0, 1));
-                    float wgt = 1.0f / FFMAX(rms, floorrms);
-                    for (k = s0; k < s1; k++)
-                        wspec[k - c_lo] = sce->coeffs[w*128 + k] * wgt;
-                }
-                /* Short blocks: unwindowed fit; Hann window zeros the edges of the
-                 * tiny region, wrecking the LPC. Long blocks keep the window. */
-                gain = ff_lpc_calc_ref_coefs_f(&s->lpc, wspec, clen, ord_g, coefs, !is8);
+        if (n_filt == 2) {
+            for (g = sfb_start; g < sce->ics.num_swb && g <= sfb_end; g++) {
+                FFPsyBand *band = &s->psy.ch[s->cur_channel].psy_bands[w*16+g];
+                    if (g > sfb_start + (sfb_len/2))
+                        en[1] += band->energy; /* End */
+                    else
+                        en[0] += band->energy; /* Start */
             }
-            /* Reject below the first gate and above the clamp (poles near unit circle). */
-            if (!isfinite(gain) || gain < TNS_PREDGAIN_GATE || gain > TNS_PG_CLAMP)
-                continue;
-            /* Negate: ff_lpc_calc_ref_coefs_f sign convention is opposite to what
-             * ff_aac_apply_tns's MA filter needs; fed unnegated, it anti-whitens. */
-            for (i = 0; i < ord_g; i++)
-                coefs[i] = -coefs[i];
-
-            /* Quantize, then build the decoder's direct-form LPC. */
-            quantize_coefs(coefs, tns->coef_idx[w][filt], tns->coef[w][filt],
-                           ord_g, c_bits);
-            compute_lpc_coefs(tns->coef[w][filt], 0, ord_g, lpc_q, 0, 0, 0, NULL);
-
-            /* Apply the quantized filter to the weighted spectrum and measure gain. */
-            const float *msrc = wspec;
-            inc = dir ? -1 : 1;
-            st  = dir ? clen - 1 : 0;
-            for (m = 0; m < clen; m++) {
-                int idx = st + m*inc;
-                float acc = msrc[idx];
-                for (i = 1; i <= FFMIN(m, ord_g); i++)
-                    acc += lpc_q[i-1] * msrc[idx - i*inc];
-                tmp[idx] = acc;
+            en[2] = en[0];
+        } else {
+            for (g = sfb_start; g < sce->ics.num_swb && g <= sfb_end; g++) {
+                FFPsyBand *band = &s->psy.ch[s->cur_channel].psy_bands[w*16+g];
+                    if (g > sfb_start + (sfb_len/2) + (sfb_len/4))
+                        en[2] += band->energy; /* End */
+                    else if (g > sfb_start + (sfb_len/2) - (sfb_len/4))
+                        en[1] += band->energy; /* Middle */
+                    else
+                        en[0] += band->energy; /* Start */
             }
-            for (m = 0; m < clen; m++) {
-                orig_e += msrc[m]*msrc[m];
-                filt_e += tmp[m]*tmp[m];
-            }
-            filt_e = FFMAX(filt_e, 1e-9f);
-
-            /* Keep only if measured post-quantization gain clears C1 (Apple's outcome gate). */
-            if (orig_e < c1*filt_e)
-                continue;
-
-            tns->order[w][filt] = ord_g;
-            tns->direction[w][filt] = dir;
-            any = 1;
+            en[3] = en[0];
         }
-        tns->n_filt[w] = any ? n_filt : 0;
-        if (any)
-            count++;
+
+        /* LPC */
+        gain = ff_lpc_calc_ref_coefs_f(&s->lpc, &sce->coeffs[w*128 + coef_start],
+                                       coef_len, order, coefs);
+
+        if (!order || !isfinite(gain) || gain < TNS_GAIN_THRESHOLD_LOW || gain > TNS_GAIN_THRESHOLD_HIGH)
+            continue;
+
+        tns->n_filt[w] = n_filt;
+        for (g = 0; g < tns->n_filt[w]; g++) {
+            tns->direction[w][g] = slant != 2 ? slant : en[g] < en[g + 1];
+            tns->order[w][g] = order/tns->n_filt[w];
+            tns->length[w][g] = sfb_len/tns->n_filt[w];
+            quantize_coefs(&coefs[oc_start], tns->coef_idx[w][g], tns->coef[w][g],
+                            tns->order[w][g], c_bits);
+            oc_start += tns->order[w][g];
+        }
+        count++;
     }
     sce->tns.present = !!count;
 }