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18 Commits

Author SHA1 Message Date
Michael Niedermayer b08d7969c5 RELEASE_NOTES: Based on the version from 5.1
Name suggested by 3 people on ML

Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2024-09-29 18:47:04 +02:00
Michael Niedermayer 69cf68f5a0 Update for 7.1
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2024-09-29 18:47:04 +02:00
Michael Niedermayer 0bad953cc5 avcodec/ffv1enc: Fix >8bit context size
Fixes: Ticket5405

Sponsored-by: Sovereign Tech Fund
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit a9c83e43f2)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2024-09-29 18:47:03 +02:00
Martin Storsjö 09e9d61d0a checkasm/sw_rgb: Revert test additions from e18b46d95f
The unaligned width test cases fail on i386; we have an assembly
function of rgb24toyv12 which is enabled only within
"#if ARCH_X86_32 && HAVE_7REGS", which seems to fail these new
test cases for unaligned widths.

As that assembly function has existed for a long time in that form,
the issue probably isn't very recent, thus skip testing these cases
for now.

Once the assembly function has been fixed, these test cases can
be readded.

Signed-off-by: Martin Storsjö <martin@martin.st>
(cherry picked from commit 157ce21939)
Signed-off-by: Martin Storsjö <martin@martin.st>
2024-09-29 14:23:06 +03:00
Marvin Scholz ebd0ca9fee avdevice/decklink_dec: fix leaks on error
In case of errors in this function, the allocated context
was not properly freed in several cases.

Signed-off-by: Marton Balint <cus@passwd.hu>
(cherry picked from commit 8e76c99301)
2024-09-29 10:30:49 +02:00
Marvin Scholz 963145483c avdevice/decklink_dec: fix leak on error
In the early return when both draw_bars and signal_loss_action
options are used, the context allocated previously was not
properly freed.

Introduced in 9bcb86b0fa

Fixes CID1619296

Signed-off-by: Marton Balint <cus@passwd.hu>
(cherry picked from commit 7554346cb5)
2024-09-29 10:30:32 +02:00
Brad Smith 05ff3eb9f2 ppc: Recognize the PPC VSX and Power8 CPU flags
Signed-off-by: Brad Smith <brad@comstyle.com>
(cherry picked from commit 6ec6b07900)
Signed-off-by: Brad Smith <brad@comstyle.com>
2024-09-30 09:08:11 -04:00
Anton Khirnov d5c308fcc5 lavu/opt: avoid accidentally propagating spurious errors
An error from read_number() is non-fatal here and should not be
forwarded.

(cherry picked from commit 505fea34e5)
Signed-off-by: Anton Khirnov <anton@khirnov.net>
2024-09-28 17:14:14 +02:00
Anton Khirnov 55d0c353fe lavu/opt: add missing 'else'
(cherry picked from commit 9b764e9b49)
Signed-off-by: Anton Khirnov <anton@khirnov.net>
2024-09-28 17:14:11 +02:00
Lynne 9ada04faa3 vulkan_encode_h265: fix rate control VBV values
The values written were placeholder values.

(cherry picked from commit 81c6e6c9ee)
2024-09-27 09:58:13 +02:00
Lynne 0482410818 vulkan_encode_h264: fix rate control VBV values
The values must be in milliseconds, not bytes.

(cherry picked from commit 934be0ff50)
2024-09-27 09:58:13 +02:00
James Almer c21472ec31 avutil/iamf: fix doxygen
Signed-off-by: James Almer <jamrial@gmail.com>
(cherry picked from commit d55ab2ba9b)
2024-09-26 16:43:30 -03:00
James Almer cb23275623 tests/fate/mov: fix rules for some tests
This was broken in 2fb2cd5c79 and
74553f0026, making a simple make fate-mov fail
and most of the test not run with a simple make fate.

Signed-off-by: James Almer <jamrial@gmail.com>
(cherry picked from commit 8b4e32f30b)
2024-09-24 21:10:40 -03:00
James Almer f7643454aa avfilter/vf_showinfo: add missing break to ViewID case
Missed in 6940a6de2f.

Signed-off-by: James Almer <jamrial@gmail.com>
(cherry picked from commit fd6cc18d60)
2024-09-24 16:09:05 -03:00
James Almer 2db706374e avcodec/hevc/sei: num_ref_displays can be up to 32
Signed-off-by: James Almer <jamrial@gmail.com>
2024-09-24 15:13:19 -03:00
James Almer a6a346e154 avcodec/cbs_h265: num_ref_displays can be up to 32
Signed-off-by: James Almer <jamrial@gmail.com>
2024-09-24 15:13:18 -03:00
James Almer 00ce36d037 avcodec/cbs_h265: fix valid range for {left,right}_view_id
view_id_len in VPS is 4 bits, so view_id values can be up to 15 bits long.

Signed-off-by: James Almer <jamrial@gmail.com>
2024-09-24 15:13:18 -03:00
Anton Khirnov f459c56b0f tests/fate/hevc: use bitexact scaling flags for fate-hevc-mv-switch
Makes the results consistent across platforms.

(cherry picked from commit 9dc0edd454)
Signed-off-by: Anton Khirnov <anton@khirnov.net>
2024-09-24 18:43:33 +02:00
5437 changed files with 115813 additions and 304791 deletions
-262
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# This file describes the expected reviewers for a PR based on the changed
# files. Unlike what the name of the file suggests they don't own the code, but
# merely have a good understanding of that area of the codebase and therefore
# are usually suited as a reviewer.
# Lines in this file match changed paths via Go-Style regular expressions:
# https://pkg.go.dev/regexp/syntax
# Mind the alphabetical order
# avcodec
# =======
libavcodec/.*aac.* @lynne
libavcodec/.*ac3.* @lynne
libavcodec/.*adpcm.* @zane @pross
libavcodec/anm.* @pross
libavcodec/amf.* @OvchinnikovDmitrii @ArazIusubov
libavcodec/ansi.* @pross
libavcodec/aom_film_grain.* @haasn
libavcodec/.*atrac9.* @lynne
libavcodec/bink.* @pross
libavcodec/bintext.* @pross
libavcodec/.*bitpacked.* @lynne
libavcodec/.*d3d12va.* @jianhuaw @tong1wu @younengxiao
libavcodec/.*dirac.* @lynne
libavcodec/.*dovi_rpu.* @haasn
libavcodec/dpx.* @pross
libavcodec/dsd.* @pross
libavcodec/eacmv.* @pross
libavcodec/eaidct.* @pross
libavcodec/eamad.* @pross
libavcodec/eat.* @pross
libavcodec/.*exif.* @Traneptora
libavcodec/.*ffv1.* @lynne @michaelni
libavcodec/g728.* @pross
libavcodec/gem.* @pross
libavcodec/golomb.* @michaelni
libavcodec/.*h266.* @frankplow @NuoMi @jianhuaw
libavcodec/h26x/.* @frankplow @NuoMi @jianhuaw
libavcodec/.*h274.* @haasn
libavcodec/iff.* @pross
libavcodec/.*jpegxl.* @lynne @Traneptora
libavcodec/jpeg2000htdec.* @cae
libavcodec/jv.* @pross
libavcodec/.*jxl.* @lynne @Traneptora
libavcodec/.*lcms2.* @haasn
libavcodec/lead.* @pross
libavcodec/mediacodec* @quink
libavcodec/mjpeg* @michaelni
libavcodec/mmvideo.* @pross
libavcodec/msp2.* @pross
libavcodec/mvc.* @pross
libavcodec/oh* @quink
libavcodec/.*opus.* @lynne
libavcodec/pictor.* @pross
libavcodec/.*png.* @Traneptora
libavcodec/.*prores.* @lynne
libavcodec/rangecoder.* @michaelni
libavcodec/ratecontrol.* @michaelni
libavcodec/rkmpp* @quink
libavcodec/rv60.* @pross
libavcodec/sgirle.* @pross
libavcodec/.*siren.* @lynne
libavcodec/smpte_436m.* @programmerjake
libavcodec/svq1.* @pross
libavcodec/svq3.* @pross
libavcodec/.*vc2.* @lynne
libavcodec/videotoolbox.* @ePirat
libavcodec/vp3.* @pross
libavcodec/vp4.* @pross
libavcodec/vp5.* @pross
libavcodec/vp6.* @pross
libavcodec/vp8.* @rbultje @pross
libavcodec/vp9.* @rbultje
libavcodec/vpx.* @rbultje @pross
libavcodec/vqc.* @pross
libavcodec/.*vvc.* @frankplow @NuoMi @jianhuaw
libavcodec/wmavoice.* @rbultje
libavcodec/wbmp.* @pross
# bitstream filters
libavcodec/bsf/eia608_to_smpte436m.* @programmerjake
libavcodec/bsf/smpte436m_to_eia608.* @programmerjake
# architecture-specific
libavcodec/aarch64/.* @lynne @mstorsjo
libavcodec/arm/.* @mstorsjo
libavcodec/ppc/.* @sean_mcg
libavcodec/riscv/.* @Courmisch
libavcodec/wasm/hevc/.* @quink
libavcodec/x86/.* @lynne
libavcodec/x86/vp8.* @rbultje
libavcodec/x86/vp9.* @rbultje
libavcodec/x86/vpx.* @rbultje
# avfilter
# =======
libavfilter/af_loudnorm.* @kylophone
libavfilter/af_whisper.* @vpalmisano
libavfilter/.*_amf* @OvchinnikovDmitrii @ArazIusubov
libavfilter/avfiltergraph.* @haasn
libavfilter/colorspace.* @rbultje
libavfilter/formats.* @haasn
libavfilter/.*f_ebur128.* @haasn
libavfilter/vf_blackdetect.* @haasn
libavfilter/vf_colordetect.* @haasn
libavfilter/vf_colorspace.* @rbultje
libavfilter/.*drawvg.* @ayosec
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
libavfilter/vf_thumbnail.* @haasn
libavfilter/vf_transpose_vt.* @quink
libavfilter/vf_yadif.* @michaelni
libavfilter/vsrc_mandelbrot.* @michaelni
libavfilter/aarch64/.* @mstorsjo
libavfilter/riscv/.* @Courmisch
libavfilter/x86/colorspace.* @rbultje
libavfilter/x86/scene_sad.* @haasn
# avformat
# =======
libavformat/alp.* @zane
libavformat/amv.* @zane
libavformat/anm.* @pross
libavformat/apm.* @zane
libavformat/argo_.* @zane
libavformat/bink.* @pross
libavformat/bintext.* @pross
libavformat/caf.* @pross
libavformat/cine.* @pross
libavformat/dsf.* @pross
libavformat/eacdata.* @pross
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
libavformat/iff.* @pross
libavformat/.*jpegxl.* @Traneptora
libavformat/jv.* @pross
libavformat/.*jxl.* @Traneptora
libavformat/kvag.* @zane
libavformat/mccdec.* @programmerjake
libavformat/mccenc.* @programmerjake
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
libavformat/scd.* @zane
libavformat/tty.* @pross
libavformat/whip.c @JackLau
libavformat/wsd.* @pross
libavformat/wtv.* @pross
# avutil
# ======
libavutil/.*_amf* @OvchinnikovDmitrii @ArazIusubov
libavutil/.*crc.* @lynne @michaelni
libavutil/.*d3d12va.* @jianhuaw @tong1wu @younengxiao
libavutil/csp.* @rbultje @haasn
libavutil/eval.* @michaelni
libavutil/film_grain.* @haasn
libavutil/dovi_meta.* @haasn
libavutil/hwcontext_oh.* @quink
libavutil/hwcontext_mediacodec.* @quink
libavutil/hwcontext_videotoolbox.* @ePirat
libavutil/iamf.* @jamrial
libavutil/integer.* @michaelni
libavutil/lfg.* @michaelni
libavutil/lls.* @michaelni
libavutil/md5.* @michaelni
libavutil/mathematics.* @michaelni
libavutil/mem.* @michaelni
libavutil/qsort.* @michaelni
libavutil/random_seed.* @michaelni
libavutil/rational.* @michaelni
libavutil/sfc.* @michaelni
libavutil/softfloat.* @michaelni
libavutil/tree.* @michaelni
libavutil/tx.* @lynne
libavutil/aarch64/.* @lynne @mstorsjo
libavutil/arm/.* @mstorsjo
libavutil/ppc/.* @sean_mcg
libavutil/riscv/.* @Courmisch
libavutil/wasm/.* @quink
libavutil/x86/.* @lynne
# swresample
# =======
libswresample/aarch64/.* @mstorsjo
libswresample/arm/.* @mstorsjo
libswresample/.* @michaelni
# swscale
# =======
libswscale/aarch64/.* @mstorsjo
libswscale/arm/.* @mstorsjo
libswscale/ppc/.* @sean_mcg
libswscale/riscv/.* @Courmisch
libswscale/.* @haasn
# tools
# =====
fftools/ffplay_renderer.* @haasn
tools/target_dec_fuzzer.c @michaelni
# doc
# ===
doc/.* @GyanD
# Frameworks
# ==========
.*d3d12va.* @jianhuaw @tong1wu @younengxiao
.*vulkan.* @lynne @haasn
# 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/CODEOWNERS @michaelni
# /
# =
MAINTAINERS @michaelni
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# Summary of the bug
Briefly describe the issue you're experiencing. Include any error messages, unexpected behavior, or relevant observations.
# Steps to reproduce
List the steps required to trigger the bug.
Include the exact CLI command used, if any.
Provide sample input files, logs, or scripts if available.
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@@ -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
```
-->
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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; }
-174
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#!/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()
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module.exports = async ({github, context}) => {
const title = (context.payload.pull_request?.title || context.payload.issue?.title || '').toLowerCase();
const labels = [];
const issueNumber = context.payload.pull_request?.number || context.payload.issue?.number;
const kwmap = {
'avcodec': 'avcodec',
'avdevice': 'avdevice',
'avfilter': 'avfilter',
'avformat': 'avformat',
'avutil': 'avutil',
'swresample': 'swresample',
'swscale': 'swscale',
'fftools': 'CLI',
'vulkan': 'vulkan'
};
async function isOrgMember(username) {
try {
const response = await github.rest.orgs.checkMembershipForUser({
org: context.repo.owner,
username: username
});
return response.status === 204;
} catch (error) {
return false;
}
}
if (context.payload.action === 'closed' ||
(context.payload.action !== 'opened' && (
context.payload.action === 'assigned' ||
context.payload.action === 'label_updated' ||
context.payload.action === 'labeled' ||
context.payload.comment) &&
await isOrgMember(context.payload.sender.login))
) {
try {
await github.rest.issues.removeLabel({
owner: context.repo.owner,
repo: context.repo.repo,
issue_number: issueNumber,
// this should say 'new', but forgejo deviates from GitHub API here and expects the ID
name: '41'
});
console.log('Removed "new" label');
} catch (error) {
if (error.status !== 404 && error.status !== 410) {
console.log('Could not remove "new" label');
}
}
} else if (context.payload.action === 'opened') {
labels.push('new');
console.log('Detected label: new');
}
if ((context.payload.action === 'opened' || context.payload.action === 'edited') && context.eventName !== 'issue_comment') {
for (const [kw, label] of Object.entries(kwmap)) {
if (title.includes(kw)) {
labels.push(label);
console.log('Detected label: ' + label);
}
}
}
if (labels.length > 0) {
await github.rest.issues.addLabels({
owner: context.repo.owner,
repo: context.repo.repo,
issue_number: issueNumber,
labels: labels,
});
}
}
-35
View File
@@ -1,35 +0,0 @@
avcodec:
- changed-files:
- any-glob-to-any-file: 'libavcodec/**'
avdevice:
- changed-files:
- any-glob-to-any-file: 'libavdevice/**'
avfilter:
- changed-files:
- any-glob-to-any-file: 'libavfilter/**'
avformat:
- changed-files:
- any-glob-to-any-file: 'libavformat/**'
avutil:
- changed-files:
- any-glob-to-any-file: 'libavutil/**'
swresample:
- changed-files:
- any-glob-to-any-file: 'libswresample/**'
swscale:
- changed-files:
- any-glob-to-any-file: 'libswscale/**'
CLI:
- changed-files:
- any-glob-to-any-file: 'fftools/**'
vulkan:
- changed-files:
- any-glob-to-any-file: '**/*vulkan*'
-51
View File
@@ -1,51 +0,0 @@
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
hooks:
- id: check-case-conflict
- id: check-executables-have-shebangs
- id: check-illegal-windows-names
- id: check-shebang-scripts-are-executable
- id: check-yaml
- id: end-of-file-fixer
- id: file-contents-sorter
files:
.forgejo/pre-commit/ignored-words.txt
args:
- --ignore-case
- id: fix-byte-order-marker
- id: mixed-line-ending
- 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$
language: script
entry: ./tools/check_arm_indent.sh --apply
pass_filenames: false
- repo: https://github.com/codespell-project/codespell
rev: v2.4.1
hooks:
- id: codespell
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/.*$
-120
View File
@@ -1,120 +0,0 @@
abl
ACN
acount
addin
alis
alls
ALOG
ALS
als
ANC
anc
ANS
ans
anull
basf
bloc
brane
BREIF
BU
bu
bufer
CAF
caf
clen
clens
Collet
compre
dum
endin
erro
FIEL
fiel
filp
fils
FILTERD
filterd
fle
fo
FPR
fro
Hald
indx
ine
inh
inout
inouts
inport
ist
LAF
laf
lastr
LinS
mapp
mis
mot
nd
nIn
offsetp
orderd
ot
outout
padd
PAETH
paeth
PARM
parm
parms
pEvents
PixelX
Psot
quater
re-use
readd
recuse
redY
Reencode
reencode
remaind
renderD
rin
SAV
SEH
SER
ser
setts
shft
SIZ
siz
skipd
sme
som
sover
STAP
startd
statics
struc
suble
TE
tE
te
tha
tne
tolen
tpye
tre
TRUN
trun
truns
Tung
TYE
ue
UES
ues
vai
vas
vie
VILL
vor
wel
wih
-32
View File
@@ -1,32 +0,0 @@
name: Autolabel
on:
pull_request_target:
types: [opened, edited, synchronize, closed, assigned, labeled, unlabeled]
issues:
types: [opened, edited, closed, assigned, labeled, unlabeled]
issue_comment:
types: [created]
jobs:
pr_labeler:
name: Labeler
runs-on: utilities
if: ${{ github.event.sender.login != 'ffmpeg-devel' }}
steps:
- name: Checkout
uses: actions/checkout@v6
- name: Label by file-changes
uses: actions/labeler@v6
if: ${{ forge.event_name == 'pull_request_target' }}
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:
script: |
const script = require('.forgejo/labeler/labeler.js')
await script({github, context})
github-token: ${{ secrets.AUTOLABELER_TOKEN }}
-34
View File
@@ -1,34 +0,0 @@
name: Lint
on:
push:
branches:
- master
pull_request:
concurrency:
cancel-in-progress: ${{ forge.event_name == 'pull_request' }}
jobs:
lint:
name: Pre-Commit
runs-on: utilities
steps:
- name: Checkout
uses: actions/checkout@v6
- name: Rebase onto target branch
uses: ./.forgejo/actions/rebase-pr
- name: Install pre-commit CI
id: install
run: |
python3 -m venv ~/pre-commit
~/pre-commit/bin/pip install --upgrade pip setuptools
~/pre-commit/bin/pip install pre-commit
echo "envhash=$({ python3 --version && cat .forgejo/pre-commit/config.yaml; } | sha256sum | cut -d' ' -f1)" >> $FORGEJO_OUTPUT
- name: Cache
uses: actions/cache@v5
with:
path: ~/.cache/pre-commit
key: pre-commit-${{ steps.install.outputs.envhash }}
- name: Run pre-commit CI
run: ~/pre-commit/bin/pre-commit run -c .forgejo/pre-commit/config.yaml --show-diff-on-failure --color=always --all-files
-145
View File
@@ -1,145 +0,0 @@
name: Test
on:
push:
branches:
- master
pull_request:
concurrency:
cancel-in-progress: ${{ forge.event_name == 'pull_request' }}
jobs:
run_fate:
name: Fate (${{ matrix.runner }}, ${{ matrix.shared }}, ${{ matrix.bits }} bit)
strategy:
fail-fast: false
matrix:
runner: [linux-aarch64]
shared: ['static']
bits: ['64']
include:
- runner: linux-amd64
shared: 'static'
bits: '32'
- runner: linux-amd64
shared: 'shared'
bits: '64'
runs-on: ${{ matrix.runner }}
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 \
$([ "${{ matrix.bits }}" != "32" ] || echo --arch=x86_32 --extra-cflags=-m32 --extra-cxxflags=-m32 --extra-ldflags=-m32) \
$([ "${{ matrix.shared }}" != "shared" ] || echo --enable-shared --disable-static) \
|| CFGRES=$?
cat ffbuild/config.log
exit ${CFGRES:-0}
- name: Build
run: make -j$(nproc)
- name: Restore Cached Fate-Suite
id: cache
uses: actions/cache/restore@v5
with:
path: fate-suite
key: fate-suite
restore-keys: |
fate-suite-
- name: Sync Fate-Suite
id: fate
run: |
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
if: ${{ format('fate-suite-{0}', steps.fate.outputs.hash) != steps.cache.outputs.cache-matched-key }}
with:
path: fate-suite
key: fate-suite-${{ 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']
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 $FF_CONFIGURE \
--cc="$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="$FF_LDEXEFLAGS" \
--enable-memory-poisoning --assert-level=2 --target-exec="${{ matrix.target_exec }}" \
--enable-hardcoded-tables \
|| CFGRES=$?
cat ffbuild/config.log
exit ${CFGRES:-0}
- name: Build
run: make -C build -j$(nproc)
- name: Restore Cached Fate-Suite
id: cache
uses: actions/cache/restore@v5
with:
path: fate-suite
key: fate-suite
restore-keys: |
fate-suite-
- name: Sync Fate-Suite
id: fate
run: |
make -C build 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
if: ${{ format('fate-suite-{0}', steps.fate.outputs.hash) != steps.cache.outputs.cache-matched-key }}
with:
path: fate-suite
key: fate-suite-${{ 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$(nproc) || FATERES=$?
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
-1
View File
@@ -1,2 +1 @@
*.pnm -diff -text
Changelog merge=union
-15
View File
@@ -1,6 +1,5 @@
*.a
*.o
*.objs
*.o.*
*.d
*.def
@@ -23,12 +22,6 @@
*.metal.air
*.metallib
*.metallib.c
*.spv
*.spv.c
*.spv.gz
*.gen.asm
*.gen.c
*.gen.S
*.ptx
*.ptx.c
*.ptx.gz
@@ -40,19 +33,11 @@
/ffmpeg
/ffplay
/ffprobe
/checkasm_config_generated.asm
/checkasm_config_generated.h
/checkasm_header_config_generated.h
/config.asm
/config.h
/config_components.asm
/config_components.h
/coverage.info
/lcov/
/src
/mapfile
/tools/python/__pycache__/
/libavcodec/vulkan/*.c
/libavfilter/vulkan/*.c
/.*/
!/.forgejo/
-2
View File
@@ -26,5 +26,3 @@ rcombs <rcombs@rcombs.me> <rodger.combs@gmail.com>
Cosmin Stejerean <cosmin@cosmin.at> Cosmin Stejerean via ffmpeg-devel <ffmpeg-devel@ffmpeg.org>
<wutong1208@outlook.com> <tong1.wu-at-intel.com@ffmpeg.org>
<wutong1208@outlook.com> <tong1.wu@intel.com>
<toqsxw@outlook.com> <jianhua.wu-at-intel.com@ffmpeg.org>
<toqsxw@outlook.com> <jianhua.wu@intel.com>
+1 -2
View File
@@ -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**.
Patches should be submitted to 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.
+9 -9
View File
@@ -55,7 +55,7 @@ modified by someone else and passed on, the recipients should know
that what they have is not the original version, so that the original
author's reputation will not be affected by problems that might be
introduced by others.
Finally, software patents pose a constant threat to the existence of
any free program. We wish to make sure that a company cannot
effectively restrict the users of a free program by obtaining a
@@ -111,7 +111,7 @@ modification follow. Pay close attention to the difference between a
"work based on the library" and a "work that uses the library". The
former contains code derived from the library, whereas the latter must
be combined with the library in order to run.
GNU LESSER GENERAL PUBLIC LICENSE
TERMS AND CONDITIONS FOR COPYING, DISTRIBUTION AND MODIFICATION
@@ -158,7 +158,7 @@ Library.
You may charge a fee for the physical act of transferring a copy,
and you may at your option offer warranty protection in exchange for a
fee.
2. You may modify your copy or copies of the Library or any portion
of it, thus forming a work based on the Library, and copy and
distribute such modifications or work under the terms of Section 1
@@ -216,7 +216,7 @@ instead of to this License. (If a newer version than version 2 of the
ordinary GNU General Public License has appeared, then you can specify
that version instead if you wish.) Do not make any other change in
these notices.
Once this change is made in a given copy, it is irreversible for
that copy, so the ordinary GNU General Public License applies to all
subsequent copies and derivative works made from that copy.
@@ -267,7 +267,7 @@ Library will still fall under Section 6.)
distribute the object code for the work under the terms of Section 6.
Any executables containing that work also fall under Section 6,
whether or not they are linked directly with the Library itself.
6. As an exception to the Sections above, you may also combine or
link a "work that uses the Library" with the Library to produce a
work containing portions of the Library, and distribute that work
@@ -329,7 +329,7 @@ restrictions of other proprietary libraries that do not normally
accompany the operating system. Such a contradiction means you cannot
use both them and the Library together in an executable that you
distribute.
7. You may place library facilities that are a work based on the
Library side-by-side in a single library together with other library
facilities not covered by this License, and distribute such a combined
@@ -370,7 +370,7 @@ subject to these terms and conditions. You may not impose any further
restrictions on the recipients' exercise of the rights granted herein.
You are not responsible for enforcing compliance by third parties with
this License.
11. If, as a consequence of a court judgment or allegation of patent
infringement or for any other reason (not limited to patent issues),
conditions are imposed on you (whether by court order, agreement or
@@ -422,7 +422,7 @@ conditions either of that version or of any later version published by
the Free Software Foundation. If the Library does not specify a
license version number, you may choose any version ever published by
the Free Software Foundation.
14. If you wish to incorporate parts of the Library into other free
programs whose distribution conditions are incompatible with these,
write to the author to ask for permission. For software which is
@@ -456,7 +456,7 @@ SUCH HOLDER OR OTHER PARTY HAS BEEN ADVISED OF THE POSSIBILITY OF SUCH
DAMAGES.
END OF TERMS AND CONDITIONS
How to Apply These Terms to Your New Libraries
If you develop a new library, and you want it to be of the greatest
-110
View File
@@ -1,101 +1,6 @@
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 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:
- ffprobe -codec option
- EXIF Metadata Parsing
- gfxcapture: Windows.Graphics.Capture based window/monitor capture
- hxvs demuxer for HXVS/HXVT IP camera format
- MPEG-H 3D Audio decoding via mpeghdec
- D3D12 H.264 encoder
- drawvg filter via libcairo
- ffmpeg CLI tiled HEIF support
- D3D12 AV1 encoder
- ProRes Vulkan hwaccel
- DPX Vulkan hwaccel
- Rockchip H.264/HEVC hardware encoder
- Add vf_scale_d3d12 filter
- JPEG-XS parser
- JPEG-XS decoder and encoder through libsvtjpegxs
- JPEG-XS raw bitstream muxer and demuxer
- IAMF Projection mode Ambisonic Audio Elements muxing and demuxing
- Add vf_mestimate_d3d12 filter
- xHE-AAC Mps212 decoding support (experimental)
- Remove the old HLS protocol handler
- Vulkan compute codec optimizations
- swscale Vulkan support
- LCEVC metadata bitstream filter
- Add vf_deinterlace_d3d12 filter
- ffprobe: only show refs field in stream section when reading frames
- ProRes Vulkan encoder
- LCEVC parser
- LCEVC enhancement layer exporting in MPEG-TS
version 8.0:
- Whisper filter
- Drop support for OpenSSL < 1.1.0
- Enable TLS peer certificate verification by default (on next major version bump)
- Drop support for OpenSSL < 1.1.1
- yasm support dropped, users need to use nasm
- VVC VAAPI decoder
- RealVideo 6.0 decoder
- OpenMAX encoders deprecated
- libx265 alpha layer encoding
- ADPCM IMA Xbox decoder
- Enhanced FLV v2: Multitrack audio/video, modern codec support
- Animated JPEG XL encoding (via libjxl)
- VVC in Matroska
- CENC AV1 support in MP4 muxer
- pngenc: set default prediction method to PAETH
- APV decoder and APV raw bitstream muxing and demuxing
- APV parser
- APV encoding support through a libopenapv wrapper
- VVC decoder supports all content of SCC (Screen Content Coding):
IBC (Inter Block Copy), Palette Mode and ACT (Adaptive Color Transform
- G.728 decoder
- pad_cuda filter
- Sanyo LD-ADPCM decoder
- APV in MP4/ISOBMFF muxing and demuxing
- OpenHarmony hardware decoder/encoder
- Colordetect filter
- Add vf_scale_d3d11 filter
- No longer disabling GCC autovectorization, on X86, ARM and AArch64
- VP9 Vulkan hwaccel
- AV1 Vulkan encoder
- ProRes RAW decoder
- ProRes RAW Vulkan hwaccel
- ffprobe -codec option
- HDR10+ metadata passthrough when decoding/encoding with libaom-av1
version 7.1:
- Raw Captions with Time (RCWT) closed caption demuxer
- LC3/LC3plus decoding/encoding using external library liblc3
@@ -122,20 +27,6 @@ version 7.1:
- LCEVC enhancement data exporting in H.26x and MP4/ISOBMFF
- LCEVC filter
- MV-HEVC decoding
- minor stream specifier syntax changes:
- when matching by metadata (:m:<key>:<val>), the colon character
in keys or values now has to be backslash-escaped
- in optional maps (-map ....?) with a metadata-matching stream specifier,
the value has to be separated from the question mark by a colon, i.e.
-map ....:m:<key>:<val>:? (otherwise it would be ambiguous whether the
question mark is a part of <val> or not)
- multiple stream types in a single specifier (e.g. :s:s:0) now cause an
error, as such a specifier makes no sense
- Mastering Display and Content Light Level metadata support in hevc_nvenc
and av1_nvenc encoders
- libswresample now accepts custom order channel layouts as input, with some
constrains
- FFV1 parser
version 7.0:
@@ -224,7 +115,6 @@ version 6.1:
variable-fields elements within the same parent element
- ffprobe -output_format option added as an alias of -of
# codespell:off
version 6.0:
- Radiance HDR image support
-7
View File
@@ -1,7 +0,0 @@
{
"drips": {
"ethereum": {
"ownedBy": "0x2f3900e7064eE63D30d749971265858612AA7139"
}
}
}
+4 -11
View File
@@ -1,7 +1,4 @@
# Installing FFmpeg
0. If you like to include source plugins, merge them before configure
for example run tools/merge-all-source-plugins
## Installing FFmpeg
1. Type `./configure` to create the configuration. A list of configure
options is printed by running `configure --help`.
@@ -14,11 +11,7 @@ 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.
## 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.
- Non system dependencies (e.g. libx264, libvpx) are disabled by default.
+2
View File
@@ -12,6 +12,7 @@ configure to activate them. In this case, FFmpeg's license changes to GPL v2+.
Specifically, the GPL parts of FFmpeg are:
- libpostproc
- optional x86 optimization in the files
- `libavcodec/x86/flac_dsp_gpl.asm`
- `libavcodec/x86/idct_mmx.c`
@@ -44,6 +45,7 @@ Specifically, the GPL parts of FFmpeg are:
- `vf_owdenoise.c`
- `vf_perspective.c`
- `vf_phase.c`
- `vf_pp.c`
- `vf_pp7.c`
- `vf_pullup.c`
- `vf_repeatfields.c`
+20 -42
View File
@@ -45,34 +45,31 @@ Commandline utility code:
QuickTime faststart:
tools/qt-faststart.c Baptiste Coudurier
Execution Graph Printing
fftools/graph, fftools/resources [2] softworkz
Miscellaneous Areas
===================
documentation Stefano Sabatini, Mike Melanson, Timothy Gu, Gyan Doshi
project server day to day operations (L: root@ffmpeg.org) Michael Niedermayer, Reimar Doeffinger, Alexander Strasser, Nikolay Aleksandrov, Timo Rothenpieler
project server emergencies (L: root@ffmpeg.org) Reimar Doeffinger, Alexander Strasser, Nikolay Aleksandrov, Timo Rothenpieler
project server day to day operations (L: root@ffmpeg.org) Árpád Gereöffy, Michael Niedermayer, Reimar Doeffinger, Alexander Strasser, Nikolay Aleksandrov, Timo Rothenpieler
project server emergencies (L: root@ffmpeg.org) Árpád Gereöffy, Reimar Doeffinger, Alexander Strasser, Nikolay Aleksandrov, Timo Rothenpieler
presets [0]
metadata subsystem Aurelien Jacobs
release management Michael Niedermayer
API tests [0]
samples-request [2] Thilo Borgmann, James Almer, Ben Littler
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
=========
@@ -85,15 +82,12 @@ Other:
aes_ctr.c, aes_ctr.h Eran Kornblau
bprint Nicolas George
bswap.h
csp.c, csp.h Leo Izen, Ronald S. Bultje
des Reimar Doeffinger
dynarray.h Nicolas George
eval.c, eval.h [2] Michael Niedermayer
float_dsp Loren Merritt
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
@@ -139,6 +133,8 @@ Generic Parts:
ratecontrol.c [2] Michael Niedermayer
simple IDCT:
simple_idct.c, simple_idct.h [2] Michael Niedermayer
postprocessing:
libpostproc/* [2] Michael Niedermayer
table generation:
tableprint.c, tableprint.h Reimar Doeffinger
fixed point FFT:
@@ -153,13 +149,13 @@ Codecs:
adpcm.c Zane van Iperen
alacenc.c Jaikrishnan Menon
alsdec.c Thilo Borgmann, Umair Khan
amfdec*,amfenc* [2] Dmitrii Ovchinnikov, Araz Iusubov
amfenc* Dmitrii Ovchinnikov
aptx.c Aurelien Jacobs
ass* Aurelien Jacobs
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
@@ -180,7 +176,6 @@ Codecs:
dss_sp.c Oleksij Rempel
dv.c Roman Shaposhnik
dvbsubdec.c Anshul Maheshwari
dxv.*, dxvenc.* Emma Worley
eacmv*, eaidct*, eat* Peter Ross
exif.c, exif.h Thilo Borgmann
ffv1* [2] Michael Niedermayer
@@ -211,8 +206,9 @@ Codecs:
libgsm.c Michel Bardiaux
libkvazaar.c Arttu Ylä-Outinen
libopenh264enc.c Martin Storsjo, Linjie Fu
libopenjpeg.c Jaikrishnan Menon
libopenjpegenc.c Michael Bradshaw
libtheoraenc.c [0]
libtheoraenc.c David Conrad
libvorbis.c David Conrad
libvpx* James Zern
libxavs.c Stefan Gehrer
@@ -233,7 +229,7 @@ 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
@@ -244,14 +240,13 @@ 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
rv10.c Michael Niedermayer
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
@@ -271,7 +266,7 @@ Codecs:
vp8 David Conrad, Ronald Bultje
vp9 Ronald Bultje
vqavideo.c Mike Melanson
vvc [2] Nuo Mi, Wu Jianhua, Frank Plowman
vvc [2] Nuo Mi
wmaprodec.c Sascha Sommer
wmavoice.c Ronald S. Bultje
wmv2.c Michael Niedermayer
@@ -279,10 +274,8 @@ Codecs:
xface Stefano Sabatini
Hardware acceleration:
amf* [2] Dmitrii Ovchinnikov, Araz Iusubov
dxva2* Hendrik Leppkes, Laurent Aimar, Steve Lhomme
d3d11va* Steve Lhomme
d3d12va* Wu Jianhua
d3d12va_encode* Tong Wu
mediacodec* Matthieu Bouron, Aman Gupta, Zhao Zhili
vaapi* Haihao Xiang
@@ -318,10 +311,6 @@ libavfilter
===========
Generic parts:
Framework and orphaned filters Nicolas George
(except hardware acceleration)
graphdump.c Nicolas George
motion_estimation.c Davinder Singh
@@ -353,9 +342,7 @@ 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>)
@@ -404,7 +391,7 @@ Muxers/Demuxers:
dss.c Oleksij Rempel
dtsdec.c foo86
dv.c Roman Shaposhnik
dvdvideodec.c [2] Marth64
dvdvideodec.c Marth64
electronicarts.c Peter Ross
evc* Samsung (Dawid Kozinski)
ffm* Baptiste Coudurier
@@ -413,9 +400,7 @@ 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
@@ -451,17 +436,16 @@ 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
pva.c Ivo van Poorten
r3d.c Baptiste Coudurier
raw.c Michael Niedermayer
rcwtdec.c [2] Marth64
rcwtenc.c [2] Marth64
rcwtenc.c Marth64
rdt.c Ronald S. Bultje
rl2.c Sascha Sommer
rmdec.c, rmenc.c Ronald S. Bultje
@@ -480,7 +464,6 @@ Muxers/Demuxers:
sdp.c Martin Storsjo
segafilm.c Mike Melanson
segment.c Stefano Sabatini
smush.c Manuel Lauss
spdif* Anssi Hannula
srtdec.c Aurelien Jacobs
swf.c Baptiste Coudurier
@@ -492,7 +475,6 @@ Muxers/Demuxers:
webm dash (matroskaenc.c) Vignesh Venkatasubramanian
webvtt* Matthew J Heaney
westwood.c Mike Melanson
whip.c [2] Jack Lau
wtv.c Peter Ross
Protocols:
@@ -529,10 +511,9 @@ Operating systems / CPU architectures
Alpha [0]
MIPS Manojkumar Bhosale, Shiyou Yin
LoongArch [2] Shiyou Yin
Darwin (macOS, iOS) [2] Marvin Scholz
Mac OS X / PowerPC [0]
Mac OS X / PowerPC Romain Dolbeau, Guillaume Poirier
Amiga / PowerPC Colin Ward
Linux / PowerPC [2] Sean McGovern (CC <gseanmcg@gmail.com>), Lauri Kasanen
Linux / PowerPC Lauri Kasanen
RISC-V [2] Rémi Denis-Courmont
Windows MinGW Alex Beregszaszi, Ramiro Polla
Windows Cygwin Victor Paesa
@@ -560,7 +541,6 @@ James Darnley
Jan Ekström
Joakim Plate
Jun Zhao
Kacper Michajłow
Kieran Kunhya
Kirill Gavrilov
Limin Wang
@@ -602,7 +582,6 @@ Benoit Fouet B22A 4F4F 43EF 636B BB66 FCDC 0023 AE1E 2985 49C8
Clément Bœsch 52D0 3A82 D445 F194 DB8B 2B16 87EE 2CB8 F4B8 FCF9
Daniel Verkamp 78A6 07ED 782C 653E C628 B8B9 F0EB 8DD8 2F0E 21C7
FFmpeg release signing key FCF9 86EA 15E6 E293 A564 4F10 B432 2F04 D676 58D8
Frank Plowman 34E2 48D6 B7DF 4769 70C7 3304 03A8 4C6A 098F 2C6B
Ganesh Ajjanagadde C96A 848E 97C3 CEA2 AB72 5CE4 45F9 6A2D 3C36 FB1B
Gwenole Beauchesne 2E63 B3A6 3E44 37E2 017D 2704 53C7 6266 B153 99C4
Haihao Xiang (haihao) 1F0C 31E8 B4FE F7A4 4DC1 DC99 E0F5 76D4 76FC 437F
@@ -628,7 +607,6 @@ Reinhard Tartler 9300 5DC2 7E87 6C37 ED7B CA9A 9808 3544 9453 48A4
Reynaldo H. Verdejo Pinochet 6E27 CD34 170C C78E 4D4F 5F40 C18E 077F 3114 452A
Robert Swain EE7A 56EA 4A81 A7B5 2001 A521 67FA 362D A2FC 3E71
Sascha Sommer 38A0 F88B 868E 9D3A 97D4 D6A0 E823 706F 1E07 0D3C
Sean McGovern (Sean_McG) 6D03 BC60 3A33 E615 6E2E 06AD 8C06 8175 6F59 8684
Stefano Sabatini 0D0B AD6B 5330 BBAD D3D6 6A0C 719C 2839 FC43 2D5F
Steinar H. Gunderson C2E9 004F F028 C18E 4EAD DB83 7F61 7561 7797 8F76
Stephan Hilb 4F38 0B3A 5F39 B99B F505 E562 8D5C 5554 4E17 8863
+18 -30
View File
@@ -11,8 +11,6 @@ vpath %.asm $(SRC_PATH)
vpath %.rc $(SRC_PATH)
vpath %.v $(SRC_PATH)
vpath %.texi $(SRC_PATH)
vpath %.glsl $(SRC_PATH)
vpath %.spv $(SRC_PATH)
vpath %.cu $(SRC_PATH)
vpath %.ptx $(SRC_PATH)
vpath %.metal $(SRC_PATH)
@@ -21,20 +19,14 @@ vpath %/fate_config.sh.template $(SRC_PATH)
TESTTOOLS = audiogen videogen rotozoom tiny_psnr tiny_ssim base64 audiomatch
HOSTPROGS := $(TESTTOOLS:%=tests/%) doc/print_options
ALLFFLIBS = \
avcodec \
avdevice \
avfilter \
avformat \
avutil \
swscale \
swresample \
ALLFFLIBS = avcodec avdevice avfilter avformat avutil postproc swscale swresample
# $(FFLIBS-yes) needs to be in linking order
FFLIBS-$(CONFIG_AVDEVICE) += avdevice
FFLIBS-$(CONFIG_AVFILTER) += avfilter
FFLIBS-$(CONFIG_AVFORMAT) += avformat
FFLIBS-$(CONFIG_AVCODEC) += avcodec
FFLIBS-$(CONFIG_POSTPROC) += postproc
FFLIBS-$(CONFIG_SWRESAMPLE) += swresample
FFLIBS-$(CONFIG_SWSCALE) += swscale
@@ -55,31 +47,31 @@ FF_DEP_LIBS := $(DEP_LIBS)
FF_STATIC_DEP_LIBS := $(STATIC_DEP_LIBS)
$(TOOLS): %$(EXESUF): %.o
$(call LINK,$(LDFLAGS) $(LDEXEFLAGS) $(LD_O) $(filter-out $(FF_DEP_LIBS), $^) $(EXTRALIBS-$(*F)) $(EXTRALIBS) $(ELIBS))
$(LD) $(LDFLAGS) $(LDEXEFLAGS) $(LD_O) $(filter-out $(FF_DEP_LIBS), $^) $(EXTRALIBS-$(*F)) $(EXTRALIBS) $(ELIBS)
target_dec_%_fuzzer$(EXESUF): target_dec_%_fuzzer.o $(FF_DEP_LIBS)
$(call LINK,$(LDFLAGS) $(LDEXEFLAGS) $(LD_O) $^ $(ELIBS) $(FF_EXTRALIBS) $(LIBFUZZER_PATH))
$(LD) $(LDFLAGS) $(LDEXEFLAGS) $(LD_O) $^ $(ELIBS) $(FF_EXTRALIBS) $(LIBFUZZER_PATH)
target_enc_%_fuzzer$(EXESUF): target_enc_%_fuzzer.o $(FF_DEP_LIBS)
$(call LINK,$(LDFLAGS) $(LDEXEFLAGS) $(LD_O) $^ $(ELIBS) $(FF_EXTRALIBS) $(LIBFUZZER_PATH))
$(LD) $(LDFLAGS) $(LDEXEFLAGS) $(LD_O) $^ $(ELIBS) $(FF_EXTRALIBS) $(LIBFUZZER_PATH)
tools/target_bsf_%_fuzzer$(EXESUF): tools/target_bsf_%_fuzzer.o $(FF_DEP_LIBS)
$(call LINK,$(LDFLAGS) $(LDEXEFLAGS) $(LD_O) $^ $(ELIBS) $(FF_EXTRALIBS) $(LIBFUZZER_PATH))
$(LD) $(LDFLAGS) $(LDEXEFLAGS) $(LD_O) $^ $(ELIBS) $(FF_EXTRALIBS) $(LIBFUZZER_PATH)
target_dem_%_fuzzer$(EXESUF): target_dem_%_fuzzer.o $(FF_DEP_LIBS)
$(call LINK,$(LDFLAGS) $(LDEXEFLAGS) $(LD_O) $^ $(ELIBS) $(FF_EXTRALIBS) $(LIBFUZZER_PATH))
$(LD) $(LDFLAGS) $(LDEXEFLAGS) $(LD_O) $^ $(ELIBS) $(FF_EXTRALIBS) $(LIBFUZZER_PATH)
tools/target_dem_fuzzer$(EXESUF): tools/target_dem_fuzzer.o $(FF_DEP_LIBS)
$(call LINK,$(LDFLAGS) $(LDEXEFLAGS) $(LD_O) $^ $(ELIBS) $(FF_EXTRALIBS) $(LIBFUZZER_PATH))
$(LD) $(LDFLAGS) $(LDEXEFLAGS) $(LD_O) $^ $(ELIBS) $(FF_EXTRALIBS) $(LIBFUZZER_PATH)
tools/target_io_dem_fuzzer$(EXESUF): tools/target_io_dem_fuzzer.o $(FF_DEP_LIBS)
$(call LINK,$(LDFLAGS) $(LDEXEFLAGS) $(LD_O) $^ $(ELIBS) $(FF_EXTRALIBS) $(LIBFUZZER_PATH))
$(LD) $(LDFLAGS) $(LDEXEFLAGS) $(LD_O) $^ $(ELIBS) $(FF_EXTRALIBS) $(LIBFUZZER_PATH)
tools/target_sws_fuzzer$(EXESUF): tools/target_sws_fuzzer.o $(FF_DEP_LIBS)
$(call LINK,$(LDFLAGS) $(LDEXEFLAGS) $(LD_O) $^ $(ELIBS) $(FF_EXTRALIBS) $(LIBFUZZER_PATH))
$(LD) $(LDFLAGS) $(LDEXEFLAGS) $(LD_O) $^ $(ELIBS) $(FF_EXTRALIBS) $(LIBFUZZER_PATH)
tools/target_swr_fuzzer$(EXESUF): tools/target_swr_fuzzer.o $(FF_DEP_LIBS)
$(call LINK,$(LDFLAGS) $(LDEXEFLAGS) $(LD_O) $^ $(ELIBS) $(FF_EXTRALIBS) $(LIBFUZZER_PATH))
$(LD) $(LDFLAGS) $(LDEXEFLAGS) $(LD_O) $^ $(ELIBS) $(FF_EXTRALIBS) $(LIBFUZZER_PATH)
tools/enum_options$(EXESUF): ELIBS = $(FF_EXTRALIBS)
tools/enum_options$(EXESUF): $(FF_DEP_LIBS)
@@ -106,14 +98,13 @@ 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 \
ALTIVEC-OBJS VSX-OBJS MMX-OBJS X86ASM-OBJS \
MIPSFPU-OBJS MIPSDSPR2-OBJS MIPSDSP-OBJS MSA-OBJS \
MMI-OBJS LSX-OBJS LASX-OBJS RV-OBJS RVV-OBJS RVVB-OBJS \
OBJS SHLIBOBJS STLIBOBJS HOSTOBJS TESTOBJS SIMD128-OBJS \
SVE-OBJS SVE2-OBJS SME-OBJS SME2-OBJS
OBJS SLIBOBJS SHLIBOBJS STLIBOBJS HOSTOBJS TESTOBJS
define RESET
$(1) :=
@@ -143,13 +134,11 @@ ifeq ($(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))
$(LD) $(LDFLAGS) $(LDEXEFLAGS) $(LD_O) $(OBJS-$*) $(FF_EXTRALIBS)
VERSION_SH = $(SRC_PATH)/ffbuild/version.sh
ifeq ($(VERSION_TRACKING),yes)
@@ -189,8 +178,7 @@ 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.h 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 +193,8 @@ 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
+2 -1
View File
@@ -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.
+1 -1
View File
@@ -1 +1 @@
8.0.git
7.1
+15
View File
@@ -0,0 +1,15 @@
┌──────────────────────────────────────┐
│ RELEASE NOTES for FFmpeg 7.1 "Péter" │
└──────────────────────────────────────┘
The FFmpeg Project proudly presents FFmpeg 7.1 "Péter", about 6
months after the release of FFmpeg 7.0.
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.
-114
View File
@@ -1,114 +0,0 @@
/*
* Android Binder handler
*
* Copyright (c) 2025 Dmitrii Okunev
*
* 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
*/
#if defined(__ANDROID__)
#include <dlfcn.h>
#include <stdint.h>
#include <stdlib.h>
#include "libavutil/log.h"
#include "binder.h"
#define THREAD_POOL_SIZE 1
static void *dlopen_libbinder_ndk(void)
{
/*
* libbinder_ndk.so often does not contain the functions we need, so making
* this dependency optional, thus using dlopen/dlsym instead of linking.
*
* See also: https://source.android.com/docs/core/architecture/aidl/aidl-backends
*/
void *h = dlopen("libbinder_ndk.so", RTLD_NOW | RTLD_LOCAL);
if (h != NULL)
return h;
av_log(NULL, AV_LOG_WARNING,
"android/binder: unable to load libbinder_ndk.so: '%s'; skipping binder threadpool init (MediaCodec likely won't work)\n",
dlerror());
return NULL;
}
static void android_binder_threadpool_init(void)
{
typedef int (*set_thread_pool_max_fn)(uint32_t);
typedef void (*start_thread_pool_fn)(void);
set_thread_pool_max_fn set_thread_pool_max = NULL;
start_thread_pool_fn start_thread_pool = NULL;
void *h = dlopen_libbinder_ndk();
if (h == NULL)
return;
unsigned thead_pool_size = THREAD_POOL_SIZE;
set_thread_pool_max =
(set_thread_pool_max_fn) dlsym(h,
"ABinderProcess_setThreadPoolMaxThreadCount");
start_thread_pool =
(start_thread_pool_fn) dlsym(h, "ABinderProcess_startThreadPool");
if (start_thread_pool == NULL) {
av_log(NULL, AV_LOG_WARNING,
"android/binder: ABinderProcess_startThreadPool not found; skipping threadpool init (MediaCodec likely won't work)\n");
return;
}
if (set_thread_pool_max != NULL) {
int ok = set_thread_pool_max(thead_pool_size);
av_log(NULL, AV_LOG_DEBUG,
"android/binder: ABinderProcess_setThreadPoolMaxThreadCount(%u) => %s\n",
thead_pool_size, ok ? "ok" : "fail");
} else {
av_log(NULL, AV_LOG_DEBUG,
"android/binder: ABinderProcess_setThreadPoolMaxThreadCount is unavailable; using the library default\n");
}
start_thread_pool();
av_log(NULL, AV_LOG_DEBUG,
"android/binder: ABinderProcess_startThreadPool() called\n");
}
void android_binder_threadpool_init_if_required(void)
{
#if __ANDROID_API__ >= 24
if (android_get_device_api_level() < 35) {
// the issue with the thread pool was introduced in Android 15 (API 35)
av_log(NULL, AV_LOG_DEBUG,
"android/binder: API<35, thus no need to initialize a thread pool\n");
return;
}
android_binder_threadpool_init();
#else
// android_get_device_api_level was introduced in API 24, so we cannot use it
// to detect the API level in API<24. For simplicity we just assume
// libbinder_ndk.so on the system running this code would have API level < 35;
av_log(NULL, AV_LOG_DEBUG,
"android/binder: is built with API<24, assuming this is not Android 15+\n");
#endif
}
#endif /* __ANDROID__ */
-4
View File
@@ -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); }
@@ -190,7 +189,4 @@ static inline __device__ float __cosf(float a) { return __nvvm_cos_approx_f(a);
static inline __device__ float __expf(float a) { return __nvvm_ex2_approx_f(a * (float)__builtin_log2(__builtin_exp(1))); }
static inline __device__ float __powf(float a, float b) { return __nvvm_ex2_approx_f(__nvvm_lg2_approx_f(a) * b); }
// Misc helper functions
extern "C" __device__ int printf(const char*, ...);
#endif /* COMPAT_CUDA_CUDA_RUNTIME_H */
+1 -1
View File
@@ -38,7 +38,7 @@ static int optind = 1;
static int optopt;
static char *optarg;
static int getopt(int argc, char *argv[], const char *opts)
static int getopt(int argc, char *argv[], char *opts)
{
static int sp = 1;
int c;
+1 -1
View File
@@ -218,7 +218,7 @@ while (<F>) {
# Lines of the form '} SOME_VERSION_NAME_1.0;'
if (/^[ \t]*\}[ \tA-Z0-9_.a-z]+;[ \t]*$/) {
$glob = 'glob';
# We tried to match symbols against this version, but none matched.
# We tried to match symbols agains this version, but none matched.
# Emit dummy hidden symbol to avoid marking this version WEAK.
if ($matches_attempted && $matched_symbols == 0) {
print " hidden:\n";
+17 -67
View File
@@ -44,14 +44,13 @@
#include "libavutil/internal.h"
#include "libavutil/mem.h"
#include "libavutil/time.h"
#include "libavutil/wchar_filename.h"
typedef struct w32pthread_t {
typedef struct pthread_t {
void *handle;
void *(*func)(void* arg);
void *arg;
void *ret;
} *pthread_t;
} pthread_t;
/* use light weight mutex/condition variable API for Windows Vista and later */
typedef SRWLOCK pthread_mutex_t;
@@ -72,46 +71,32 @@ typedef CONDITION_VARIABLE pthread_cond_t;
#define THREADFUNC_RETTYPE unsigned
#endif
av_unused static THREADFUNC_RETTYPE
static av_unused THREADFUNC_RETTYPE
__stdcall attribute_align_arg win32thread_worker(void *arg)
{
pthread_t h = (pthread_t)arg;
pthread_t *h = (pthread_t*)arg;
h->ret = h->func(h->arg);
return 0;
}
av_unused static int pthread_create(pthread_t *thread, const void *unused_attr,
static av_unused int pthread_create(pthread_t *thread, const void *unused_attr,
void *(*start_routine)(void*), void *arg)
{
pthread_t ret;
ret = (pthread_t)av_mallocz(sizeof(*ret));
if (!ret)
return EAGAIN;
ret->func = start_routine;
ret->arg = arg;
thread->func = start_routine;
thread->arg = arg;
#if HAVE_WINRT
ret->handle = (void*)CreateThread(NULL, 0, win32thread_worker, ret,
0, NULL);
thread->handle = (void*)CreateThread(NULL, 0, win32thread_worker, thread,
0, NULL);
#else
ret->handle = (void*)_beginthreadex(NULL, 0, win32thread_worker, ret,
0, NULL);
thread->handle = (void*)_beginthreadex(NULL, 0, win32thread_worker, thread,
0, NULL);
#endif
if (!ret->handle) {
av_free(ret);
return EAGAIN;
}
*thread = ret;
return 0;
return !thread->handle;
}
av_unused static int pthread_join(pthread_t thread, void **value_ptr)
static av_unused int pthread_join(pthread_t thread, void **value_ptr)
{
DWORD ret = WaitForSingleObject(thread->handle, INFINITE);
DWORD ret = WaitForSingleObject(thread.handle, INFINITE);
if (ret != WAIT_OBJECT_0) {
if (ret == WAIT_ABANDONED)
return EINVAL;
@@ -119,9 +104,8 @@ av_unused static int pthread_join(pthread_t thread, void **value_ptr)
return EDEADLK;
}
if (value_ptr)
*value_ptr = thread->ret;
CloseHandle(thread->handle);
av_free(thread);
*value_ptr = thread.ret;
CloseHandle(thread.handle);
return 0;
}
@@ -149,7 +133,7 @@ static inline int pthread_mutex_unlock(pthread_mutex_t *m)
typedef INIT_ONCE pthread_once_t;
#define PTHREAD_ONCE_INIT INIT_ONCE_STATIC_INIT
av_unused static int pthread_once(pthread_once_t *once_control, void (*init_routine)(void))
static av_unused int pthread_once(pthread_once_t *once_control, void (*init_routine)(void))
{
BOOL pending = FALSE;
InitOnceBeginInitialize(once_control, 0, &pending, NULL);
@@ -210,38 +194,4 @@ static inline int pthread_setcancelstate(int state, int *oldstate)
return 0;
}
static inline int win32_thread_setname(const char *name)
{
#if !HAVE_UWP
typedef HRESULT (WINAPI *SetThreadDescriptionFn)(HANDLE, PCWSTR);
// Although SetThreadDescription lives in kernel32.dll, on Windows Server 2016,
// Windows 10 LTSB 2016 and Windows 10 version 1607, it was only available in
// kernelbase.dll. So, load it from there for maximum coverage.
HMODULE kernelbase = GetModuleHandleW(L"kernelbase.dll");
if (!kernelbase)
return AVERROR(ENOSYS);
SetThreadDescriptionFn pSetThreadDescription =
(SetThreadDescriptionFn)GetProcAddress(kernelbase, "SetThreadDescription");
if (!pSetThreadDescription)
return AVERROR(ENOSYS);
wchar_t *wname;
if (utf8towchar(name, &wname) < 0)
return AVERROR(ENOMEM);
HRESULT hr = pSetThreadDescription(GetCurrentThread(), wname);
av_free(wname);
return SUCCEEDED(hr) ? 0 : AVERROR(EINVAL);
#else
// UWP is not supported because we cannot use LoadLibrary/GetProcAddress to
// detect the availability of the SetThreadDescription API. There is a small
// gap in Windows builds 1507-1607 where it was not available. UWP allows
// querying the availability of APIs with QueryOptionalDelayLoadedAPI, but it
// requires /DELAYLOAD:kernel32.dll during linking, and we cannot enforce that.
return AVERROR(ENOSYS);
#endif
}
#endif /* COMPAT_W32PTHREADS_H */
+5 -25
View File
@@ -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
@@ -53,13 +48,7 @@ trap 'rm -f -- $libname' EXIT
if [ -n "$AR" ]; then
$AR rcs ${libname} $@ >/dev/null
else
machine_flag=""
case "$LDFLAGS" in
*"machine:arm64ec"*)
machine_flag="-machine:arm64ec"
;;
esac
lib.exe ${machine_flag} -out:${libname} $@ >/dev/null
lib.exe -out:${libname} $@ >/dev/null
fi
if [ $? != 0 ]; then
echo "Could not create temporary library." >&2
@@ -117,21 +106,12 @@ if [ -n "$NM" ]; then
grep -v : |
grep -v ^$ |
cut -d' ' -f3 |
sed -e "s/^${prefix}//" -e "s/^#//" |
grep -v '\$entry_thunk' |
grep -v '\$exit_thunk')
sed -e "s/^${prefix}//")
else
member=1
case "$LDFLAGS" in
*"machine:arm64ec"*)
member=32
;;
esac
dump=$(dumpbin.exe -linkermember:${member} ${libname} |
sed -e '/public symbols/,$!d' -e '/^ \{1,\}Summary/,$d' -e 's/^[[:space:]]*[0-9A-Fa-f]\{1,\}[[:space:]]\{1,\}//' -e "s/^${prefix}//" -e 's/^#//' |
dump=$(dumpbin.exe -linkermember:1 ${libname} |
sed -e '/public symbols/,$!d' -e '/^ \{1,\}Summary/,$d' -e "s/ \{1,\}${prefix}/ /" -e 's/ \{1,\}/ /g' |
tail -n +2 |
grep -v '\$entry_thunk' |
grep -v '\$exit_thunk')
cut -d' ' -f3)
fi
rm ${libname}
Vendored
+414 -1096
View File
File diff suppressed because it is too large Load Diff
+3 -424
View File
@@ -1,428 +1,7 @@
The last version increases of all libraries were on 2026-06-23.
The last version increases of all libraries were on 2024-03-07
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
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.
2026-03-07 - f37cfb370f8 - lavc 62.25.100 - avcodec.h
Deprecate AVCodecContext.intra_dc_precision.
2026-02-26 - 26d6ac56c10 - lsws 9.4.100 - swscale.h
Add sws_test_hw_format().
2026-02-19 - ca011ee7547 - lavf 62.10.100 - avformat.h
Add demuxer command API consisting of:
- Add avformat_send_command().
- Add avformat_receive_command_reply().
- Add enum AVFormatCommandID.
2026-02-13 - 66766bb1c76 - lavu 60.25.100 - avassert.h
Deprecate av_assert0_fpu() and av_assert2_fpu() without replacement.
2026-02-09 - f2b13f783ac - lavf 62.9.100 - avformat.h
Add AVFormatContext.name.
2026-01-26 - 714152dd9a0 - lavu 60.24.100 - hwcontext_d3d11va.h
Add BindFlags and MiscFlags to AVD3D11VADeviceContext
2026-01-04 - 0645c484537 - lavu 60.23.100 - cpu.h
Add AV_CPU_FLAG_ARM_CRC.
2026-01-04 - e382772e4a1 - lavu 60.22.100 - cpu.h
Add AV_CPU_FLAG_CLMUL.
2025-12-29 - a6e5fa3fbb5 - lavu 60.21.100 - iamf.h
Add nb_demixing_matrix to AVIAMFLayer
2025-12-13 - b8ee68cffe - lavc 62.23.100 - codec_id.h
Add AV_CODEC_ID_JPEGXS.
2025-12-12 - c9b5e145b71 - lavf 62.7.100 - avformat.h
Add av_mime_codec_str
2025-12-08 - 212eb8413a6 - lavu 60.20.100 - hwcontext_vulkan.h
Add av_vk_get_optional_instance_extensions().
Add av_vk_get_optional_device_extensions().
2025-12-07 - 00caeba050d - lavc 62.22.101 - avcodec.h
Add avcodec_receive_frame_flags().
Add AV_CODEC_RECEIVE_FRAME_FLAG_SYNCHRONOUS.
2025-11-26 - 623669a02cd - lavfi 11.10.100 - buffersrc.h
Add av_buffersrc_get_status().
2025-11-18 - 140b4f28c30 - lavu 60.19.100 - hwcontext_amf.h
avutil/hwcontext_amf: add lock and unlock for AVAMFDeviceContext.
2025-11-16 - 643e2e10f98 - lavu 60.18.100 - cpu.h
Deprecate AV_CPU_FLAG_FORCE without replacement.
2025-11-01 - 8a322c956f3 - lavc 62.19.100 - avcodec.h
Schedule AVCodecParser and av_parser_init() to use enum AVCodecID
for codec ids on the next major version bump.
2025-11-01 - e0b0ca81112 - lavc 62.18.100 - avcodec.h
Deprecate AVCodecParser fields priv_data_size, parser_init,
parser_parse, parser_close, and split with no replacement.
2025-10-30 - 6879c8ee5dd - lavc 62.17.100 - packet.h
Add av_packet_side_data_from_frame() and av_packet_side_data_to_frame().
2025-10-28 - 71c7b9156bd - lavu 60.16.100 - pixfmt.h
Add AVCOL_TRC_EXT_BASE, AVCOL_TRC_V_LOG,
AVCOL_PRI_EXT_BASE and AVCOL_PRI_V_GAMUT.
2025-10-24 - 00e43619cd3 - lavu 60.14.100 - csp.h
Add av_csp_approximate_eotf_gamma().
2025-08-xx - xxxxxxxxxx - lavf 62.6.100 - oggparsevorbis.h oggparseopus.h oggparseflac.h
Drop header packets from secondary chained ogg/{flac, opus, vorbis} streams
from demuxer output.
2025-09-08 - 1b97966199f - lavu 60.13.100 - hwcontext_d3d12va.h
Add resource_flags and heap_flags to AVD3D12VADeviceContext
Add heap_flags to AVD3D12VAFramesContext
2025-09-05 - 9458a6f8ec2 - lavf 62.5.100 - avformat.h
Add AV_FRAME_FILENAME_FLAGS_IGNORE_TRUNCATION
2025-09-03 - c2ce3873859 - lavu 60.12.100 - hwcontext_d3d12va.h
Add support for texture array mode AVD3D12VAFrame.subresource_index,
AVD3D12VAFramesContext.texture_array
Add enum AVD3D12VAFrameFlags to define the behaviours of frame allocation.
Renanme AVD3D12VAFramesContext.flags to AVD3D12VAFramesContext.resource_flags.
Add flags to AVD3D12VAFramesContext
Add flags to AVD3D12VAFrame
2025-09-02 - de297ae2fca - lavfi 11.8.100 - buffersrc.h
Add AVBufferSrcParameters.alpha_mode.
2025-09-02 - 0fbd90d78f8 - lavfi 11.7.100 - buffersink.h
Add av_buffersink_get_alpha_mode().
2025-09-02 - ecebf9c6935 - lavc 62.15.100 - avcodec.h codec_par.h
Add AVCodecContext.alpha_mode, AVCodecParameters.alpha_mode, and
AV_CODEC_CONFIG_ALPHA_MODE.
2025-09-02 - eea99a77eda - lavfi 11.6.100 - avfilter.h
Add AVFilterLink.alpha_mode.
2025-09-02 - 22929bd044c - lavu 60.11.100 - frame.h pixfmt.h
Add AVAlphaMode, AVFrame.alpha_mode, av_alpha_mode_name() and
av_alpha_mode_from_name().
2025-09-01 - ce0938da8cd - lsws 9.3.100 - swscale.h
Add SWS_UNSTABLE flag.
2025-09-01 - 4ffd621523e - lavc 62.14.100 - packet.h
Add AV_PKT_DATA_EXIF
2025-08-24 - 368b5e0ffc2 - lavc 62.13.101 - exif.h
Add AV_EXIF_FLAG_RECURSIVE
2025-08-19 - ad77345a5d1..fe496b0308f - lavc 62.13.100 - exif.h
Add:
- enum AVTiffDataType, enum AVExifHeaderMode
- struct AVExifMetadata, struct AVExifEntry
- av_exif_get_tag_name, av_exif_get_tag_id,
av_exif_set_entry, av_exif_get_entry,
av_exif_remove_entry, av_exif_parse_buffer,
av_exif_write, av_exif_free,
av_exif_ifd_to_dict, av_exif_clone_ifd,
av_exif_matrix_to_orientation,
av_exif_orientation_to_matrix,
2025-08-19 - bb90b262d6d - lavu 60.10.100 - frame.h
Add AV_FRAME_DATA_EXIF.
-------- 8< --------- FFmpeg 8.0 was cut here -------- 8< ---------
2025-07-29 - 1c85a3832af - lavc 62.10.100 - smpte_436m.h
Add a new public header smpte_436m.h with API for
manipulating AV_CODEC_ID_SMPTE_436M_ANC data.
2025-07-10 - a566fcb9dc0 - lavf 62.2.100
mxf [de]muxer now uses AV_CODEC_ID_SMPTE_436M_ANC for
the vbi_vanc_smpte_436M streams instead of AV_CODEC_ID_NONE.
2025-07-10 - f4ff379baea - lavc 62.10.100 - codec_id.h
Add AV_CODEC_ID_SMPTE_436M_ANC.
2025-08-08 - 83b36f54108 - lavc 62.9.100 - codec_id.h
Add AV_CODEC_ID_PRORES_RAW.
2025-07-31 - 119d127d05c - lavu 60.7.100 - spherical.h
Add AV_SPHERICAL_PARAMETRIC_IMMERSIVE.
2025-07-20 - 157d3b007e9 - lavu 60.6.100 - attributes.h, avstring.h
Add av_scanf_format() and use it on av_sscanf().
2025-07-18 - fbda5ffb953 - lavu 60.5.100 - pixfmt.h
Add AV_PIX_FMT_OHCODEC.
2025-07-18 - fbda5ffb953 - lavu 60.5.100 - hwcontext.h
Add AV_HWDEVICE_TYPE_OHCODEC and AVOHCodecDeviceContext.
2025-07-14 - b24155cae11 - lavfi 11.2.100 - avfilter.h
Add AVFilterGraph->max_buffered_frames.
2025-07-07 - eca477da52 - lavc 62.6.100 - packet.h
Add AV_PKT_DATA_RTCP_SR.
2025-07-01 - 39d5a998bd - lavc 62.4.101 - packet.h
Add AV_PKT_DATA_3D_REFERENCE_DISPLAYS.
2025-07-01 - b2e4b0e282 - lavu 60.4.101 - frame.h
Add AV_FRAME_DATA_3D_REFERENCE_DISPLAYS.
2025-07-01 - 80a05bea4f - lavu 60.4.100 - tdrdi.h
Add AV3DReferenceDisplaysInfo and AV3DReferenceDisplay structs.
Add av_tdrdi_alloc() and av_tdrdi_get_display().
2025-05-21 - 004cc60f0e3 - lavu 60.3.100 - avassert.h
Add av_unreachable() and av_assume() macros.
2025-02-15 - e2f39671ae2 - lavfi 10.10.100 - avfilter.h
Add avfilter_link_get_hw_frames_ctx().
2025-04-21 - bf1579c904a - lavu 60.2.100 - log.h
Add AV_CLASS_CATEGORY_HWDEVICE.
2025-04-16 - c818c67991 - libpostproc 59.1.100 - postprocess.h
Deprecate PP_CPU_CAPS_3DNOW.
2025-04-07 - 19e9a203b7 - lavu 60.01.100 - dict.h
Add AV_DICT_DEDUP.
2025-03-17 - 49af9746e8f - lavu 59.60.100 - pixfmt.h
Add AV_PIX_FMT_GBRAP32BE and AV_PIX_FMT_GBRAP32LE.
2025-03-10 - 61fc9b6fee1 - lavu 59.59.100 - pixfmt.h
Add AV_PIX_FMT_YAF16BE, AV_PIX_FMT_YAF16LE, AV_PIX_FMT_YAF32BE,
and AV_PIX_FMT_YAF32LE.
2025-03-01 - 0245e9382c7 - lavu 59.58.100 - pixfmt.h
Add AV_PIX_FMT_GRAY32BE and AV_PIX_FMT_GRAY32LE.
2025-02-04 - 0ef678f5c50 - lavu 59.56.000 - pixfmt.h
Add AV_PIX_FMT_AMF_SURFACE.
2025-01-09 - a73760da537 - lavu 59.55.100 - pixfmt.h
Add AV_PIX_FMT_GBRPF16BE, AV_PIX_FMT_GBRPF16LE, AV_PIX_FMT_GBRAPF16BE,
AV_PIX_FMT_GBRAPF16LE, AV_PIX_FMT_GRAYF16BE, and AV_PIX_FMT_GRAYF16LE.
2025-02-16 - c79cdae3777 - lavu 59.57.100 - log.h
Add flags AV_LOG_PRINT_TIME and AV_LOG_PRINT_DATETIME.
2025-02-09 - 9fb806fa577 - lavc 61.32.100 - codec_id.h
Add AV_CODEC_ID_IVTV_VBI.
2025-01-25 - ea3c3b42dff - lavu 59.56.100 - frame.h
Add AV_SIDE_DATA_PROP_CHANNEL_DEPENDENT.
2025-01-25 - 6707d970c04 - lavfi 10.9.100 - buffersink.h
Add av_buffersink_get_side_data().
2025-01-25 - 7a025e1cb5f - lavfi 10.8.100 - buffersrc.h
Add AVBufferSrcParameters.side_data and AVBufferSrcParameters.nb_side_data
2025-01-25 - ef1cb1c9c81 - lavfi 10.7.100 - avfilter.h
Add AVFilterLink.side_data and AVFilterLink.nb_side_data
2025-01-05 - 42e72d5c8b5 - lavu 59.55.100 - frame.h
Add AV_FRAME_SIDE_DATA_FLAG_NEW_REF.
2025-01-05 - 19c95ecbff8 - lavc 61.31.100 - avcodec.h
Deprecate AVCodecContext->properties.
2025-01-05 - 2d91f89445d - lavc 61.30.100 - frame.h
Add AV_FRAME_FLAG_LOSSLESS.
2025-01-03 - f3c40826455 - lavc 61.29.100 - codec_id.h
Add AV_CODEC_ID_JPEGXL_ANIM.
2025-01-03 - da9dcaba69d - lavu 59.54.100 - frame.h
Add AV_CH_LAYOUT_5POINT1POINT2 and AV_CHANNEL_LAYOUT_5POINT1POINT2.
2024-12-23 - b88944a8aa5 - lavu 59.53.100 - frame.h
Add av_frame_side_data_remove_by_props().
2024-12-23 - 3428a8d8303 - lavu 59.52.100 - frame.h
Add AV_SIDE_DATA_PROP_SIZE_DEPENDENT and AV_FRAME_DATA_PROP_COLOR_DEPENDENT.
2024-12-23 - 45f0a7ad338 - lsws 8.13.100 - swscale.h
Add enum SwsIntent and SwsContext.intent.
2024-12-15 - 2ac34d08542 - lavc 61.27.100 packet.h
Add av_container_fifo_alloc_avpacket().
2024-12-15 - 56ba57b6725 - lavu 59.51.100 - refstruct.h container_fifo.h
Add a new public header refstruct.h with new API for
reference-counted objects.
Add a new public header container_fifo.h with new API for
a FIFO of container objects (e.g. AVFrame or AVPacket).
2024-12-13 - 6eb4bf04e92 - lavu 59.50.100 - channel_layout.h
Add AV_CH_LAYOUT_9POINT1POINT6 and AV_CHANNEL_LAYOUT_9POINT1POINT6.
2024-12-05 - 06f084468e0 - lavu 59.49.100 - csp.h
Add av_csp_itu_eotf() and av_csp_itu_eotf_inv().
2024-12-05 - bf0a6c41111 - lavu 59.48.100 - csp.h
Add av_csp_trc_func_inv_from_id().
2024-11-25 - 2a091d4f2ee - lsws 8.12.100 - swscale.h
Allow using sws_frame_scale() dynamically, without first initializing the
SwsContext. Deprecate sws_init_context(). Add sws_frame_setup() instead.
2024-11-25 - fb169640092 - lsws 8.11.100 - swscale.h
Replace #define-based SWS_* flags by enum SwsFlags.
2024-11-25 - ed5dd675624 - lsws 8.10.100 - swscale.h
Publicly expose struct SwsContext, enum SwsDither, and enum SwsAlphaBlend.
2024-11-16 - 46cb7b8d9dc - lavu 59.47.101 - frame.h
av_frame_get_buffer() now also aligns the data pointers according to
the requested alignment.
2024-11-13 - 20af68b63a4 - lavu 59.47.100 - channel_layout.h
Add AV_CHAN_BINAURAL_LEFT, AV_CHAN_BINAURAL_RIGHT
Add AV_CH_BINAURAL_LEFT, AV_CH_BINAURAL_RIGHT
Add AV_CH_LAYOUT_BINAURAL, AV_CHANNEL_LAYOUT_BINAURAL
2024-10-26 - e02a3b40a5e - lavu 59.46.100 - pixfmt.h
Add AV_PIX_FMT_XV48.
2024-10-23 - b03c758600f - lsws 8.9.100 - swscale.h
Add sws_is_noop().
2024-10-23 - 5e50a56b9c4 - lsws 8.8.100 - swscale.h
Add frame property testing API:
- sws_test_format()
- sws_test_colorspace()
- sws_test_primaries()
- sws_test_transfer()
- sws_test_frame()
2024-10-23 - 87baf9ab2c2 - lsws 8.7.100 - swscale.h
Add sws_free_context().
2024-10-23 - f462ba05f54 - lavu 59.45.100 - pixfmt.h
Add AV_PIX_FMT_Y216.
2024-10-15 - 2336e685657 - lavu 59.44.100 - pixfmt.h
Add AV_PIX_FMT_RGB96 and AV_PIX_FMT_RGBA128.
2024-10-14 - c993a91bea - lavu 59.43.100 - pixfmt.h
Add AV_PIX_FMT_RGBF16.
2024-10-08 - 29ea34728f1 - lavu 59.42.100 - pixfmt.h
Add AV_PIX_FMT_AYUV, AV_PIX_FMT_UYVA, AV_PIX_FMT_VYU444,
and AV_PIX_FMT_V30X.
2024-10-01 - 0548ab2e425 - lavu 59.41.100 - log.h
Add AVClass.state_flags_offset and AV_CLASS_STATE_INITIALIZED.
2024-09-30 - 50d1b89fa0d - lavf 61.9.100 - avformat.h
Add {nb_}coded_side_data to AVStreamGroupTileGrid.
2024-09-30 - df9b80d21a2 - lavu 59
Deprecate av_int_list_length_for_size(), av_int_list_length(), and
av_opt_set_int_list() without replacement. All AVOptions using these
should be replaced with AV_OPT_TYPE_FLAG_ARRAY.
2024-09-30 - 1efcdbc54d9 - lavfi 10.6.100
Buffersink now has array-type options
- pixel_formats
- colorspaces
- colorranges
replacing the int-list options
- pix_fmts
- color_spaces
- color_ranges
abuffersink now has array-type options
- sample_formats
- samplerates
- channel_layouts
replacing the int-list/string options
- sample_fmts
- sample_rates
- ch_layouts
-------- 8< --------- FFmpeg 7.1 was cut here -------- 8< ---------
2024-09-23 - 6940a6de2f0 - lavu 59.38.100 - frame.h
Add AV_FRAME_DATA_VIEW_ID.
@@ -833,7 +412,7 @@ Add IAMF frame side data types to enum AVFrameSideDataType:
Deprecate AVFrame.palette_has_changed without replacement.
2023-05-15 - 7d1d61cc5f5 - lavc 60 - avcodec.h
Deprecate AVCodecContext.ticks_per_frame in favor of
Depreate AVCodecContext.ticks_per_frame in favor of
AVCodecContext.framerate (encoding) and
AV_CODEC_PROP_FIELDS (decoding).
@@ -841,7 +420,7 @@ Add IAMF frame side data types to enum AVFrameSideDataType:
Add AV_CODEC_PROP_FIELDS.
2023-05-15 - 8b20d0dcb5c - lavc 60 - codec.h
Deprecate AV_CODEC_CAP_SUBFRAMES without replacement.
Depreate AV_CODEC_CAP_SUBFRAMES without replacement.
2023-05-07 - c2ae8e30b7f - lavc 60.11.100 - codec_par.h
Add AVCodecParameters.framerate.
-40
View File
@@ -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
+4 -4
View File
@@ -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 = 7.1
# 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
@@ -295,7 +295,7 @@ MARKDOWN_SUPPORT = YES
# When enabled doxygen tries to link words that correspond to documented
# classes, or namespaces to their corresponding documentation. Such a link can
# be prevented in individual cases by putting a % sign in front of the word
# be prevented in individual cases by by putting a % sign in front of the word
# or globally by setting AUTOLINK_SUPPORT to NO.
# The default value is: YES.
@@ -1093,7 +1093,7 @@ HTML_STYLESHEET =
# cascading style sheets that are included after the standard style sheets
# created by doxygen. Using this option one can overrule certain style aspects.
# This is preferred over using HTML_STYLESHEET since it does not replace the
# standard style sheet and is therefore more robust against future updates.
# standard style sheet and is therefor more robust against future updates.
# Doxygen will copy the style sheet files to the output directory.
# Note: The order of the extra stylesheet files is of importance (e.g. the last
# stylesheet in the list overrules the setting of the previous ones in the
@@ -1636,7 +1636,7 @@ EXTRA_PACKAGES =
# Note: Only use a user-defined header if you know what you are doing! The
# following commands have a special meaning inside the header: $title,
# $datetime, $date, $doxygenversion, $projectname, $projectnumber,
# $projectbrief, $projectlogo. Doxygen will replace $title with the empty string,
# $projectbrief, $projectlogo. Doxygen will replace $title with the empy string,
# for the replacement values of the other commands the user is referred to
# HTML_HEADER.
# This tag requires that the tag GENERATE_LATEX is set to YES.
+3 -4
View File
@@ -28,7 +28,6 @@ HTMLPAGES = $(AVPROGS-yes:%=doc/%.html) $(AVPROGS-yes:%=doc/%-all.html) $(COMP
doc/mailing-list-faq.html \
doc/nut.html \
doc/platform.html \
doc/drawvg-reference.html \
$(SRC_PATH)/doc/bootstrap.min.css \
$(SRC_PATH)/doc/style.min.css \
$(SRC_PATH)/doc/default.css \
@@ -54,7 +53,7 @@ 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 +68,11 @@ 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)
+6 -98
View File
@@ -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
@@ -216,52 +189,6 @@ see page 44-46 or section 5.5 of
Extract the core from a E-AC-3 stream, dropping extra channels.
@section eia608_to_smpte436m
Convert from a @code{EIA_608} stream to a @code{SMPTE_436M_ANC} data stream, wrapping the closed captions in CTA-708 CDP VANC packets.
@table @option
@item line_number
Choose which line number the generated VANC packets should go on. You generally want either line 9 (the default) or 11.
@item wrapping_type
Choose the SMPTE 436M wrapping type, defaults to @samp{vanc_frame}.
It accepts the values:
@table @samp
@item vanc_frame
VANC frame (interlaced or segmented progressive frame)
@item vanc_field_1
@item vanc_field_2
@item vanc_progressive_frame
@end table
@item sample_coding
Choose the SMPTE 436M sample coding, defaults to @samp{8bit_luma}.
It accepts the values:
@table @samp
@item 8bit_luma
8-bit component luma samples
@item 8bit_color_diff
8-bit component color difference samples
@item 8bit_luma_and_color_diff
8-bit component luma and color difference samples
@item 10bit_luma
10-bit component luma samples
@item 10bit_color_diff
10-bit component color difference samples
@item 10bit_luma_and_color_diff
10-bit component luma and color difference samples
@item 8bit_luma_parity_error
8-bit component luma samples with parity error
@item 8bit_color_diff_parity_error
8-bit component color difference samples with parity error
@item 8bit_luma_and_color_diff_parity_error
8-bit component luma and color difference samples with parity error
@end table
@item initial_cdp_sequence_cntr
The initial value of the CDP's 16-bit unsigned integer @code{cdp_hdr_sequence_cntr} and @code{cdp_ftr_sequence_cntr} fields. Defaults to 0.
@item cdp_frame_rate
Set the CDP's @code{cdp_frame_rate} field. This doesn't actually change the timing of the data stream, it just changes the values inserted in that field in the generated CDP packets. Defaults to @samp{30000/1001}.
@end table
@section extract_extradata
Extract the in-band extradata.
@@ -496,21 +423,9 @@ Please note that this filter is auto-inserted for MPEG-TS (muxer
@section h264_redundant_pps
This applies a specific fixup to some Blu-ray BDMV H264 streams
which contain redundant PPSs. The PPSs modify irrelevant parameters
of the stream, confusing other transformations which require
the correct extradata.
The encoder used on these impacted streams adds extra PPSs throughout
the stream, varying the initial QP and whether weighted prediction
was enabled. This causes issues after copying the stream into
a global header container, as the starting PPS is not suitable
for the rest of the stream. One side effect, for example,
is seeking will return garbled output until a new PPS appears.
This BSF removes the extra PPSs and rewrites the slice headers
such that the stream uses a single leading PPS in the global header,
which resolves the issue.
This applies a specific fixup to some Blu-ray streams which contain
redundant PPSs modifying irrelevant parameters of the stream which
confuse other transformations which require correct extradata.
@section hevc_metadata
@@ -781,12 +696,12 @@ ffmpeg -i INPUT -c copy -bsf noise=1 output.mkv
Drop every video packet not marked as a keyframe after timestamp 30s but do not
modify any of the remaining packets.
@example
ffmpeg -i INPUT -c copy -bsf:v noise=drop='gt(pts*tb\,30)*not(key)' output.mkv
ffmpeg -i INPUT -c copy -bsf:v noise=drop='gt(t\,30)*not(key)' output.mkv
@end example
Drop one second of audio every 10 seconds and add some random noise to the rest.
@example
ffmpeg -i INPUT -c copy -bsf:a noise=amount=-1:drop='between(mod(pts*tb\,10)\,9\,10)' output.mkv
ffmpeg -i INPUT -c copy -bsf:a noise=amount=-1:drop='between(mod(t\,10)\,9\,10)' output.mkv
@end example
@section null
@@ -940,8 +855,6 @@ Set expressions for PTS, DTS or both.
Set expression for duration.
@item time_base
Set output time base.
@item prescale
Set whether to convert time fields to user-set output time base before evaluation of expressions. Defaults to 0.
@end table
The expressions are evaluated through the eval API and can contain the following
@@ -1002,7 +915,7 @@ The next input PTS.
The next input duration.
@item TB
The timebase in which time fields are denominated. The user-set output timebase if prescale is enabled, else the input timebase.
The timebase of stream packet belongs.
@item TB_OUT
The output timebase.
@@ -1023,11 +936,6 @@ ffmpeg -i INPUT -c:a copy -bsf:a setts=pts=DTS out.mkv
Log basic packet information. Mainly useful for testing, debugging,
and development.
@section smpte436m_to_eia608
Convert from a @code{SMPTE_436M_ANC} data stream to a @code{EIA_608} stream,
extracting the closed captions from CTA-708 CDP VANC packets, and ignoring all other data.
@anchor{text2movsub}
@section text2movsub
+6 -7
View File
@@ -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.
@@ -25,13 +30,6 @@ fate
fate-list
List all fate/regression test targets.
fate-list-failing
List the fate tests that failed the last time they were executed.
fate-clear-reports
Remove the test reports from previous test executions (getting rid of
potentially stale results from fate-list-failing).
install
Install headers, libraries and programs.
@@ -65,3 +63,4 @@ make -j<num>
make -k
Continue build in case of errors, this is useful for the regression tests
sometimes but note that it will still not run all reg tests.
-12
View File
@@ -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})
@@ -666,8 +664,6 @@ for codecs that support it. At present, those are H.264 and VP9.
@item film_grain
Export film grain parameters through frame side data (see @code{AV_FRAME_DATA_FILM_GRAIN_PARAMS}).
Supported at present by AV1 decoders.
@item enhancements
Export picture enhancement metadata through frame side data, e.g. LCEVC (see @code{AV_FRAME_DATA_LCEVC}).
@end table
@item threads @var{integer} (@emph{decoding/encoding,video})
@@ -915,14 +911,6 @@ Possible values:
@end table
@item alpha_mode @var{integer} (@emph{decoding/encoding,video})
Possible values:
@table @samp
@item premultiplied
@item straight
@end table
@item log_level_offset @var{integer}
Set the log level offset.
+23 -14
View File
@@ -119,6 +119,16 @@ The following options are supported by the libdav1d wrapper.
@table @option
@item framethreads
Set amount of frame threads to use during decoding. The default value is 0 (autodetect).
This option is deprecated for libdav1d >= 1.0 and will be removed in the future. Use the
option @code{max_frame_delay} and the global option @code{threads} instead.
@item tilethreads
Set amount of tile threads to use during decoding. The default value is 0 (autodetect).
This option is deprecated for libdav1d >= 1.0 and will be removed in the future. Use the
global option @code{threads} instead.
@item max_frame_delay
Set max amount of frames the decoder may buffer internally. The default value is 0
(autodetect).
@@ -307,6 +317,14 @@ 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.
@@ -338,15 +356,6 @@ value is 0 (disabled).
@end table
@section libmpeghdec
libmpeghdec decoder wrapper.
libmpeghdec allows libmpeghdec to decode the MPEG-H 3D audio codec.
Requires the presence of the libmpeghdec headers and library during
configuration. You need to explicitly configure the build with
@code{--enable-libmpeghdec --enable-nonfree}.
@section libopencore-amrnb
libopencore-amrnb decoder wrapper.
@@ -386,7 +395,7 @@ without this library.
@c man end AUDIO DECODERS
@chapter Subtitles Decoders
@c man begin SUBTITLES DECODERS
@c man begin SUBTILES DECODERS
@section libaribb24
@@ -418,7 +427,7 @@ Enabled by default.
Yet another ARIB STD-B24 caption decoder using external @dfn{libaribcaption}
library.
Implements profiles A and C of the Japanese ARIB STD-B24 standard,
Implements profiles A and C of the Japanse ARIB STD-B24 standard,
Brazilian ABNT NBR 15606-1, and Philippines version of ISDB-T.
Requires the presence of the libaribcaption headers and library
@@ -468,7 +477,7 @@ Specify comma-separated list of font family names to be used for @dfn{bitmap}
or @dfn{ass} type subtitle rendering.
Only first font name is used for @dfn{ass} type subtitle.
If not specified, use internally defined default font family.
If not specified, use internaly defined default font family.
@item -ass_single_rect @var{boolean}
ARIB STD-B24 specifies that some captions may be displayed at different
@@ -486,7 +495,7 @@ default behavior at compilation.
@item -force_outline_text @var{boolean}
Specify whether always render outline text for all characters regardless of
the indication by character style.
the indication by charactor style.
The default is @var{false}.
@@ -687,4 +696,4 @@ box and an end box, typically subtitles. Default value is 0 if
@end table
@c man end SUBTITLES DECODERS
@c man end SUBTILES DECODERS
+31 -101
View File
@@ -281,11 +281,7 @@ This demuxer accepts the following option:
@table @option
@item cenc_decryption_key
Default 16-byte key, in hex, to decrypt files encrypted using ISO Common Encryption (CENC/AES-128 CTR; ISO/IEC 23001-7).
@item cenc_decryption_keys
Dictionary of 16-byte key ID => 16-byte key, both in hex, to decrypt files encrypted using ISO Common Encryption
(CENC/AES-128 CTR; ISO/IEC 23001-7).
16-byte key, in hex, to decrypt files encrypted using ISO Common Encryption (CENC/AES-128 CTR; ISO/IEC 23001-7).
@end table
@@ -296,6 +292,7 @@ DVD-Video demuxer, powered by libdvdnav and libdvdread.
Can directly ingest DVD titles, specifically sequential PGCs, into
a conversion pipeline. Menu assets, such as background video or audio,
can also be demuxed given the menu's coordinates (at best effort).
Seeking is not supported at this time.
Block devices (DVD drives), ISO files, and directory structures are accepted.
Activate with @code{-f dvdvideo} in front of one of these inputs.
@@ -383,11 +380,11 @@ Default is false.
@item menu_lu @var{int}
The menu language to demux. In DVD, menus are grouped by language.
Default is 1, the first language unit.
Default is 0, the first language unit.
@item menu_vts @var{int}
The VTS where the menu lives, or 0 if it is a VMG menu (root-level).
Default is 1, menu of the first VTS.
Default is 0, VMG menu.
@item pgc @var{int}
The entry PGC to start playback, in conjunction with @option{pg}.
@@ -400,7 +397,8 @@ Default is 0, automatically resolve from value of @option{title}.
The entry PG to start playback, in conjunction with @option{pgc}.
Alternative to setting @option{title}.
Chapter markers are not supported at this time.
Default is 1, the first PG of the PGC.
Default is 0, automatically resolve from value of @option{title}, or
start from the beginning (PG 1) of the menu.
@item preindex @var{bool}
Enable this to have accurate chapter (PTT) markers and duration measurement,
@@ -408,6 +406,7 @@ which requires a slow second pass read in order to index the chapter marker
timestamps from NAV packets. This is non-ideal extra work for real optical drives.
It is recommended and faster to use this option with a backup of the DVD structure
stored on a hard drive. Not compatible with @option{pgc} and @option{pg}.
Not applicable to menus.
Default is 0, false.
@item trim @var{bool}
@@ -568,13 +567,6 @@ prefer to use #EXT-X-START if it's in playlist instead of live_start_index.
@item allowed_extensions
',' separated list of file extensions that hls is allowed to access.
@item extension_picky
This blocks disallowed extensions from probing
It also requires all available segments to have matching extensions to the format
except mpegts, which is always allowed.
It is recommended to set the whitelists correctly instead of depending on extensions
Enabled by default.
@item max_reload
Maximum number of times a insufficient list is attempted to be reloaded.
Default value is 1000.
@@ -668,9 +660,30 @@ Select a glob wildcard pattern type.
The pattern is interpreted like a @code{glob()} pattern. This is only
selectable if libavformat was compiled with globbing support.
@item glob_sequence @emph{(deprecated, will be removed)}
Select a mixed glob wildcard/sequence pattern.
If your version of libavformat was compiled with globbing support, and
the provided pattern contains at least one glob meta character among
@code{%*?[]@{@}} that is preceded by an unescaped "%", the pattern is
interpreted like a @code{glob()} pattern, otherwise it is interpreted
like a sequence pattern.
All glob special characters @code{%*?[]@{@}} must be prefixed
with "%". To escape a literal "%" you shall use "%%".
For example the pattern @code{foo-%*.jpeg} will match all the
filenames prefixed by "foo-" and terminating with ".jpeg", and
@code{foo-%?%?%?.jpeg} will match all the filenames prefixed with
"foo-", followed by a sequence of three characters, and terminating
with ".jpeg".
This pattern type is deprecated in favor of @var{glob} and
@var{sequence}.
@end table
Default value is @var{sequence}.
Default value is @var{glob_sequence}.
@item pixel_format
Set the pixel format of the images to read. If not specified the pixel
format is guessed from the first image file in the sequence.
@@ -838,32 +851,6 @@ Set the sample rate for libopenmpt to output.
Range is from 1000 to INT_MAX. The value default is 48000.
@end table
@anchor{mccdec}
@section mcc
Demuxer for MacCaption MCC files, it supports MCC versions 1.0 and 2.0.
MCC files store VANC data, which can include closed captions (EIA-608 and CEA-708), ancillary time code, pan-scan data, etc.
By default, for backward compatibility, the MCC demuxer extracts just the EIA-608 and CEA-708 closed captions and returns a @code{EIA_608} stream, ignoring all other VANC data.
You can change it to return all VANC data in a @code{SMPTE_436M_ANC} data stream by setting @option{-eia608_extract 0}
@subsection Examples
@itemize
@item
Convert a MCC file to Scenarist (SCC) format:
@example
ffmpeg -i CC.mcc -c:s copy CC.scc
@end example
Note that the SCC format only supports EIA-608, so this will discard all other data such as CEA-708 extensions.
@item
Merge a MCC file into a MXF file:
@example
ffmpeg -i video_and_audio.mxf -eia608_extract 0 -i CC.mcc -c copy -map 0 -map 1 out.mxf
@end example
This retains all VANC data and inserts it into the output MXF file as a @code{SMPTE_436M_ANC} data stream.
@end itemize
@section mov/mp4/3gp
Demuxer for Quicktime File Format & ISO/IEC Base Media File Format (ISO/IEC 14496-12 or MPEG-4 Part 12, ISO/IEC 15444-12 or JPEG 2000 Part 12).
@@ -944,11 +931,7 @@ Fixed key used for handling Audible AAX/AAX+ files. It has been pre-set so shoul
specify.
@item decryption_key
Default 16-byte key, in hex, to decrypt files encrypted using ISO Common Encryption (CENC/AES-128 CTR; ISO/IEC 23001-7).
@item decryption_keys
Dictionary of 16-byte key ID => 16-byte key, both in hex, to decrypt files encrypted using ISO Common Encryption
(CENC/AES-128 CTR; ISO/IEC 23001-7).
16-byte key, in hex, to decrypt files encrypted using ISO Common Encryption (CENC/AES-128 CTR; ISO/IEC 23001-7).
@item max_stts_delta
Very high sample deltas written in a trak's stts box may occasionally be intended but usually they are written in
@@ -1003,7 +986,7 @@ to 1 (-1 means automatic setting, 1 means enabled, 0 means
disabled). Default value is -1.
@item merge_pmt_versions
Reuse existing streams when a PMT's version is updated and elementary
Re-use existing streams when a PMT's version is updated and elementary
streams move to different PIDs. Default value is 0.
@item max_packet_size
@@ -1043,11 +1026,6 @@ Set input video frame rate. Default value is 25.
@item pixel_format
Set the input video pixel format. Default value is @code{yuv420p}.
@item stride
Set frame line size in bytes, only required if there are extra padding.
For multiplane formats, @option{stride} is a list of line size for each
plane.
@item video_size
Set the input video size. This value must be specified explicitly.
@end table
@@ -1060,13 +1038,6 @@ the command:
ffplay -f rawvideo -pixel_format rgb24 -video_size 320x240 -framerate 10 input.raw
@end example
Read a rawvideo with @command{ffplay}, assuming a pixel format of @code{yuv420p},
a video size of @code{1080x1920}, has 8 bytes of padding in each line of Y plane,
and 4 bytes of padding with UV plane,
@example
ffplay -f rawvideo -pixel_format yuv420p -video_size 1080x1920 -stride 1088,544,544 input.raw
@end example
@anchor{rcwtdec}
@section rcwt
@@ -1181,51 +1152,10 @@ 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
+12 -160
View File
@@ -45,7 +45,6 @@ maintained and developed.
@end itemize
All proposed code changes should be submitted for review to
@url{https://code.ffmpeg.org/FFmpeg/FFmpeg/pulls, Forgejo} or
@url{mailto:ffmpeg-devel@@ffmpeg.org, the development mailing list}, as
described in more detail in the @ref{Submitting patches} chapter. The code
should comply with the @ref{Development Policy} and follow the @ref{Coding Rules}.
@@ -71,6 +70,9 @@ variable-length arrays;
@item
complex numbers;
@item
mixed statements and declarations.
@end itemize
@subsection SIMD/DSP
@@ -113,7 +115,7 @@ Objective-C where required for interacting with macOS-specific interfaces.
@section Code formatting conventions
There are the following guidelines regarding the code style in files:
There are the following guidelines regarding the indentation in files:
@itemize @bullet
@item
@@ -133,123 +135,6 @@ K&R coding style is used.
@end itemize
The presentation is one inspired by 'indent -i4 -kr -nut'.
@subsection Examples
Some notable examples to illustrate common code style in FFmpeg:
@itemize @bullet
@item
Space around assignments and after
@code{if}/@code{do}/@code{while}/@code{for} keywords:
@example c, good
// Good
if (condition)
av_foo();
@end example
@example c, good
// Good
for (size_t i = 0; i < len; i++)
av_bar(i);
@end example
@example c, good
// Good
size_t size = 0;
@end example
However no spaces between the parentheses and condition, unless it helps
readability of complex conditions, so the following should not be done:
@example c, bad
// Bad style
if ( condition )
av_foo();
@end example
@item
No unnecessary parentheses, unless it helps readability:
@example c, good
// Good
int fields = ilace ? 2 : 1;
@end example
@item
Don't wrap single-line blocks in braces. Use braces only if there is an accompanying else statement. This keeps future code changes easier to keep track of.
@example c, good
// Good
if (bits_pixel == 24) @{
avctx->pix_fmt = AV_PIX_FMT_BGR24;
@} else if (bits_pixel == 8) @{
avctx->pix_fmt = AV_PIX_FMT_GRAY8;
@} else
return AVERROR_INVALIDDATA;
@end example
@item
Avoid assignments in conditions where it makes sense:
@example c, good
// Good
video_enc->chroma_intra_matrix = av_mallocz(sizeof(*video_enc->chroma_intra_matrix) * 64)
if (!video_enc->chroma_intra_matrix)
return AVERROR(ENOMEM);
@end example
@example c, bad
// Bad style
if (!(video_enc->chroma_intra_matrix = av_mallocz(sizeof(*video_enc->chroma_intra_matrix) * 64)))
return AVERROR(ENOMEM);
@end example
@example c, good
// Ok
while ((entry = av_dict_iterate(options, entry)))
av_log(ctx, AV_LOG_INFO, "Item '%s': '%s'\n", entry->key, entry->value);
@end example
@item
When declaring a pointer variable, the @code{*} goes with the variable not the type:
@example c, good
// Good
AVStream *stream;
@end example
@example c, bad
// Bad style
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,
change the parts that you are editing to follow these guidelines but do
not make unrelated changes in the file to make it conform to these.
@subsection Vim configuration
In order to configure Vim to follow FFmpeg formatting conventions, paste
the following snippet into your @file{.vimrc}:
@@ -386,9 +271,6 @@ symbols. If in doubt, just avoid names starting with @code{_} altogether.
@item
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.
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
@anchor{Development Policy}
@@ -488,25 +370,15 @@ 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
@@ -579,7 +451,7 @@ FFmpeg also has a defined scope - your new API must fit within it.
@subsubheading Replacing existing APIs
If your new API is replacing an existing one, it should be strictly superior to
it, so that the advantages of using the new API outweigh the cost to the
it, so that the advantages of using the new API outweight the cost to the
callers of changing their code. After adding the new API you should then
deprecate the old one and schedule it for removal, as described in
@ref{Removing interfaces}.
@@ -629,7 +501,7 @@ change in @file{doc/APIchanges}.
Backward-incompatible API or ABI changes require incrementing (bumping) the
major version number, as described in @ref{Major version bumps}. Major
bumps are significant events that happen on a schedule - so if your change
strictly requires one you should add it under @code{#if} preprocessor guards that
strictly requires one you should add it under @code{#if} preprocesor guards that
disable it until the next major bump happens.
New APIs that can be added without breaking API or ABI compatibility require
@@ -755,8 +627,7 @@ and has no lrint()')
Also please if you send several patches, send each patch as a separate mail,
do not attach several unrelated patches to the same mail.
Patches should be posted to
@uref{https://code.ffmpeg.org/FFmpeg/FFmpeg/pulls, Forgejo} or the
Patches should be posted to the
@uref{https://lists.ffmpeg.org/mailman/listinfo/ffmpeg-devel, ffmpeg-devel}
mailing list. Use @code{git send-email} when possible since it will properly
send patches without requiring extra care. If you cannot, then send patches
@@ -951,10 +822,10 @@ improves readability.
Consider adding a regression test for your code. All new modules
should be covered by tests. That includes demuxers, muxers, decoders, encoders
filters, bitstream filters, parsers. If its not possible to do that, add
an explanation why to your patchset, its ok to not test if there's a reason.
an explanation why to your patchset, its ok to not test if theres a reason.
@item
If you added NASM code please check that things still work with --disable-x86asm.
If you added YASM code please check that things still work with --disable-yasm.
@item
Test your code with valgrind and or Address Sanitizer to ensure it's free
@@ -1056,25 +927,6 @@ In case you need finer control over how valgrind is invoked, use the
@code{--target-exec='valgrind <your_custom_valgrind_options>} option in
your configure line instead.
@anchor{Maintenance}
@chapter Maintenance process
@anchor{MAINTAINERS}
@section MAINTAINERS
The developers maintaining each part of the codebase are listed in @file{MAINTAINERS}.
Being listed in @file{MAINTAINERS}, gives one the right to have git write access to
the specific repository.
@anchor{Becoming a maintainer}
@section Becoming a maintainer
People add themselves to @file{MAINTAINERS} by sending a patch like any other code
change. These get reviewed by the community like any other patch. It is expected
that, if someone has an objection to a new maintainer, she is willing to object
in public with her full name and is willing to take over maintainership for the area.
@anchor{Release process}
@chapter Release process
File diff suppressed because it is too large Load Diff
+20 -334
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@@ -106,8 +106,15 @@ debugging by setting the option to "disable".
Enables the use of the long term prediction extension which increases coding
efficiency in very low bandwidth situations such as encoding of voice or
solo piano music by extending constant harmonic peaks in bands throughout
frames. This option is implied by profile:a aac_low.
Use in conjunction with @option{-ar} to decrease the samplerate.
frames. This option is implied by profile:a aac_low and is incompatible with
aac_pred. Use in conjunction with @option{-ar} to decrease the samplerate.
@item aac_pred
Enables the use of a more traditional style of prediction where the spectral
coefficients transmitted are replaced by the difference of the current
coefficients minus the previous "predicted" coefficients. In theory and sometimes
in practice this can improve quality for low to mid bitrate audio.
This option implies the aac_main profile and is incompatible with aac_ltp.
@item profile
Sets the encoding profile, possible values:
@@ -125,6 +132,10 @@ MPEG4 specifications.
Long term prediction profile, is enabled by and will enable the @option{aac_ltp}
option. Introduced in MPEG4.
@item aac_main
Main-type prediction profile, is enabled by and will enable the @option{aac_pred}
option. Introduced in MPEG2.
@end table
If this option is unspecified it is set to @samp{aac_low}.
@end table
@@ -1038,17 +1049,13 @@ forces a wideband cutoff for bitrates < 15 kbps, unless CELT-only
Set channel mapping family to be used by the encoder. The default value of -1
uses mapping family 0 for mono and stereo inputs, and mapping family 1
otherwise. The default also disables the surround masking and LFE bandwidth
optimizations in libopus, and requires that the input contains 8 channels or
optimzations in libopus, and requires that the input contains 8 channels or
fewer.
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
@@ -1384,48 +1391,6 @@ Higher is better but slower.
@end table
@anchor{ffv1}
@section ffv1
FFv1 Encoder
@subsection Options
The following options are supported by FFmpeg's FFv1 encoder.
@table @option
@item context
Sets the context size, 0 (default) is small, 1 is big.
@item coder
Set the coder,
@table @samp
@item rice
Golomb rice coder
@item range_def
Range coder with default table
@item range_tab
Range coder with custom table
@end table
@item slicecrc
-1 (default, automatic), 1 use crc with zero initial and final state, 2 use crc with non zero initial and final state
@item qtable
@table @samp
@item default
default, automatic
@item 8bit
use 8bit default
@item greater8bit
use >8bit default
@end table
@item remap_optimizer
0 - 5, default 3, how much effort the encoder puts into optimizing the remap table.
@end table
@section GIF
GIF image/animation encoder.
@@ -1893,42 +1858,6 @@ ffmpeg -i input -c:v libaom-av1 -b:v 500K -aom-params tune=psnr:enable-tpl-model
@end table
@section liboapv
Advanced Professional Video codec encoder wrapper.
This encoder requires the presence of the liboapv headers and library
during configuration. You need to explicitly configure the build with
@option{--enable-liboapv}.
@float NOTE
Many liboapv encoder options are mapped to FFmpeg global codec options,
while unique encoder options are provided through private options.
@end float
The apv project website is at @url{https://github.com/AcademySoftwareFoundation/openapv}.
@subsection Options
The following options are supported by the liboapv wrapper.
@float NOTE
To get a more extensive documentation of the liboapv options, consult the
liboapv documentation.
@end float
@table @option
@item preset
Set the quality-speed tradeoff [fastest, fast, medium, slow, placebo, default]
@item qp
Set the quantization parameter value for CQP rate control mode.
@item oapv-params (@emph{parse_apv_params})
Set liboapvenc options using a list of @var{key}=@var{value} pairs separated
by ":". See the liboapv encoder user guide for a list of accepted parameters.
@end table
@section libsvtav1
SVT-AV1 encoder wrapper.
@@ -2000,49 +1929,6 @@ by ":". See the SVT-AV1 encoder user guide for a list of accepted parameters.
@end table
@section libsvtjpegxs
SVT-JPEG-XS encoder wrapper.
Requires the presence of the SVT-JPEG-XS headers and library during configuration.
You need to explicitly configure the build with @code{--enable-libsvtjpegxs}.
@subsection Options
@table @option
@item decomp_v
Set vertical decomposition level
@item decomp_h
Set horizontal decomposition level
@item quantization
Set the quantization algorithm.
@table @samp
@item deadzone
@item uniform
@end table
@item coding-signs
Enable Signs handling strategy
@table @samp
@item disable
@item fast
@item full
@end table
@item coding-sigf
Enable Significance coding
@item coding-vpred
Enable Vertical Prediction coding
@table @samp
@item disable
@item no_residuals
@item no_coeffs
@end table
@end table
@section libjxl
libjxl JPEG XL encoder wrapper.
@@ -3087,10 +2973,6 @@ Quantizer curve compression factor
Normally, when forcing a I-frame type, the encoder can select any type
of I-frame. This option forces it to choose an IDR-frame.
@item x265-stats
Specify the file name for 2-pass stats. This is set automatically when using
the @option{-passlogfile} option.
@item udu_sei @var{boolean}
Import user data unregistered SEI if available into output. Default is 0 (off).
@@ -3393,168 +3275,14 @@ fastest.
@end table
@section MediaCodec
MediaCodec encoder wrapper enables hardware-accelerated video encoding on
Android device. It supports H.264, H.265 (HEVC), VP8, VP9, MPEG-4, and AV1
encoding (whether works or not is device dependent).
Android provides two sets of APIs: Java MediaCodec and NDK MediaCodec. The
MediaCodec encoder wrapper supports both. Note that the NDK MediaCodec API
operates without requiring JVM, but may fail to function outside the JVM
environment due to dependencies on system framework services, particularly
after Android 15.
@table @option
@item ndk_codec @var{boolean}
Use the NDK-based MediaCodec API instead of the Java API. Enabled by default
if @code{av_jni_get_java_vm()} return NULL.
@item ndk_async @var{boolean}
Use NDK MediaCodec in async mode. Async mode has less overhead than poll in a
loop in sync mode. The drawback of async mode is AV_CODEC_FLAG_GLOBAL_HEADER
doesn't work (use extract_extradata bsf when necessary). It doesn't work and
will be disabled automatically on devices below Android 8.0.
@item codec_name @var{string}
A codec type can have multiple implementations on a single device, this option
specify which backend to use (via MediaCodec createCodecByName API). It's NULL
by default, and encoder is created by createEncoderByType.
@item bitrate_mode @var{integer}
Possible values:
@table @samp
@item cq
Constant quality mode
@item vbr
Variable bitrate mode
@item cbr
Constant bitrate mode
@item cbr_fd
Constant bitrate mode with frame drops
@end table
@item pts_as_dts @var{boolean}
Use PTS as DTS. This is a workaround since MediaCodec API doesn't provide
decoding timestamp. It is enabled automatically if B frame is 0.
@item operating_rate @var{integer}
The desired operating rate that the codec will need to operate at, zero for
unspecified. This is used for cases like high-speed/slow-motion video capture,
where the video encoder format contains the target playback rate (e.g. 30fps),
but the component must be able to handle the high operating capture rate (e.g.
240fps). This rate will be used by codec for resource planning and setting the
operating points.
@item qp_i_min @var{integer}
Minimum quantization parameter for I frame.
@item qp_p_min @var{integer}
Minimum quantization parameter for P frame.
@item qp_b_min @var{integer}
Minimum quantization parameter for B frame.
@item qp_i_max @var{integer}
Maximum quantization parameter for I frame.
@item qp_p_max @var{integer}
Maximum quantization parameter for P frame.
@item qp_b_max @var{integer}
Maximum quantization parameter for B frame.
@end table
@section MediaFoundation
The following wrappers for encoders in the MediaFoundation framework are
available:
@itemize
@item h264_mf
@item hevc_mf
@item av1_mf
@end itemize
These support both software and hardware encoding.
This provides wrappers to encoders (both audio and video) in the
MediaFoundation framework. It can access both SW and HW encoders.
Video encoders can take input in either of nv12 or yuv420p form
(some encoders support both, some support only either - in practice,
nv12 is the safer choice, especially among HW encoders).
Hardware-accelerated encoding requires D3D11, including hardware
scaling capabilities through the scale_d3d11 filter.
To list all available options for the MediaFoundation encoders, use:
@command{ffmpeg -h encoder=<encoder>} e.g. @command{ffmpeg -h encoder=h264_mf}
@subsection Options
@table @option
@item rate_control
Select rate control mode. Available modes:
@table @samp
@item default
Default mode
@item cbr
CBR mode
@item pc_vbr
Peak constrained VBR mode
@item u_vbr
Unconstrained VBR mode
@item quality
Quality mode
@item ld_vbr
Low delay VBR mode (requires Windows 8+)
@item g_vbr
Global VBR mode (requires Windows 8+)
@item gld_vbr
Global low delay VBR mode (requires Windows 8+)
@end table
@item scenario
Select usage scenario. Available scenarios:
@table @samp
@item default
Default scenario
@item display_remoting
Display remoting scenario
@item video_conference
Video conference scenario
@item archive
Archive scenario
@item live_streaming
Live streaming scenario
@item camera_record
Camera record scenario
@item display_remoting_with_feature_map
Display remoting with feature map scenario
@end table
@item quality
Set encoding quality (0-100). -1 means default quality.
@item hw_encoding
Force hardware encoding (0-1). Default is 0 (disabled).
@end table
@subsection Examples
Hardware encoding:
@example
ffmpeg -i input.mp4 -c:v h264_mf -hw_encoding 1 output.mp4
@end example
Hardware-accelerated decoding with hardware encoding:
@example
ffmpeg -hwaccel d3d11va -i input.mp4 -c:v h264_mf -hw_encoding 1 output.mp4
@end example
Hardware-accelerated decoding, HW scaling and encoding with quality setting:
@example
ffmpeg -hwaccel d3d11va -hwaccel_output_format d3d11 -i input.mp4 -vf scale_d3d11=1920:1080 -c:v hevc_mf -hw_encoding 1 -quality 80 output.mp4
@end example
@section Microsoft RLE
Microsoft RLE aka MSRLE encoder.
@@ -3629,13 +3357,6 @@ Default is 1 (on).
PNG image encoder.
@subsection Options
@table @option
@item compression_level
Sets the compression level, from 0 to 9(default)
@end table
@subsection Private options
@table @option
@@ -3643,8 +3364,6 @@ Sets the compression level, from 0 to 9(default)
Set physical density of pixels, in dots per inch, unset by default
@item dpm @var{integer}
Set physical density of pixels, in dots per meter, unset by default
@item pred @var{method}
Set prediction method (none, sub, up, avg, paeth, mixed), default is paeth
@end table
@section ProRes
@@ -3839,7 +3558,7 @@ For encoders set this flag to ON to reduce power consumption and GPU usage.
@end table
@subsection Runtime Options
Following options can be used during qsv encoding.
Following options can be used durning qsv encoding.
@table @option
@item @var{global_quality}
@@ -3949,7 +3668,7 @@ improves subjective visual quality. Enabling this flag may have negative impact
on performance and objective visual quality metric.
@item @var{low_delay_brc}
Setting this flag turns on or off LowDelayBRC feature in qsv plugin, which provides
Setting this flag turns on or off LowDelayBRC feautre in qsv plugin, which provides
more accurate bitrate control to minimize the variance of bitstream size frame
by frame. Value: -1-default 0-off 1-on
@@ -4148,7 +3867,7 @@ improves subjective visual quality. Enabling this flag may have negative impact
on performance and objective visual quality metric.
@item @var{low_delay_brc}
Setting this flag turns on or off LowDelayBRC feature in qsv plugin, which provides
Setting this flag turns on or off LowDelayBRC feautre in qsv plugin, which provides
more accurate bitrate control to minimize the variance of bitstream size frame
by frame. Value: -1-default 0-off 1-on
@@ -4382,7 +4101,7 @@ Extended bitrate control.
Depth of look ahead in number frames, available when extbrc option is enabled.
@item @var{low_delay_brc}
Setting this flag turns on or off LowDelayBRC feature in qsv plugin, which provides
Setting this flag turns on or off LowDelayBRC feautre in qsv plugin, which provides
more accurate bitrate control to minimize the variance of bitstream size frame
by frame. Value: -1-default 0-off 1-on
@@ -4738,25 +4457,6 @@ Reduces detail but attempts to preserve color at extremely low bitrates.
@chapter Subtitles Encoders
@c man begin SUBTITLES ENCODERS
@section dvbsub
This codec encodes the bitmap subtitle format that is used in DVB
broadcasts and recordings. The bitmaps are typically embedded in a
container such as MPEG-TS as a separate stream.
@subsection Options
@table @option
@item min_bpp @var{integer (2, 4, or 8)}
Set a minimum bits-per-pixel value for the subtitle color lookup tables.
DVB supports 2, 4, and 8 bits-per-pixel color lookup tables. This
option enables forcing a particular bits-per-pixel value regardless of
the number of colors. Since not all players support or properly
support 2 bits-per-pixel, this value defaults to 4.
@end table
@section dvdsub
This codec encodes the bitmap subtitle format that is used in DVDs.
@@ -4784,18 +4484,4 @@ one byte per subtitle on average.
By default, this work-around is disabled.
@end table
@section lrc
This codec encodes the LRC lyrics format.
@subsection Options
@table @option
@item precision
Specify the precision of the fractional part of the timestamp. Time base is
determined based on this value.
Defaults to 2 for centiseconds.
@end table
@c man end SUBTITLES ENCODERS
-1
View File
@@ -96,7 +96,6 @@ int main(int argc, char *argv[])
avio_ctx = avio_alloc_context(avio_ctx_buffer, avio_ctx_buffer_size,
0, &bd, &read_packet, NULL, NULL);
if (!avio_ctx) {
av_freep(&avio_ctx_buffer);
ret = AVERROR(ENOMEM);
goto end;
}
+1 -5
View File
@@ -128,10 +128,6 @@ int main(int argc, char **argv)
outfilename = argv[2];
pkt = av_packet_alloc();
if (!pkt) {
fprintf(stderr, "Could not allocate AVPacket\n");
exit(1); /* or proper cleanup and returning */
}
/* find the MPEG audio decoder */
codec = avcodec_find_decoder(AV_CODEC_ID_MP2);
@@ -165,7 +161,7 @@ int main(int argc, char **argv)
}
outfile = fopen(outfilename, "wb");
if (!outfile) {
fprintf(stderr, "Could not open %s\n", outfilename);
av_free(c);
exit(1);
}
+13 -16
View File
@@ -30,6 +30,8 @@
* file to be played with ffplay.
*/
#include <unistd.h>
#include <libavcodec/avcodec.h>
#include <libavformat/avformat.h>
#include <libavfilter/buffersink.h>
@@ -94,7 +96,8 @@ static int init_filters(const char *filters_descr)
const AVFilter *abuffersink = avfilter_get_by_name("abuffersink");
AVFilterInOut *outputs = avfilter_inout_alloc();
AVFilterInOut *inputs = avfilter_inout_alloc();
static const int out_sample_rate = 8000;
static const enum AVSampleFormat out_sample_fmts[] = { AV_SAMPLE_FMT_S16, -1 };
static const int out_sample_rates[] = { 8000, -1 };
const AVFilterLink *outlink;
AVRational time_base = fmt_ctx->streams[audio_stream_index]->time_base;
@@ -120,40 +123,34 @@ static int init_filters(const char *filters_descr)
}
/* buffer audio sink: to terminate the filter chain. */
buffersink_ctx = avfilter_graph_alloc_filter(filter_graph, abuffersink, "out");
if (!buffersink_ctx) {
ret = avfilter_graph_create_filter(&buffersink_ctx, abuffersink, "out",
NULL, NULL, filter_graph);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot create audio buffer sink\n");
ret = AVERROR(ENOMEM);
goto end;
}
ret = av_opt_set(buffersink_ctx, "sample_formats", "s16",
AV_OPT_SEARCH_CHILDREN);
ret = av_opt_set_int_list(buffersink_ctx, "sample_fmts", out_sample_fmts, -1,
AV_OPT_SEARCH_CHILDREN);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot set output sample format\n");
goto end;
}
ret = av_opt_set(buffersink_ctx, "channel_layouts", "mono",
AV_OPT_SEARCH_CHILDREN);
ret = av_opt_set(buffersink_ctx, "ch_layouts", "mono",
AV_OPT_SEARCH_CHILDREN);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot set output channel layout\n");
goto end;
}
ret = av_opt_set_array(buffersink_ctx, "samplerates", AV_OPT_SEARCH_CHILDREN,
0, 1, AV_OPT_TYPE_INT, &out_sample_rate);
ret = av_opt_set_int_list(buffersink_ctx, "sample_rates", out_sample_rates, -1,
AV_OPT_SEARCH_CHILDREN);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot set output sample rate\n");
goto end;
}
ret = avfilter_init_dict(buffersink_ctx, NULL);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot initialize audio buffer sink\n");
goto end;
}
/*
* Set the endpoints for the filter graph. The filter_graph will
* be linked to the graph described by filters_descr.
+9 -13
View File
@@ -27,6 +27,8 @@
* @example decode_filter_video.c
*/
#define _XOPEN_SOURCE 600 /* for usleep */
#include <unistd.h>
#include <stdio.h>
#include <stdlib.h>
@@ -36,7 +38,6 @@
#include <libavfilter/buffersrc.h>
#include <libavutil/mem.h>
#include <libavutil/opt.h>
#include <libavutil/time.h>
const char *filter_descr = "scale=78:24,transpose=cclock";
/* other way:
@@ -98,6 +99,7 @@ static int init_filters(const char *filters_descr)
AVFilterInOut *outputs = avfilter_inout_alloc();
AVFilterInOut *inputs = avfilter_inout_alloc();
AVRational time_base = fmt_ctx->streams[video_stream_index]->time_base;
enum AVPixelFormat pix_fmts[] = { AV_PIX_FMT_GRAY8, AV_PIX_FMT_NONE };
filter_graph = avfilter_graph_alloc();
if (!outputs || !inputs || !filter_graph) {
@@ -120,26 +122,20 @@ static int init_filters(const char *filters_descr)
}
/* buffer video sink: to terminate the filter chain. */
buffersink_ctx = avfilter_graph_alloc_filter(filter_graph, buffersink, "out");
if (!buffersink_ctx) {
ret = avfilter_graph_create_filter(&buffersink_ctx, buffersink, "out",
NULL, NULL, filter_graph);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot create buffer sink\n");
ret = AVERROR(ENOMEM);
goto end;
}
ret = av_opt_set(buffersink_ctx, "pixel_formats", "gray8",
AV_OPT_SEARCH_CHILDREN);
ret = av_opt_set_int_list(buffersink_ctx, "pix_fmts", pix_fmts,
AV_PIX_FMT_NONE, AV_OPT_SEARCH_CHILDREN);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot set output pixel format\n");
goto end;
}
ret = avfilter_init_dict(buffersink_ctx, NULL);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot initialize buffer sink\n");
goto end;
}
/*
* Set the endpoints for the filter graph. The filter_graph will
* be linked to the graph described by filters_descr.
@@ -194,7 +190,7 @@ static void display_frame(const AVFrame *frame, AVRational time_base)
delay = av_rescale_q(frame->pts - last_pts,
time_base, AV_TIME_BASE_Q);
if (delay > 0 && delay < 1000000)
av_usleep(delay);
usleep(delay);
}
last_pts = frame->pts;
}
+1 -5
View File
@@ -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);
+5 -10
View File
@@ -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);
}
+9 -31
View File
@@ -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);
-1
View File
@@ -109,7 +109,6 @@ static int open_codec_context(AVFormatContext *fmt_ctx, enum AVMediaType type)
ret = avcodec_parameters_to_context(dec_ctx, st->codecpar);
if (ret < 0) {
fprintf(stderr, "Failed to copy codec parameters to codec context\n");
avcodec_free_context(&dec_ctx);
return ret;
}
+4 -9
View File
@@ -270,6 +270,7 @@ int main(int argc, char *argv[])
AVFilterGraph *graph;
AVFilterContext *src, *sink;
AVFrame *frame;
uint8_t errstr[1024];
float duration;
int err, nb_frames, i;
@@ -294,7 +295,6 @@ int main(int argc, char *argv[])
md5 = av_md5_alloc();
if (!md5) {
av_frame_free(&frame);
fprintf(stderr, "Error allocating the MD5 context\n");
return 1;
}
@@ -302,10 +302,8 @@ int main(int argc, char *argv[])
/* Set up the filtergraph. */
err = init_filter_graph(&graph, &src, &sink);
if (err < 0) {
av_frame_free(&frame);
av_freep(&md5);
fprintf(stderr, "Unable to init filter graph:");
return 1;
goto fail;
}
/* the main filtering loop */
@@ -356,10 +354,7 @@ int main(int argc, char *argv[])
return 0;
fail:
avfilter_graph_free(&graph);
av_frame_free(&frame);
av_freep(&md5);
fprintf(stderr, "%s\n", av_err2str(err));
av_strerror(err, errstr, sizeof(errstr));
fprintf(stderr, "%s\n", errstr);
return 1;
}
+2 -9
View File
@@ -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;
}
@@ -216,10 +215,8 @@ int main(int argc, char *argv[])
return AVERROR(ENOMEM);
video = input_ctx->streams[video_stream];
if (avcodec_parameters_to_context(decoder_ctx, video->codecpar) < 0) {
avcodec_free_context(&decoder_ctx);
if (avcodec_parameters_to_context(decoder_ctx, video->codecpar) < 0)
return -1;
}
decoder_ctx->get_format = get_hw_format;
@@ -233,10 +230,6 @@ 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) {
+10 -26
View File
@@ -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;
@@ -434,7 +418,7 @@ static void open_video(AVFormatContext *oc, const AVCodec *codec,
exit(1);
}
/* allocate and init a reusable frame */
/* allocate and init a re-usable frame */
ost->frame = alloc_frame(c->pix_fmt, c->width, c->height);
if (!ost->frame) {
fprintf(stderr, "Could not allocate video frame\n");
+8 -13
View File
@@ -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);
@@ -227,8 +219,11 @@ int main(int argc, char **argv)
ret = decode_packet(decoder_ctx, frame, sw_frame, NULL, output_ctx);
finish:
if (ret < 0)
fprintf(stderr, "%s\n", av_err2str(ret));
if (ret < 0) {
char buf[1024];
av_strerror(ret, buf, sizeof(buf));
fprintf(stderr, "%s\n", buf);
}
avformat_close_input(&input_ctx);
+4 -10
View File
@@ -101,7 +101,7 @@ static int dynamic_set_parameter(AVCodecContext *avctx)
/* Set codec specific option */
if ((ret = av_opt_set_dict(avctx->priv_data, &opts)) < 0)
goto fail;
/* There is no "framerate" option in common option list. Use "-r" to set
/* There is no "framerate" option in commom option list. Use "-r" to set
* framerate, which is compatible with ffmpeg commandline. The video is
* assumed to be average frame rate, so set time_base to 1/framerate. */
e = av_dict_get(opts, "r", NULL, 0);
@@ -180,7 +180,7 @@ static int open_input_file(char *filename)
decoder = avcodec_find_decoder_by_name("mjpeg_qsv");
break;
default:
fprintf(stderr, "Codec is not supported by qsv\n");
fprintf(stderr, "Codec is not supportted by qsv\n");
return AVERROR(EINVAL);
}
@@ -289,7 +289,7 @@ static int dec_enc(AVPacket *pkt, const AVCodec *enc_codec, char *optstr)
fprintf(stderr, "Failed to set encoding parameter.\n");
goto fail;
}
/* There is no "framerate" option in common option list. Use "-r" to
/* There is no "framerate" option in commom option list. Use "-r" to
* set framerate, which is compatible with ffmpeg commandline. The
* video is assumed to be average frame rate, so set time_base to
* 1/framerate. */
@@ -351,10 +351,6 @@ int main(int argc, char **argv)
}
setting_number = (argc - 5) / 2;
dynamic_setting = av_malloc(setting_number * sizeof(*dynamic_setting));
if (!dynamic_setting) {
ret = AVERROR(ENOMEM);
goto end;
}
current_setting_number = 0;
for (int i = 0; i < setting_number; i++) {
dynamic_setting[i].frame_number = atoi(argv[i*2 + 5]);
@@ -430,9 +426,7 @@ 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);
+1 -1
View File
@@ -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);
+11 -41
View File
@@ -171,38 +171,23 @@ static int open_output_file(const char *filename)
* sample rate etc.). These properties can be changed for output
* streams easily using filters */
if (dec_ctx->codec_type == AVMEDIA_TYPE_VIDEO) {
const enum AVPixelFormat *pix_fmts = NULL;
enc_ctx->height = dec_ctx->height;
enc_ctx->width = dec_ctx->width;
enc_ctx->sample_aspect_ratio = dec_ctx->sample_aspect_ratio;
ret = avcodec_get_supported_config(enc_ctx, NULL,
AV_CODEC_CONFIG_PIX_FORMAT, 0,
(const void**)&pix_fmts, NULL);
/* take first format from list of supported formats */
enc_ctx->pix_fmt = (ret >= 0 && pix_fmts) ?
pix_fmts[0] : dec_ctx->pix_fmt;
if (encoder->pix_fmts)
enc_ctx->pix_fmt = encoder->pix_fmts[0];
else
enc_ctx->pix_fmt = dec_ctx->pix_fmt;
/* video time_base can be set to whatever is handy and supported by encoder */
enc_ctx->time_base = av_inv_q(dec_ctx->framerate);
} else {
const enum AVSampleFormat *sample_fmts = NULL;
enc_ctx->sample_rate = dec_ctx->sample_rate;
ret = av_channel_layout_copy(&enc_ctx->ch_layout, &dec_ctx->ch_layout);
if (ret < 0)
return ret;
ret = avcodec_get_supported_config(enc_ctx, NULL,
AV_CODEC_CONFIG_SAMPLE_FORMAT, 0,
(const void**)&sample_fmts, NULL);
/* take first format from list of supported formats */
enc_ctx->sample_fmt = (ret >= 0 && sample_fmts) ?
sample_fmts[0] : dec_ctx->sample_fmt;
enc_ctx->sample_fmt = encoder->sample_fmts[0];
enc_ctx->time_base = (AVRational){1, enc_ctx->sample_rate};
}
@@ -298,10 +283,10 @@ static int init_filter(FilteringContext* fctx, AVCodecContext *dec_ctx,
goto end;
}
buffersink_ctx = avfilter_graph_alloc_filter(filter_graph, buffersink, "out");
if (!buffersink_ctx) {
ret = avfilter_graph_create_filter(&buffersink_ctx, buffersink, "out",
NULL, NULL, filter_graph);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot create buffer sink\n");
ret = AVERROR(ENOMEM);
goto end;
}
@@ -312,12 +297,6 @@ static int init_filter(FilteringContext* fctx, AVCodecContext *dec_ctx,
av_log(NULL, AV_LOG_ERROR, "Cannot set output pixel format\n");
goto end;
}
ret = avfilter_init_dict(buffersink_ctx, NULL);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot initialize buffer sink\n");
goto end;
}
} else if (dec_ctx->codec_type == AVMEDIA_TYPE_AUDIO) {
char buf[64];
buffersrc = avfilter_get_by_name("abuffer");
@@ -343,10 +322,10 @@ static int init_filter(FilteringContext* fctx, AVCodecContext *dec_ctx,
goto end;
}
buffersink_ctx = avfilter_graph_alloc_filter(filter_graph, buffersink, "out");
if (!buffersink_ctx) {
ret = avfilter_graph_create_filter(&buffersink_ctx, buffersink, "out",
NULL, NULL, filter_graph);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot create audio buffer sink\n");
ret = AVERROR(ENOMEM);
goto end;
}
@@ -373,15 +352,6 @@ static int init_filter(FilteringContext* fctx, AVCodecContext *dec_ctx,
av_log(NULL, AV_LOG_ERROR, "Cannot set output sample rate\n");
goto end;
}
if (enc_ctx->frame_size > 0)
av_buffersink_set_frame_size(buffersink_ctx, enc_ctx->frame_size);
ret = avfilter_init_dict(buffersink_ctx, NULL);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot initialize audio buffer sink\n");
goto end;
}
} else {
ret = AVERROR_UNKNOWN;
goto end;
+1 -6
View File
@@ -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. */
+4 -4
View File
@@ -88,7 +88,7 @@ static int encode_write(AVCodecContext *avctx, AVFrame *frame, FILE *fout)
enc_pkt->stream_index = 0;
ret = fwrite(enc_pkt->data, enc_pkt->size, 1, fout);
av_packet_unref(enc_pkt);
if (!ret) {
if (ret != enc_pkt->size) {
ret = AVERROR(errno);
break;
}
@@ -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;
}
+1 -3
View File
@@ -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);
+18 -45
View File
@@ -42,60 +42,41 @@ targets and variables.
@chapter Using FATE from your FFmpeg source directory
If you want to run FATE on your machine, FFmpeg needs to be configured first.
You may specify the location of samples during configuration by adding the
argument @code{--samples=/path/to/fate-suite}.
If you want to run FATE on your machine you need to have the samples
in place. You can get the samples via the build target fate-rsync.
Use this command from the top-level source directory:
@example
./configure --enable-somelib --samples=/path/to/fate-suite
make fate-rsync SAMPLES=fate-suite/
make fate SAMPLES=fate-suite/
@end example
After successful configuration, you need to add and/or update the samples in place:
The above commands set the samples location by passing a makefile
variable via command line. It is also possible to set the samples
location at source configuration time by invoking configure with
@option{--samples=<path to the samples directory>}. Afterwards you can
invoke the makefile targets without setting the @var{SAMPLES} makefile
variable. This is illustrated by the following commands:
@example
make fate-rsync
@end example
Now you are ready to run FATE:
@example
make fate
@end example
See @ref{makefile variables} for a list of arguments that can be added.
If you did not set the samples path during configuration, or if you wish to
override it just before starting FATE, you can do so in one of three ways.
Either by setting a make variable:
@example
make fate-rsync SAMPLES=/path/to/fate-suite
make fate SAMPLES=/path/to/fate-suite
@end example
or by setting an environment variable for the current session:
@example
export FATE_SAMPLES=/path/to/fate-suite
./configure --samples=fate-suite/
make fate-rsync
make fate
@end example
or in isolation for a single command by prepending it:
Yet another way to tell FATE about the location of the sample
directory is by making sure the environment variable FATE_SAMPLES
contains the path to your samples directory. This can be achieved
by e.g. putting that variable in your shell profile or by setting
it in your interactive session.
@example
FATE_SAMPLES=/path/to/fate-suite make fate-rsync
FATE_SAMPLES=/path/to/fate-suite make fate
FATE_SAMPLES=fate-suite/ make fate
@end example
This variable can also be set in your shell profile to make it permanent.
@float NOTE
Do not put a '~' character in the samples path to indicate a home
directory. Because of shell nuances, this will cause FATE to fail.
Also, on Windows the path has to be relative to the build path
which in this case is the FFmpeg source directory.
@end float
Beware that some assertions are disabled by default, so mind setting
@@ -227,19 +208,11 @@ Download/synchronize sample files to the configured samples directory.
@item fate-list
Will list all fate/regression test targets.
@item fate-list-failing
List the fate tests that failed the last time they were executed.
@item fate-clear-reports
Remove the test reports from previous test executions (getting rid of
potentially stale results from fate-list-failing).
@item fate
Run the FATE test suite (requires the fate-suite dataset).
@end table
@section Makefile variables
@anchor{makefile variables}
@table @env
@item V
+1 -8
View File
@@ -1,33 +1,26 @@
slot= # some unique identifier
repo=https://git.ffmpeg.org/ffmpeg.git # the source repository
repo=git://source.ffmpeg.org/ffmpeg.git # the source repository
#branch=release/2.6 # the branch to test
samples= # path to samples directory
workdir= # directory in which to do all the work
#fate_recv="ssh -T fate@fate.ffmpeg.org" # command to submit report
comment= # optional description
build_only= # set to "yes" for a compile-only instance that skips tests
skip_clean= # set to "yes" to preserve build/install directories
ignore_tests=
# the following are optional and map to configure options
arch=
cpu=
toolchain=
cross_prefix=
ar=
as=
cc=
cxx=
ld=
nm=
target_os=
sysroot=
target_exec=
target_path=
target_samples=
extra_cflags=
extra_cxxflags=
extra_objcflags=
extra_ldflags=
extra_libs=
extra_conf= # extra configure options not covered above
+156 -543
View File
@@ -21,24 +21,22 @@ ffmpeg [@var{global_options}] @{[@var{input_file_options}] -i @file{input_url}@}
inputs - including live grabbing/recording devices - filter, and transcode them
into a plethora of output formats.
@command{ffmpeg} reads from an arbitrary number of inputs (which can be regular
@command{ffmpeg} reads from an arbitrary number of input "files" (which can be regular
files, pipes, network streams, grabbing devices, etc.), specified by the
@code{-i} option, and writes to an arbitrary number of outputs, which are
specified by a plain output url. Anything found on the command line which cannot
be interpreted as an option is considered to be an output url.
@code{-i} option, and writes to an arbitrary number of output "files", which are
specified by a plain output url. Anything found on the command line which
cannot be interpreted as an option is considered to be an output url.
Each input or output can, in principle, contain any number of elementary streams
of different types (video/audio/subtitle/attachment/data), though the allowed
stream counts and/or types may be limited by the container format. Selecting
which streams from which inputs will go into which output is either done
automatically or with the @code{-map} option (see the @ref{Stream selection}
chapter).
Each input or output url can, in principle, contain any number of streams of
different types (video/audio/subtitle/attachment/data). The allowed number and/or
types of streams may be limited by the container format. Selecting which
streams from which inputs will go into which output is either done automatically
or with the @code{-map} option (see the Stream selection chapter).
To refer to inputs/outputs in options, you must use their indices (0-based).
E.g. the first input is @code{0}, the second is @code{1}, etc. Similarly,
streams within an input/output are referred to by their indices. E.g. @code{2:3}
refers to the fourth stream in the third input or output. Also see the
@ref{Stream specifiers} chapter.
To refer to input files in options, you must use their indices (0-based). E.g.
the first input file is @code{0}, the second is @code{1}, etc. Similarly, streams
within a file are referred to by their indices. E.g. @code{2:3} refers to the
fourth stream in the third input file. Also see the Stream specifiers chapter.
As a general rule, options are applied to the next specified
file. Therefore, order is important, and you can have the same
@@ -87,405 +85,140 @@ The format option may be needed for raw input files.
@chapter Detailed description
@c man begin DETAILED DESCRIPTION
@command{ffmpeg} builds a transcoding pipeline out of the components listed
below. The program's operation then consists of input data chunks flowing from
the sources down the pipes towards the sinks, while being transformed by the
components they encounter along the way.
The transcoding process in @command{ffmpeg} for each output can be described by
the following diagram:
The following kinds of components are available:
@itemize
@item
@emph{Demuxers} (short for "demultiplexers") read an input source in order to
extract
@itemize
@item
global properties such as metadata or chapters;
@item
list of input elementary streams and their properties
@end itemize
One demuxer instance is created for each @option{-i} option, and sends encoded
@emph{packets} to @emph{decoders} or @emph{muxers}.
In other literature, demuxers are sometimes called @emph{splitters}, because
their main function is splitting a file into elementary streams (though some
files only contain one elementary stream).
A schematic representation of a demuxer looks like this:
@verbatim
┌──────────┬───────────────────────┐
│ demuxer │ │ packets for stream 0
╞══════════╡ elementary stream 0 ├──────────────────────►
│ │ │
│ global ├───────────────────────┤
│properties│ │ packets for stream 1
│ and │ elementary stream 1 ├──────────────────────►
│ metadata │ │
│ ├───────────────────────┤
│ │ │
│ │ ........... │
│ │ │
│ ├───────────────────────┤
│ │ │ packets for stream N
│ │ elementary stream N ├──────────────────────►
│ │ │
└──────────┴───────────────────────┘
│ read from file, network stream,
│ grabbing device, etc.
_______ ______________
| | | |
| input | demuxer | encoded data | decoder
| file | ---------> | packets | -----+
|_______| |______________| |
v
_________
| |
| decoded |
| frames |
|_________|
________ ______________ |
| | | | |
| output | <-------- | encoded data | <----+
| file | muxer | packets | encoder
|________| |______________|
@end verbatim
@item
@emph{Decoders} receive encoded (compressed) @emph{packets} for an audio, video,
or subtitle elementary stream, and decode them into raw @emph{frames} (arrays of
pixels for video, PCM for audio). A decoder is typically associated with (and
receives its input from) an elementary stream in a @emph{demuxer}, but sometimes
may also exist on its own (see @ref{Loopback decoders}).
A schematic representation of a decoder looks like this:
@verbatim
┌─────────┐
packets │ │ raw frames
─────────►│ decoder ├────────────►
│ │
└─────────┘
@end verbatim
@item
@emph{Filtergraphs} process and transform raw audio or video @emph{frames}. A
filtergraph consists of one or more individual @emph{filters} linked into a
graph. Filtergraphs come in two flavors - @emph{simple} and @emph{complex},
configured with the @option{-filter} and @option{-filter_complex} options,
respectively.
A simple filtergraph is associated with an @emph{output elementary stream}; it
receives the input to be filtered from a @emph{decoder} and sends filtered
output to that output stream's @emph{encoder}.
A simple video filtergraph that performs deinterlacing (using the @code{yadif}
deinterlacer) followed by resizing (using the @code{scale} filter) can look like
this:
@verbatim
┌────────────────────────┐
│ simple filtergraph │
frames from ╞════════════════════════╡ frames for
a decoder │ ┌───────┐ ┌───────┐ │ an encoder
────────────►├─►│ yadif ├─►│ scale ├─►│────────────►
│ └───────┘ └───────┘ │
└────────────────────────┘
@end verbatim
A complex filtergraph is standalone and not associated with any specific stream.
It may have multiple (or zero) inputs, potentially of different types (audio or
video), each of which receiving data either from a decoder or another complex
filtergraph's output. It also has one or more outputs that feed either an
encoder or another complex filtergraph's input.
The following example diagram represents a complex filtergraph with 3 inputs and
2 outputs (all video):
@verbatim
┌─────────────────────────────────────────────────┐
│ complex filtergraph │
╞═════════════════════════════════════════════════╡
frames ├───────┐ ┌─────────┐ ┌─────────┐ ┌────────┤ frames
─────────►│input 0├─►│ overlay ├─────►│ overlay ├─►│output 0├────────►
├───────┘ │ │ │ │ └────────┤
frames ├───────┐╭►│ │ ╭►│ │ │
─────────►│input 1├╯ └─────────┘ │ └─────────┘ │
├───────┘ │ │
frames ├───────┐ ┌─────┐ ┌─────┬─╯ ┌────────┤ frames
─────────►│input 2├►│scale├►│split├───────────────►│output 1├────────►
├───────┘ └─────┘ └─────┘ └────────┤
└─────────────────────────────────────────────────┘
@end verbatim
Frames from second input are overlaid over those from the first. Frames from the
third input are rescaled, then the duplicated into two identical streams. One of
them is overlaid over the combined first two inputs, with the result exposed as
the filtergraph's first output. The other duplicate ends up being the
filtergraph's second output.
@item
@emph{Encoders} receive raw audio, video, or subtitle @emph{frames} and encode
them into encoded @emph{packets}. The encoding (compression) process is
typically @emph{lossy} - it degrades stream quality to make the output smaller;
some encoders are @emph{lossless}, but at the cost of much higher output size. A
video or audio encoder receives its input from some filtergraph's output,
subtitle encoders receive input from a decoder (since subtitle filtering is not
supported yet). Every encoder is associated with some muxer's @emph{output
elementary stream} and sends its output to that muxer.
A schematic representation of an encoder looks like this:
@verbatim
┌─────────┐
raw frames │ │ packets
────────────►│ encoder ├─────────►
│ │
└─────────┘
@end verbatim
@item
@emph{Muxers} (short for "multiplexers") receive encoded @emph{packets} for
their elementary streams from encoders (the @emph{transcoding} path) or directly
from demuxers (the @emph{streamcopy} path), interleave them (when there is more
than one elementary stream), and write the resulting bytes into the output file
(or pipe, network stream, etc.).
A schematic representation of a muxer looks like this:
@verbatim
┌──────────────────────┬───────────┐
packets for stream 0 │ │ muxer │
──────────────────────►│ elementary stream 0 ╞═══════════╡
│ │ │
├──────────────────────┤ global │
packets for stream 1 │ │properties │
──────────────────────►│ elementary stream 1 │ and │
│ │ metadata │
├──────────────────────┤ │
│ │ │
│ ........... │ │
│ │ │
├──────────────────────┤ │
packets for stream N │ │ │
──────────────────────►│ elementary stream N │ │
│ │ │
└──────────────────────┴─────┬─────┘
write to file, network stream, │
grabbing device, etc. │
@end verbatim
@end itemize
@section Streamcopy
The simplest pipeline in @command{ffmpeg} is single-stream
@emph{streamcopy}, that is copying one @emph{input elementary stream}'s packets
without decoding, filtering, or encoding them. As an example, consider an input
file called @file{INPUT.mkv} with 3 elementary streams, from which we take the
second and write it to file @file{OUTPUT.mp4}. A schematic representation of
such a pipeline looks like this:
@verbatim
┌──────────┬─────────────────────┐
│ demuxer │ │ unused
╞══════════╡ elementary stream 0 ├────────╳
│ │ │
│INPUT.mkv ├─────────────────────┤ ┌──────────────────────┬───────────┐
│ │ │ packets │ │ muxer │
│ │ elementary stream 1 ├─────────►│ elementary stream 0 ╞═══════════╡
│ │ │ │ │OUTPUT.mp4 │
│ ├─────────────────────┤ └──────────────────────┴───────────┘
│ │ │ unused
│ │ elementary stream 2 ├────────╳
│ │ │
└──────────┴─────────────────────┘
@end verbatim
The above pipeline can be constructed with the following commandline:
@example
ffmpeg -i INPUT.mkv -map 0:1 -c copy OUTPUT.mp4
@end example
In this commandline
@itemize
@item
there is a single input @file{INPUT.mkv};
@item
there are no input options for this input;
@item
there is a single output @file{OUTPUT.mp4};
@item
there are two output options for this output:
@itemize
@item
@code{-map 0:1} selects the input stream to be used - from input with index 0
(i.e. the first one) the stream with index 1 (i.e. the second one);
@item
@code{-c copy} selects the @code{copy} encoder, i.e. streamcopy with no decoding
or encoding.
@end itemize
@end itemize
Streamcopy is useful for changing the elementary stream count, container format,
or modifying container-level metadata. Since there is no decoding or encoding,
it is very fast and there is no quality loss. However, it might not work in some
cases because of a variety of factors (e.g. certain information required by the
target container is not available in the source). Applying filters is obviously
also impossible, since filters work on decoded frames.
More complex streamcopy scenarios can be constructed - e.g. combining streams
from two input files into a single output:
@verbatim
┌──────────┬────────────────────┐ ┌────────────────────┬───────────┐
│ demuxer 0│ │ packets │ │ muxer │
╞══════════╡elementary stream 0 ├────────►│elementary stream 0 ╞═══════════╡
│INPUT0.mkv│ │ │ │OUTPUT.mp4 │
└──────────┴────────────────────┘ ├────────────────────┤ │
┌──────────┬────────────────────┐ │ │ │
│ demuxer 1│ │ packets │elementary stream 1 │ │
╞══════════╡elementary stream 0 ├────────►│ │ │
│INPUT1.aac│ │ └────────────────────┴───────────┘
└──────────┴────────────────────┘
@end verbatim
that can be built by the commandline
@example
ffmpeg -i INPUT0.mkv -i INPUT1.aac -map 0:0 -map 1:0 -c copy OUTPUT.mp4
@end example
The output @option{-map} option is used twice here, creating two streams in the
output file - one fed by the first input and one by the second. The single
instance of the @option{-c} option selects streamcopy for both of those streams.
You could also use multiple instances of this option together with
@ref{Stream specifiers} to apply different values to each stream, as will be
demonstrated in following sections.
A converse scenario is splitting multiple streams from a single input into
multiple outputs:
@verbatim
┌──────────┬─────────────────────┐ ┌───────────────────┬───────────┐
│ demuxer │ │ packets │ │ muxer 0 │
╞══════════╡ elementary stream 0 ├─────────►│elementary stream 0╞═══════════╡
│ │ │ │ │OUTPUT0.mp4│
│INPUT.mkv ├─────────────────────┤ └───────────────────┴───────────┘
│ │ │ packets ┌───────────────────┬───────────┐
│ │ elementary stream 1 ├─────────►│ │ muxer 1 │
│ │ │ │elementary stream 0╞═══════════╡
└──────────┴─────────────────────┘ │ │OUTPUT1.mp4│
└───────────────────┴───────────┘
@end verbatim
built with
@example
ffmpeg -i INPUT.mkv -map 0:0 -c copy OUTPUT0.mp4 -map 0:1 -c copy OUTPUT1.mp4
@end example
Note how a separate instance of the @option{-c} option is needed for every
output file even though their values are the same. This is because non-global
options (which is most of them) only apply in the context of the file before
which they are placed.
These examples can of course be further generalized into arbitrary remappings
of any number of inputs into any number of outputs.
@section Transcoding
@emph{Transcoding} is the process of decoding a stream and then encoding it
again. Since encoding tends to be computationally expensive and in most cases
degrades the stream quality (i.e. it is @emph{lossy}), you should only transcode
when you need to and perform streamcopy otherwise. Typical reasons to transcode
are:
@itemize
@item
applying filters - e.g. resizing, deinterlacing, or overlaying video; resampling
or mixing audio;
@item
you want to feed the stream to something that cannot decode the original codec.
@end itemize
Note that @command{ffmpeg} will transcode all audio, video, and subtitle streams
unless you specify @option{-c copy} for them.
Consider an example pipeline that reads an input file with one audio and one
video stream, transcodes the video and copies the audio into a single output
file. This can be schematically represented as follows
@verbatim
┌──────────┬─────────────────────┐
│ demuxer │ │ audio packets
╞══════════╡ stream 0 (audio) ├─────────────────────────────────────╮
│ │ │ │
│INPUT.mkv ├─────────────────────┤ video ┌─────────┐ raw │
│ │ │ packets │ video │ video frames │
│ │ stream 1 (video) ├─────────►│ decoder ├──────────────╮ │
│ │ │ │ │ │ │
└──────────┴─────────────────────┘ └─────────┘ │ │
▼ ▼
│ │
┌──────────┬─────────────────────┐ video ┌─────────┐ │ │
│ muxer │ │ packets │ video │ │ │
╞══════════╡ stream 0 (video) │◄─────────┤ encoder ├──────────────╯ │
│ │ │ │(libx264)│ │
│OUTPUT.mp4├─────────────────────┤ └─────────┘ │
│ │ │ │
│ │ stream 1 (audio) │◄────────────────────────────────────╯
│ │ │
└──────────┴─────────────────────┘
@end verbatim
and implemented with the following commandline:
@example
ffmpeg -i INPUT.mkv -map 0:v -map 0:a -c:v libx264 -c:a copy OUTPUT.mp4
@end example
Note how it uses stream specifiers @code{:v} and @code{:a} to select input
streams and apply different values of the @option{-c} option to them; see the
@ref{Stream specifiers} section for more details.
@command{ffmpeg} calls the libavformat library (containing demuxers) to read
input files and get packets containing encoded data from them. When there are
multiple input files, @command{ffmpeg} tries to keep them synchronized by
tracking lowest timestamp on any active input stream.
Encoded packets are then passed to the decoder (unless streamcopy is selected
for the stream, see further for a description). The decoder produces
uncompressed frames (raw video/PCM audio/...) which can be processed further by
filtering (see next section). After filtering, the frames are passed to the
encoder, which encodes them and outputs encoded packets. Finally, those are
passed to the muxer, which writes the encoded packets to the output file.
@section Filtering
When transcoding, audio and video streams can be filtered before encoding, with
either a @emph{simple} or @emph{complex} filtergraph.
Before encoding, @command{ffmpeg} can process raw audio and video frames using
filters from the libavfilter library. Several chained filters form a filter
graph. @command{ffmpeg} distinguishes between two types of filtergraphs:
simple and complex.
@subsection Simple filtergraphs
Simple filtergraphs are those that have exactly one input and output, both of
the same type (audio or video). They are configured with the per-stream
@option{-filter} option (with @option{-vf} and @option{-af} aliases for
@option{-filter:v} (video) and @option{-filter:a} (audio) respectively). Note
that simple filtergraphs are tied to their output stream, so e.g. if you have
multiple audio streams, @option{-af} will create a separate filtergraph for each
one.
the same type. In the above diagram they can be represented by simply inserting
an additional step between decoding and encoding:
Taking the transcoding example from above, adding filtering (and omitting audio,
for clarity) makes it look like this:
@verbatim
┌──────────┬───────────────┐
│ demuxer │ │ ┌─────────┐
╞══════════╡ video stream │ packets │ video │ frames
│INPUT.mkv │ ├─────────►│ decoder ├─────►───╮
│ │ │ └─────────┘ │
└──────────┴───────────────┘ │
╭───────────◄───────────╯
│ ┌────────────────────────┐
│ │ simple filtergraph │
│ ╞════════════════════════╡
│ │ ┌───────┐ ┌───────┐ │
╰──►├─►│ yadif ├─►│ scale ├─►├╮
│ └───────┘ └───────┘ ││
└────────────────────────┘│
┌──────────┬───────────────┐ video ┌─────────┐ │
│ muxer │ │ packets │ video │ │
╞══════════╡ video stream │◄─────────┤ encoder ├───────◄───────╯
│OUTPUT.mp4│ │ │ │
│ │ │ └─────────┘
└──────────┴───────────────┘
_________ ______________
| | | |
| decoded | | encoded data |
| frames |\ _ | packets |
|_________| \ /||______________|
\ __________ /
simple _\|| | / encoder
filtergraph | filtered |/
| frames |
|__________|
@end verbatim
@subsection Complex filtergraphs
Simple filtergraphs are configured with the per-stream @option{-filter} option
(with @option{-vf} and @option{-af} aliases for video and audio respectively).
A simple filtergraph for video can look for example like this:
@verbatim
_______ _____________ _______ ________
| | | | | | | |
| input | ---> | deinterlace | ---> | scale | ---> | output |
|_______| |_____________| |_______| |________|
@end verbatim
Note that some filters change frame properties but not frame contents. E.g. the
@code{fps} filter in the example above changes number of frames, but does not
touch the frame contents. Another example is the @code{setpts} filter, which
only sets timestamps and otherwise passes the frames unchanged.
@subsection Complex filtergraphs
Complex filtergraphs are those which cannot be described as simply a linear
processing chain applied to one stream. This is the case, for example, when the
graph has more than one input and/or output, or when output stream type is
different from input. Complex filtergraphs are configured with the
@option{-filter_complex} option. Note that this option is global, since a
complex filtergraph, by its nature, cannot be unambiguously associated with a
single stream or file. Each instance of @option{-filter_complex} creates a new
complex filtergraph, and there can be any number of them.
processing chain applied to one stream. This is the case, for example, when the graph has
more than one input and/or output, or when output stream type is different from
input. They can be represented with the following diagram:
@verbatim
_________
| |
| input 0 |\ __________
|_________| \ | |
\ _________ /| output 0 |
\ | | / |__________|
_________ \| complex | /
| | | |/
| input 1 |---->| filter |\
|_________| | | \ __________
/| graph | \ | |
/ | | \| output 1 |
_________ / |_________| |__________|
| | /
| input 2 |/
|_________|
@end verbatim
Complex filtergraphs are configured with the @option{-filter_complex} option.
Note that this option is global, since a complex filtergraph, by its nature,
cannot be unambiguously associated with a single stream or file.
The @option{-lavfi} option is equivalent to @option{-filter_complex}.
A trivial example of a complex filtergraph is the @code{overlay} filter, which
has two video inputs and one video output, containing one video overlaid on top
of the other. Its audio counterpart is the @code{amix} filter.
@anchor{Loopback decoders}
@section Stream copy
Stream copy is a mode selected by supplying the @code{copy} parameter to the
@option{-codec} option. It makes @command{ffmpeg} omit the decoding and encoding
step for the specified stream, so it does only demuxing and muxing. It is useful
for changing the container format or modifying container-level metadata. The
diagram above will, in this case, simplify to this:
@verbatim
_______ ______________ ________
| | | | | |
| input | demuxer | encoded data | muxer | output |
| file | ---------> | packets | -------> | file |
|_______| |______________| |________|
@end verbatim
Since there is no decoding or encoding, it is very fast and there is no quality
loss. However, it might not work in some cases because of many factors. Applying
filters is obviously also impossible, since filters work on uncompressed data.
@section Loopback decoders
While decoders are normally associated with demuxer streams, it is also possible
to create "loopback" decoders that decode the output from some encoder and allow
@@ -526,41 +259,8 @@ reads an input video and
@end itemize
Such a transcoding pipeline can be represented with the following diagram:
@verbatim
┌──────────┬───────────────┐
│ demuxer │ │ ┌─────────┐ ┌─────────┐ ┌────────────────────┐
╞══════════╡ video stream │ │ video │ │ video │ │ null muxer │
│ INPUT │ ├──►│ decoder ├──┬────────►│ encoder ├─┬─►│(discards its input)│
│ │ │ └─────────┘ │ │(libx264)│ │ └────────────────────┘
└──────────┴───────────────┘ │ └─────────┘ │
╭───────◄──╯ ┌─────────┐ │
│ │loopback │ │
│ ╭─────◄──────┤ decoder ├────◄──╯
│ │ └─────────┘
│ │
│ │
│ │ ┌───────────────────┐
│ │ │complex filtergraph│
│ │ ╞═══════════════════╡
│ │ │ ┌─────────────┐ │
╰─╫─►├─►│ hstack ├─►├╮
╰─►├─►│ │ ││
│ └─────────────┘ ││
└───────────────────┘│
┌──────────┬───────────────┐ ┌─────────┐ │
│ muxer │ │ │ video │ │
╞══════════╡ video stream │◄─┤ encoder ├───────◄──────────╯
│ OUTPUT │ │ │ (ffv1) │
│ │ │ └─────────┘
└──────────┴───────────────┘
@end verbatim
@c man end DETAILED DESCRIPTION
@anchor{Stream selection}
@chapter Stream selection
@c man begin STREAM SELECTION
@@ -921,25 +621,24 @@ ffmpeg -i INPUT -metadata:s:a:0 language=eng OUTPUT
@end example
@item -disposition[:stream_specifier] @var{value} (@emph{output,per-stream})
Sets the disposition flags for a stream.
Sets the disposition for a stream.
Default value: by default, all disposition flags are copied from the input stream,
unless the output stream this option applies to is fed by a complex filtergraph
- in that case no disposition flags are set by default.
By default, the disposition is copied from the input stream, unless the output
stream this option applies to is fed by a complex filtergraph - in that case the
disposition is unset by default.
@var{value} is a sequence of disposition flags separated by '+' or '-'. A '+'
prefix adds the given disposition, '-' removes it. If the first flag is also
prefixed with '+' or '-', the resulting disposition is the default value
updated by @var{value}. If the first flag is not prefixed, the resulting
disposition is @var{value}. It is also possible to clear the disposition by
setting it to 0.
@var{value} is a sequence of items separated by '+' or '-'. The first item may
also be prefixed with '+' or '-', in which case this option modifies the default
value. Otherwise (the first item is not prefixed) this options overrides the
default value. A '+' prefix adds the given disposition, '-' removes it. It is
also possible to clear the disposition by setting it to 0.
If no @code{-disposition} options were specified for an output file, ffmpeg will
automatically set the 'default' disposition flag on the first stream of each type,
automatically set the 'default' disposition on the first stream of each type,
when there are multiple streams of this type in the output file and no stream of
that type is already marked as default.
The @code{-dispositions} option lists the known disposition flags.
The @code{-dispositions} option lists the known dispositions.
For example, to make the second audio stream the default stream:
@example
@@ -957,29 +656,6 @@ To add an embedded cover/thumbnail:
ffmpeg -i in.mp4 -i IMAGE -map 0 -map 1 -c copy -c:v:1 png -disposition:v:1 attached_pic out.mp4
@end example
To add the 'original' and remove the 'comment' disposition flag from the first
audio stream without removing its other disposition flags:
@example
ffmpeg -i in.mkv -c copy -disposition:a:0 +original-comment out.mkv
@end example
To remove the 'original' and add the 'comment' disposition flag to the first
audio stream without removing its other disposition flags:
@example
ffmpeg -i in.mkv -c copy -disposition:a:0 -original+comment out.mkv
@end example
To set only the 'original' and 'comment' disposition flags on the first audio
stream (and remove its other disposition flags):
@example
ffmpeg -i in.mkv -c copy -disposition:a:0 original+comment out.mkv
@end example
To remove all disposition flags from the first audio stream:
@example
ffmpeg -i in.mkv -c copy -disposition:a:0 0 out.mkv
@end example
Not all muxers support embedded thumbnails, and those who do, only support a few formats, like JPEG or PNG.
@item -program [title=@var{title}:][program_num=@var{program_num}:]st=@var{stream}[:st=@var{stream}...] (@emph{output})
@@ -1048,7 +724,7 @@ The following flags are available:
@table @option
@item recon_gain
Whether to signal if recon_gain is present as metadata in parameter blocks within frames
Wether to signal if recon_gain is present as metadata in parameter blocks within frames
@end table
@item output_gain
@@ -1196,9 +872,9 @@ ffmpeg -i front.wav -i back.wav -i center.wav -i lfe.wav
demixing=parameter_id=998,
recon_gain=parameter_id=101,
layer=ch_layout=stereo,
layer=ch_layout=5.1(side),
layer=ch_layout=5.1,
-stream_group type=iamf_mix_presentation:id=2:stg=0:annotations=en-us=Mix_Presentation,
submix=parameter_id=100:parameter_rate=48000|element=stg=0:parameter_id=100:annotations=en-us=Scalable_Submix|layout=sound_system=stereo|layout=sound_system=5.1(side)
submix=parameter_id=100:parameter_rate=48000|element=stg=0:parameter_id=100:annotations=en-us=Scalable_Submix|layout=sound_system=stereo|layout=sound_system=5.1
-streamid 0:0 -streamid 1:1 -streamid 2:2 -streamid 3:3 output.iamf
@end example
@@ -1373,62 +1049,31 @@ The properties where a change triggers reinitialization are,
for video, frame resolution or pixel format;
for audio, sample format, sample rate, channel count or channel layout.
@item -drop_changed[:@var{stream_specifier}] @var{integer} (@emph{input,per-stream})
This boolean option determines whether a frame with differing frame parameters mid-stream
gets dropped instead of leading to filtergraph reinitialization, as that would lead to loss
of filter state. Generally useful to avoid corrupted yet decodable packets in live streaming
inputs. Default is false.
@item -filter_threads @var{nb_threads} (@emph{global})
Defines how many threads are used to process a filter pipeline. Each pipeline
will produce a thread pool with this many threads available for parallel processing.
The default is the number of available CPUs.
@item -filter_buffered_frames @var{nb_frames} (@emph{global})
Defines the maximum number of buffered frames allowed in a filtergraph. Under
normal circumstances, a filtergraph should not buffer more than a few frames,
especially if frames are being fed to it and read from it in a balanced way
(which is the intended behavior in ffmpeg). That said, this option allows you
to limit the total number of frames buffered across all links in a filtergraph.
If more frames are generated, filtering is aborted and an error is returned.
The default value is 0, which means no limit.
@item -pre[:@var{stream_specifier}] @var{preset_name} (@emph{output,per-stream})
Specify the preset for matching stream(s).
@item -stats (@emph{global})
Log encoding progress/statistics as "info"-level log (see @code{-loglevel}).
It is on by default, to explicitly disable it you need to specify @code{-nostats}.
Print encoding progress/statistics. It is on by default, to explicitly
disable it you need to specify @code{-nostats}.
@item -stats_period @var{time} (@emph{global})
Set period at which encoding progress/statistics are updated. Default is 0.5 seconds.
@item -print_graphs (@emph{global})
Prints execution graph details to stderr in the format set via -print_graphs_format.
@item -print_graphs_file @var{filename} (@emph{global})
Writes execution graph details to the specified file in the format set via -print_graphs_format.
@item -print_graphs_format @var{format} (@emph{global})
Sets the output format (available formats are: default, compact, csv, flat, ini, json, xml, mermaid, mermaidhtml)
The default format is json.
@item -progress @var{url} (@emph{global})
Send program-friendly progress information to @var{url}.
Progress information is written periodically and at the end of
the encoding process. It is made of "@var{key}=@var{value}" lines. @var{key}
consists of only alphanumeric characters. The last key of a sequence of
progress information is always "progress" with the value "continue" or "end".
progress information is always "progress".
The update period is set using @code{-stats_period}.
For example, log progress information to stdout:
@example
ffmpeg -progress pipe:1 -i in.mkv out.mkv
@end example
@anchor{stdin option}
@item -stdin
Enable interaction on standard input. On by default unless standard input is
@@ -1569,27 +1214,6 @@ 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}
@@ -1686,7 +1310,6 @@ Force video tag/fourcc. This is an alias for @code{-tag:v}.
@item -force_key_frames[:@var{stream_specifier}] @var{time}[,@var{time}...] (@emph{output,per-stream})
@item -force_key_frames[:@var{stream_specifier}] expr:@var{expr} (@emph{output,per-stream})
@item -force_key_frames[:@var{stream_specifier}] source (@emph{output,per-stream})
@item -force_key_frames[:@var{stream_specifier}] scd_metadata (@emph{output,per-stream})
@var{force_key_frames} can take arguments of the following form:
@@ -1750,14 +1373,6 @@ the current frame being encoded is marked as a key frame in its source.
In cases where this particular source frame has to be dropped,
enforce the next available frame to become a key frame instead.
@item scd_metadata
If the argument is @code{scd_metadata}, ffmpeg will force a key frame if
the current frame contains a metadata entry with the key @code{lavfi.scd.time}.
The metadata can be added by filters like @code{scdet} and @code{scdet_vulkan}.
Avoid inserting filters that duplicate frames after @code{scdet}, as this can
cause duplicate metadata for multiple frames and repeated insertion of key
frames.
@end table
Note that forcing too many keyframes is very harmful for the lookahead
@@ -2040,9 +1655,6 @@ transcoding, without copying the frames into the system memory.
For it to work, both the decoder and the encoder must support QSV acceleration
and no filters must be used.
@item videotoolbox
Use Video Toolbox hardware acceleration.
@end table
This option has no effect if the selected hwaccel is not available or not
@@ -2187,7 +1799,7 @@ Set the size of the canvas used to render subtitles.
@section Advanced options
@table @option
@item -map [-]@var{input_file_id}[:@var{stream_specifier}][:@var{view_specifier}][:?] | @var{[linklabel]} (@emph{output})
@item -map [-]@var{input_file_id}[:@var{stream_specifier}][:@var{view_specifier}][?] | @var{[linklabel]} (@emph{output})
Create one or more streams in the output file. This option has two forms for
specifying the data source(s): the first selects one or more streams from some
@@ -2383,24 +1995,25 @@ Read input at native frame rate. This is equivalent to setting @code{-readrate 1
@item -readrate_initial_burst @var{seconds}
Set an initial read burst time, in seconds, after which @option{-re/-readrate}
will be enforced.
@item -readrate_catchup @var{speed} (@emph{input})
If either the input or output is blocked leading to actual read speed falling behind the
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 -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.
@item -fps_mode[:@var{stream_specifier}] @var{parameter} (@emph{output,per-stream})
Set video sync method / framerate mode.
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 +2043,7 @@ 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.
+4 -10
View File
@@ -208,14 +208,6 @@ Vulkan configuration using a list of @var{key}=@var{value} pairs separated by
Use HW accelerated decoding. Enable this option will enable vulkan renderer
automatically.
@item -video_bg @var{pattern}
Set the video background pattern used for transparent videos. For solid color
patterns you can use the color name or color code as specified in the
@ref{color syntax,,Color section in the ffmpeg-utils(1) manual,ffmpeg-utils}.
You can also use the special @code{tiles} keyword for a checker board style
pattern. This is also the default. The @code{none} pattern does rendering
without alpha blending effectively ignoring alpha channel.
@end table
@section While playing
@@ -259,13 +251,15 @@ Pause if the stream is not already paused, step to the next video
frame, and pause.
@item left/right
Seek backward/forward by 10 seconds or a custom interval if -seek_interval is set.
Seek backward/forward 10 seconds.
@item down/up
Seek backward/forward 1 minute.
@item page down/page up
Seek to previous/next chapter or backward/forward 10 minutes if no chapters.
Seek to the previous/next chapter.
or if there are no chapters
Seek backward/forward 10 minutes.
@item right mouse click
Seek to percentage in file corresponding to fraction of width.
+8 -26
View File
@@ -117,8 +117,7 @@ ffprobe -show_packets -select_streams v:1 INPUT
@end example
@item -show_data
Show payload data, as a hexadecimal and ASCII dump (other formats can be
selected using @option{-data_dump_format}). Coupled with
Show payload data, as a hexadecimal and ASCII dump. Coupled with
@option{-show_packets}, it will dump the packets' data. Coupled with
@option{-show_streams}, it will dump the codec extradata.
@@ -128,11 +127,6 @@ The dump is printed as the "data" field. It may contain newlines.
Show a hash of payload data, for packets with @option{-show_packets} and for
codec extradata with @option{-show_streams}.
@item -data_dump_format @var{format}
Select a format used for the data dumps enabled with the @option{-show_data}
option. The default is @code{xxd} which is a hexdump format compatible with the
well-known @command{xxd} program. @code{base64} is also supported.
@item -show_error
Show information about the error found when trying to probe the input.
@@ -145,6 +139,13 @@ stream.
All the container format information is printed within a section with
name "FORMAT".
@item -show_format_entry @var{name}
Like @option{-show_format}, but only prints the specified entry of the
container format information, rather than all. This option may be given more
than once, then all specified entries will be shown.
This option is deprecated, use @code{show_entries} instead.
@item -show_entries @var{section_entries}
Set list of entries to show.
@@ -350,19 +351,6 @@ while other writers always print them. This option enables one to control this b
Valid values are @code{always}/@code{1}, @code{never}/@code{0} and @code{auto}/@code{-1}.
Default is @var{auto}.
@item -analyze_frames
Analyze frames and/or their side data up to the provided read interval,
providing additional information that may be useful at a stream level.
Must be paired with the @option{-show_streams} option or it will have no effect.
Currently, the additional fields provided by this option when enabled are the
@code{closed_captions} and @code{film_grain} fields.
For example, to analyze the first 20 seconds and populate these fields:
@example
ffprobe -show_streams -analyze_frames -read_intervals "%+20" INPUT
@end example
@item -bitexact
Force bitexact output, useful to produce output which is not dependent
on the specific build.
@@ -374,12 +362,6 @@ Read @var{input_url}.
Write output to @var{output_url}. If not specified, the output is sent
to stdout.
@item -c:@var{media_specifier} @var{codec_name}
@itemx -codec:@var{media_specifier} @var{codec_name}
Force a specific decoder implementation for the stream identified by
@var{media_specifier}, which can assume the values @code{a} (audio),
@code{v} (video), @code{s} (subtitle), and @code{d} (data).
@end table
@c man end
-3
View File
@@ -129,7 +129,6 @@
<xsd:attribute name="pict_type" type="xsd:string"/>
<xsd:attribute name="interlaced_frame" type="xsd:int" />
<xsd:attribute name="top_field_first" type="xsd:int" />
<xsd:attribute name="lossless" type="xsd:int" />
<xsd:attribute name="repeat_pict" type="xsd:int" />
<xsd:attribute name="color_range" type="xsd:string"/>
<xsd:attribute name="color_space" type="xsd:string"/>
@@ -276,7 +275,6 @@
<xsd:attribute name="extradata" type="xsd:string" />
<xsd:attribute name="extradata_size" type="xsd:int" />
<xsd:attribute name="extradata_hash" type="xsd:string" />
<xsd:attribute name="mime_codec_string" type="xsd:string" />
<!-- video attributes -->
<xsd:attribute name="width" type="xsd:int"/>
@@ -357,7 +355,6 @@
<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>
+2 -7
View File
@@ -78,8 +78,7 @@ Match the stream by stream id (e.g. PID in MPEG-TS container).
@item m:@var{key}[:@var{value}]
Matches streams with the metadata tag @var{key} having the specified value. If
@var{value} is not given, matches streams that contain the given tag with any
value. The colon character ':' in @var{key} or @var{value} needs to be
backslash-escaped.
value.
@item disp:@var{dispositions}[:@var{additional_stream_specifier}]
Matches streams with the given disposition(s). @var{dispositions} is a list of
one or more dispositions (as printed by the @option{-dispositions} option)
@@ -98,7 +97,7 @@ These options are shared amongst the ff* tools.
@table @option
@item -L, -license
@item -L
Show license.
@item -h, -?, -help, --help [@var{arg}]
@@ -226,10 +225,6 @@ and the "Last message repeated n times" line will be omitted.
Indicates that log output should add a @code{[level]} prefix to each message
line. This can be used as an alternative to log coloring, e.g. when dumping the
log to file.
@item time
Indicates that log lines should be prefixed with time information.
@item datetime
Indicates that log lines should be prefixed with date and time information.
@end table
Flags can also be used alone by adding a '+'/'-' prefix to set/reset a single
flag without affecting other @var{flags} or changing @var{loglevel}. When
-1
View File
@@ -214,7 +214,6 @@ Frame scheduling
FF_FILTER_FORWARD_STATUS(inlink, outlink);
FF_FILTER_FORWARD_STATUS_ALL(inlink, filter);
FF_FILTER_FORWARD_WANTED(outlink, inlink);
FF_FILTER_FORWARD_WANTED_ANY(filter, inlink);
filter_frame
------------
+953 -2073
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File diff suppressed because it is too large Load Diff
-3
View File
@@ -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})
+3 -62
View File
@@ -162,7 +162,7 @@ Then pass @code{--enable-libmp3lame} to configure to enable it.
@section LCEVCdec
FFmpeg can make use of the liblcevc_dec library for LCEVC enhancement layer
FFmpeg can make use of the liblcevc_dec library for LCEVC enhacement layer
decoding on supported bitstreams.
Go to @url{https://github.com/v-novaltd/LCEVCdec} and follow the instructions
@@ -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
@@ -283,14 +250,6 @@ Go to @url{http://sourceforge.net/projects/opencore-amr/} and follow the
instructions for installing the library.
Then pass @code{--enable-libfdk-aac} to configure to enable it.
@subsection Fraunhofer MPEG-H 3D Audio decoder library
FFmpeg can make use of the Fraunhofer MPEG-H decoder library for MPEG-H 3DA decoding.
Go to @url{https://github.com/Fraunhofer-IIS/mpeghdec} and follow the
instructions for installing the library.
Then pass @code{--enable-libmpeghdec --enable-nonfree} to configure to enable it.
@subsection LC3 library
FFmpeg can make use of the Google LC3 library for LC3 decoding & encoding.
@@ -336,15 +295,6 @@ Go to @url{https://gitlab.com/AOMediaCodec/SVT-AV1/} and follow the instructions
for installing the library. Then pass @code{--enable-libsvtav1} to configure to
enable it.
@section SVT-JPEG-XS
FFmpeg can make use of the Scalable Video Technology for JPEG-XS library for JPEG-XS
decoding and encoding.
Go to @url{https://github.com/OpenVisualCloud/SVT-JPEG-XS} and follow the instructions
for installing the library. Then pass @code{--enable-libsvtjpegxs} to configure to
enable it.
@section TwoLAME
FFmpeg can make use of the TwoLAME library for MP2 encoding.
@@ -675,7 +625,6 @@ library:
@item raw AMR-NB @tab @tab X
@item raw AMR-WB @tab @tab X
@item raw APAC @tab @tab X
@item raw APV @tab X @tab X
@item raw aptX @tab X @tab X
@item raw aptX HD @tab X @tab X
@item raw Bonk @tab @tab X
@@ -688,7 +637,6 @@ library:
@item raw E-AC-3 @tab X @tab X
@item raw EVC @tab X @tab X
@item raw FLAC @tab X @tab X
@item raw G.728 @tab @tab X
@item raw GSM @tab @tab X
@item raw H.261 @tab X @tab X
@item raw H.263 @tab X @tab X
@@ -861,8 +809,6 @@ following image formats are supported:
@tab Progressive JPEG is not supported.
@item JPEG 2000 @tab X @tab X
@item JPEG-LS @tab X @tab X
@item JPEG-XS @tab E @tab E
@tab Supported through external library libsvtjpegxs
@item LJPEG @tab X @tab
@tab Lossless JPEG
@item Media 100 @tab @tab X
@@ -949,7 +895,6 @@ following image formats are supported:
@tab fourcc: apch,apcn,apcs,apco,ap4h,ap4x
@item Apple QuickDraw @tab @tab X
@tab fourcc: qdrw
@item APV @tab @tab X
@item Argonaut Video @tab @tab X
@tab Used in some Argonaut games.
@item Asus v1 @tab X @tab X
@@ -1164,7 +1109,6 @@ following image formats are supported:
@item RealVideo 3.0 @tab @tab X
@tab still far from ideal
@item RealVideo 4.0 @tab @tab X
@item RealVideo 6.0 @tab @tab X
@item Renderware TXD (TeXture Dictionary) @tab @tab X
@tab Texture dictionaries used by the Renderware Engine.
@item RivaTuner Video @tab @tab X
@@ -1288,7 +1232,6 @@ following image formats are supported:
@item ADPCM IMA Duck DK4 @tab @tab X
@tab Used in some Sega Saturn console games.
@item ADPCM IMA Radical @tab @tab X
@item ADPCM IMA Xbox @tab @tab X
@item ADPCM Microsoft @tab X @tab X
@item ADPCM MS IMA @tab X @tab X
@item ADPCM Nintendo Gamecube AFC @tab @tab X
@@ -1296,7 +1239,6 @@ following image formats are supported:
@item ADPCM Nintendo THP @tab @tab X
@item ADPCM Playstation @tab @tab X
@item ADPCM QT IMA @tab X @tab X
@item ADPCM Sanyo @tab @tab X
@item ADPCM SEGA CRI ADX @tab X @tab X
@tab Used in Sega Dreamcast games.
@item ADPCM Shockwave Flash @tab X @tab X
@@ -1328,6 +1270,8 @@ 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
@@ -1369,7 +1313,6 @@ following image formats are supported:
@item FLAC (Free Lossless Audio Codec) @tab X @tab IX
@item FTR Voice @tab @tab X
@item G.723.1 @tab X @tab X
@item G.728 @tab @tab X
@item G.729 @tab @tab X
@item GSM @tab E @tab X
@tab encoding supported through external library libgsm
@@ -1394,8 +1337,6 @@ following image formats are supported:
@item MP3 (MPEG audio layer 3) @tab E @tab IX
@tab encoding supported through external library LAME, ADU MP3 and MP3onMP4 also supported
@item MPEG-4 Audio Lossless Coding (ALS) @tab @tab X
@item MPEG-H 3D Audio @tab @tab E
@tab decoding supported through external library libmpeghdec
@item MobiClip FastAudio @tab @tab X
@item Musepack SV7 @tab @tab X
@item Musepack SV8 @tab @tab X
+2 -1
View File
@@ -71,6 +71,7 @@ git clone git@@ffmpeg.org:ffmpeg-web <target>
This will put the source of the FFmpeg website into the directory
@var{<target>} and let you push back your changes to the remote repository.
(Note that @var{gil} stands for GItoLite and is not a typo of @var{git}.)
If you don't have write-access to the ffmpeg-web repository, you can
create patches after making a read-only ffmpeg-web clone:
@@ -142,7 +143,7 @@ git log <filename(s)>
@end example
You may also use the graphical tools like @command{gitview} or @command{gitk}
or the web interface available at @url{https://git.ffmpeg.org/ffmpeg.git}.
or the web interface available at @url{http://source.ffmpeg.org/}.
@section Checking source tree status
-7
View File
@@ -1,7 +0,0 @@
ffmpeg mono ./ffmpeg.html
ffmpeg-filters mono ./ffmpeg-filters.html
ffmpeg-formats mono ./ffmpeg-formats.html
ffmpeg-resampler mono ./ffmpeg-resampler.html
ffmpeg-scaler mono ./ffmpeg-scaler.html
ffmpeg-utils mono ./ffmpeg-utils.html
drawvg-reference mono ./drawvg-reference.html
+36 -1
View File
@@ -220,6 +220,41 @@ $ ffmpeg -f avfoundation -capture_raw_data true -i "zr100:none" out.dv
@end itemize
@section bktr
BSD video input device. Deprecated and will be removed - please contact
the developers if you are interested in maintaining it.
@subsection Options
@table @option
@item framerate
Set the frame rate.
@item video_size
Set the video frame size. Default is @code{vga}.
@item standard
Available values are:
@table @samp
@item pal
@item ntsc
@item secam
@item paln
@item palm
@item ntscj
@end table
@end table
@section decklink
The decklink input device provides capture capabilities for Blackmagic
@@ -704,7 +739,7 @@ Win32 GDI-based screen capture device.
This device allows you to capture a region of the display on Windows.
Amongst options for the input filenames are such elements as:
Amongst options for the imput filenames are such elements as:
@example
desktop
@end example
+1 -76
View File
@@ -1,18 +1,8 @@
FFmpeg Infrastructure:
======================
Trademark:
~~~~~~~~~~
ffmpeg trademark registered in france by ffmpeg creator.
Domain + NS:
~~~~~~~~~~~~
ffmpeg.org domain name
ns1.avcodec.org Primary Name server (provided by Telepoint, hosted at Telepoint in bulgaria)
ns2.avcodec.org Replica Name server (provided by an ffmpeg developer, hosted at Hetzner in germany)
ns3.avcodec.org Replica Name server (provided by an ffmpeg developer, hosted at Prometeus Cdlan in italy)
Servers:
~~~~~~~~
@@ -24,26 +14,6 @@ Our Main server is hosted at telepoint.bg
for more details see: https://www.ffmpeg.org/#thanks_sponsor_0001
Nothing runs on our main server directly, instead several VMs run on it.
Main server security:
---------------------
Telepoint is one of the largest Bulgarian DC providers with multiple sibling companies offering
Internet, SDH, DWDM, peering exchange and hosting. They have multiple DC buildings in Sofia and
FFmpeg is hosted in XXXXXXXXXXXXXXXXXXXXXXXXXXXXXXX. The building is locked down and accessible
only with personal key cards that are registered. People who are granted access to a rack have
to go through the access center with their ID to get logged and receive a one-time access card
that can open the service elevator and only the hall where the destination rack is. All racks are
locked, once access is granted to someone they will get a key for the rack for the duration of
their visit. There are security cameras everywhere and personnel in the access center 24/7. As for
software security, our BIOS and IPMI are protected by password and encrypted connection, and
the machines can be accessed only by root administrators with their SSH keys. They're using
a Ubuntu LTS release and get regular security updates as they are released. We also get
notified by email for various security related events (e.g. failed sudo).
Side note - Telepoint provides the rack and physical infrastructure, their sibling company called
Telehouse which is an ISP provides the connectivity, they are directly connected with
over 15 tier-1 service providers and have over 100 PoPs (points of presence)
in various cities. You can see more on www.telehouse.bg
ffmpeg.org VM:
--------------
@@ -52,10 +22,7 @@ Web, mail, and public facing git, also website git
fftrac VM:
----------
trac.ffmpeg.org Legacy issue tracking
code.ffmpeg.org Primary issue tracking
gpg encrypted backups of the trac repositories are created once a day
and can be downloaded by any of the admins.
trac.ffmpeg.org Issue tracking
ffaux VM:
@@ -98,9 +65,6 @@ Github mirrors are redundantly synced by multiple people
You need a new git repository related to FFmpeg ? contact root at ffmpeg.org
git repositories are managed by gitolite, every change to permissions is
logged, including when, what and by whom
Fate:
~~~~~
@@ -123,47 +87,8 @@ You need a VM, docker container for FFmpeg? contact root at ffmpeg.org
Multimedia Wiki:
~~~~~~~~~~~~~~~~
The Multimedia Wiki http://wiki.multimedia.cx is ran by Mike Melanson.
While not directly part of FFmpeg infrastructure, technical codec and format
information written by FFmpeg developers can be found within.
It is our unofficial official tech wiki. For access contact Mike.
IRC:
~~~~
irc channels are at https://libera.chat/
irc channel archives are at https://libera.irclog.whitequark.org
#ffmpeg and #ffmpeg-devel founder/admins: BtbN, Michael, Compn
#ffmpeg-meeting founder/admins: BtbN, Michael
Twitter aka X:
~~~~~~~~~~~~~~
https://twitter.com/FFmpeg or https://x.com/FFmpeg
If you would like to post to twitter please contact twitter MAINTAINERS
for access. We want more developers posting to twitter!
Reddit:
~~~~~~~
https://www.reddit.com/r/ffmpeg/
moderated by Gyan
Facebook:
~~~~~~~~~
https://www.facebook.com/ffmpeg
???
Wikipedia entry:
~~~~~~~~~~~~~~~~
https://en.wikipedia.org/wiki/FFmpeg
+1 -4
View File
@@ -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:
+115
View File
@@ -0,0 +1,115 @@
CONTEXT
=======
The FFmpeg project merges all the changes from the Libav project
(https://libav.org) since the origin of the fork (around 2011).
With the exceptions of some commits due to technical/political disagreements or
issues, the changes are merged on a more or less regular schedule (daily for
years thanks to Michael, but more sparse nowadays).
WHY
===
The majority of the active developers believe the project needs to keep this
policy for various reasons.
The most important one is that we don't want our users to have to choose
between two distributors of libraries of the exact same name in order to have a
different set of features and bugfixes. By taking the responsibility of
unifying the two codebases, we allow users to benefit from the changes from the
two teams.
Today, FFmpeg has a much larger user database (we are distributed by every
major distribution), so we consider this mission a priority.
A different approach to the merge could have been to pick the changes we are
interested in and drop most of the cosmetics and other less important changes.
Unfortunately, this makes the following picks much harder, especially since the
Libav project is involved in various deep API changes. As a result, we decide
to virtually take everything done there.
Any Libav developer is of course welcome anytime to contribute directly to the
FFmpeg tree. Of course, we fully understand and are forced to accept that very
few Libav developers are interested in doing so, but we still want to recognize
their work. This leads us to create merge commits for every single one from
Libav. The original commit appears totally unchanged with full authorship in
our history (and the conflict are solved in the merge one). That way, not a
single thing from Libav will be lost in the future in case some reunification
happens, or that project disappears one way or another.
DOWNSIDES
=========
Of course, there are many downsides to this approach.
- It causes a non negligible merge commits pollution. We make sure there are
not several level of merges entangled (we do a 1:1 merge/commit), but it's
still a non-linear history.
- Many duplicated work. For instance, we added libavresample in our tree to
keep compatibility with Libav when our libswresample was already covering the
exact same purpose. The same thing happened for various elements such as the
ProRes support (but differences in features, bugs, licenses, ...). There are
many work to do to unify them, and any help is very much welcome.
- So much manpower from both FFmpeg and Libav is lost because of this mess. We
know it, and we don't know how to fix it. It takes incredible time to do
these merges, so we have even less time to work on things we personally care
about. The bad vibes also do not help with keeping our developers motivated.
- There is a growing technical risk factor with the merges due to the codebase
differing more and more.
MERGE GUIDELINES
================
The following gives developer guidelines on how to proceed when merging Libav commits.
Before starting, you can reduce the risk of errors on merge conflicts by using
a different merge conflict style:
$ git config --global merge.conflictstyle diff3
tools/libav-merge-next-commit is a script to help merging the next commit in
the queue. It assumes a remote named libav. It has two modes: merge, and noop.
The noop mode creates a merge with no change to the HEAD. You can pass a hash
as extra argument to reference a justification (it is common that we already
have the change done in FFmpeg).
Also see tools/murge, you can copy and paste a 3 way conflict into its stdin
and it will display colored diffs. Any arguments to murge (like ones to suppress
whitespace differences) are passed into colordiff.
TODO/FIXME/UNMERGED
===================
Stuff that didn't reach the codebase:
-------------------------------------
- HEVC DSP and x86 MC SIMD improvements from Libav (see https://ffmpeg.org/pipermail/ffmpeg-devel/2015-December/184777.html)
- 1f821750f hevcdsp: split the qpel functions by width instead of by the subpixel fraction
- 818bfe7f0 hevcdsp: split the epel functions by width
- 688417399 hevcdsp: split the pred functions by width
- a853388d2 hevc: change the stride of the MC buffer to be in bytes instead of elements
- 0cef06df0 checkasm: add HEVC MC tests
- e7078e842 hevcdsp: add x86 SIMD for MC
- 7993ec19a hevc: Add hevc_get_pixel_4/8/12/16/24/32/48/64
- use av_cpu_max_align() instead of hardcoding alignment requirements (see https://ffmpeg.org/pipermail/ffmpeg-devel/2017-September/215834.html)
- f44ec22e0 lavc: use av_cpu_max_align() instead of hardcoding alignment requirements
- 4de220d2e frame: allow align=0 (meaning automatic) for av_frame_get_buffer()
- Support recovery from an already present HLS playlist (see 16cb06bb30)
- Remove all output devices (see 8e7e042d41, 8d3db95f20, 6ce13070bd, d46cd24986 and https://ffmpeg.org/pipermail/ffmpeg-devel/2017-September/216904.html)
- avcodec/libaomenc: export the Sequence Header OBU as extradata (See a024c3ce9a)
Collateral damage that needs work locally:
------------------------------------------
- Merge proresenc_anatoliy.c and proresenc_kostya.c
- Fix MIPS AC3 downmix
Extra changes needed to be aligned with Libav:
----------------------------------------------
- Switching our examples to the new encode/decode API (see 67d28f4a0f)
- HEVC IDCT bit depth 12-bit support (Libav added 8 and 10 but doesn't have 12)
+1 -1
View File
@@ -341,7 +341,7 @@ recommended.
Avoid sending the same message to multiple mailing lists.
@item
Please follow our @url{https://ffmpeg.org/community.html#Code-of-Conduct, Code of Conduct}.
Please follow our @url{https://ffmpeg.org/community.html#Code-of-conduct, Code of Conduct}.
@end itemize
@chapter Help
+1
View File
@@ -60,5 +60,6 @@ Files that have MIPS copyright notice in them:
compute_antialias_float.h
lsp_mips.h
fmtconvert_mips.c
iirfilter_mips.c
mpegaudiodsp_mips_fixed.c
mpegaudiodsp_mips_float.c
+11 -306
View File
@@ -319,7 +319,7 @@ This is the same as the @samp{vob} muxer with a few differences.
@table @option
@item muxrate @var{rate}
Set user-defined mux rate expressed as a number of bits/s. If not
specified the automatically computed mux rate is employed. Default value
specied the automatically computed mux rate is employed. Default value
is @code{0}.
@item preload @var{delay}
@@ -772,7 +772,7 @@ Force a delay expressed in seconds after the last frame of each
repetition. Default value is @code{0.0}.
@item plays @var{repetitions}
specify how many times to play the content, @code{0} causes an infinite
specify how many times to play the content, @code{0} causes an infinte
loop, with @code{1} there is no loop
@end table
@@ -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
@@ -1780,7 +1770,7 @@ for looping indefinitely (default).
@item final_delay @var{delay}
Force the delay (expressed in centiseconds) after the last frame. Each frame
ends with a delay until the next frame. The default is @code{-1}, which is a
special value to tell the muxer to reuse the previous delay. In case of a
special value to tell the muxer to re-use the previous delay. In case of a
loop, you might want to customize this value to mark a pause for instance.
@end table
@@ -1866,7 +1856,7 @@ This muxer creates an .f4m (Adobe Flash Media Manifest File) manifest, an .abst
(Adobe Bootstrap File) for each stream, and segment files in a directory
specified as the output.
These needs to be accessed by an HDS player through HTTPS for it to be able to
These needs to be accessed by an HDS player throuhg HTTPS for it to be able to
perform playback on the generated stream.
@subsection Options
@@ -2446,14 +2436,13 @@ ffmpeg -re -i in.ts -b:a:0 32k -b:a:1 64k -b:v:0 1000k \
@item
Create a single variant stream. Add the @code{#EXT-X-MEDIA} tag with
@code{TYPE=SUBTITLES} in the master playlist with webvtt subtitle group name
'subtitle' and optional subtitle name, e.g. 'English'. Make sure the input
file has one text subtitle stream at least.
'subtitle'. Make sure the input file has one text subtitle stream at least.
@example
ffmpeg -y -i input_with_subtitle.mkv \
-b:v:0 5250k -c:v h264 -pix_fmt yuv420p -profile:v main -level 4.1 \
-b:a:0 256k \
-c:s webvtt -c:a mp2 -ar 48000 -ac 2 -map 0:v -map 0:a:0 -map 0:s:0 \
-f hls -var_stream_map "v:0,a:0,s:0,sgroup:subtitle,sname:English" \
-f hls -var_stream_map "v:0,a:0,s:0,sgroup:subtitle" \
-master_pl_name master.m3u8 -t 300 -hls_time 10 -hls_init_time 4 -hls_list_size \
10 -master_pl_publish_rate 10 -hls_flags \
delete_segments+discont_start+split_by_time ./tmp/video.m3u8
@@ -2548,7 +2537,7 @@ these applications, audio may be played back on a wide range of devices, e.g.,
headphones, mobile phones, tablets, TVs, sound bars, home theater systems, and
big screens.
This format was promoted and designed by Alliance for Open Media.
This format was promoted and desgined by Alliance for Open Media.
For more information about this format, see @url{https://aomedia.org/iamf/}.
@@ -2654,12 +2643,6 @@ 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 +2651,7 @@ 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 +2659,7 @@ 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 +2711,6 @@ 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 +2750,6 @@ 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:
@@ -3016,44 +2939,6 @@ ffmpeg -i INPUT -f md5 -
@end example
@end itemize
@anchor{mccenc}
@section mcc
Muxer for MacCaption MCC files, it supports MCC versions 1.0 and 2.0.
MCC files store VANC data, which can include closed captions (EIA-608 and CEA-708), ancillary time code, pan-scan data, etc.
@subsection Options
The muxer options are:
@table @option
@item override_time_code_rate
Override the @code{Time Code Rate} value in the output. Defaults to trying to deduce from the stream's @code{time_base}, which often doesn't work.
@item use_u_alias
Use the @code{U} alias for the byte sequence @code{E1h 00h 00h 00h}.
Disabled by default because some @file{.mcc} files disagree on whether it has 2 or 3 zero bytes.
@item mcc_version
The MCC file format version. Must be either 1 or 2, defaults to 2.
@item creation_program
The creation program. Defaults to this version of FFmpeg.
@item creation_time
The creation time. Defaults to the current time.
@end table
@subsection Examples
@itemize
@item
Extract a MXF @code{SMPTE_436M_ANC} stream from a MXF file and write it to a MCC file at 30 fps.
@example
ffmpeg -i input.mxf -c copy -map 0:d -override_time_code_rate 30 out.mcc
@end example
@item
Extract EIA-608/CTA-708 closed captions from a @file{.mp4} file and write them to a MCC file at 29.97 fps.
@example
ffmpeg -f lavfi -i "movie=input.mp4[out+subcc]" -c:s copy -map 0:s -override_time_code_rate 30000/1001 out.mcc
@end example
@end itemize
@section microdvd
MicroDVD subtitle format muxer.
@@ -3145,7 +3030,7 @@ 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 +3057,6 @@ 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 +3233,6 @@ 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 +3478,6 @@ 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 +3504,6 @@ 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 +3769,6 @@ 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.
@@ -4104,73 +3878,4 @@ ffmpeg -f webm_dash_manifest -i video1.webm \
manifest.xml
@end example
@anchor{whip}
@section whip
WebRTC (Real-Time Communication) muxer that supports sub-second latency streaming according to
the WHIP (WebRTC-HTTP ingestion protocol) specification.
This is an experimental feature.
It uses HTTP as a signaling protocol to exchange SDP capabilities and ICE lite candidates. Then,
it uses STUN binding requests and responses to establish a session over UDP. Subsequently, it
initiates a DTLS handshake to exchange the SRTP encryption keys. Lastly, it splits video and
audio frames into RTP packets and encrypts them using SRTP.
Ensure that you use H.264 without B frames and Opus for the audio codec. For example, to convert
an input file with @command{ffmpeg} to WebRTC:
@example
ffmpeg -re -i input.mp4 -acodec libopus -ar 48000 -ac 2 \
-vcodec libx264 -profile:v baseline -tune zerolatency -threads 1 -bf 0 \
-f whip "http://localhost:1985/rtc/v1/whip/?app=live&stream=livestream"
@end example
For this example, we have employed low latency options, resulting in an end-to-end latency of
approximately 150ms.
@subsection Options
This muxer supports the following options:
@table @option
@item handshake_timeout @var{integer}
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.
@item ts_buffer_size @var{integer}
Set the buffer size, in bytes, of underlying protocol.
Default value is -1(auto). The UDP auto selects a reasonable value.
@item whip_flags @var{flags}
Possible values:
@table @samp
@item dtls_active
The muxer will try to set dtls active role and send the first client hello.
@end table
@item rtp_history @var{integer}
Set the number of RTP history items to store.
Default value is 512.
@item authorization @var{string}
The optional Bearer token for WHIP Authorization.
@item cert_file @var{string}
The optional certificate file path for DTLS.
@item key_file @var{string}
The optional private key file path for DTLS.
@end table
@c man end MUXERS
+1
View File
@@ -157,3 +157,4 @@ PFD[32] would for example be signed 32 bit little-endian IEEE float
@item XVID @tab non-compliant MPEG-4 generated by old Xvid
@item XVIX @tab non-compliant MPEG-4 generated by old Xvid with interlacing bug
@end multitable
+9 -4
View File
@@ -5,7 +5,7 @@ What to optimize:
-----------------
If you plan to do non-x86 architecture specific optimizations (SIMD normally),
then take a look in the x86/ directory, as most important functions are
already optimized for x86.
already optimized for MMX.
If you want to do x86 optimizations then you can either try to fine-tune the
stuff in the x86 directory or find some other functions in the C source to
@@ -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.
@@ -188,13 +188,18 @@ Code that depends on data in registries being untouched, should be written as
a single __asm__() statement. Ideally, a single function contains only one
__asm__() block.
Use external asm (nasm) or inline asm (__asm__()), do not use intrinsics.
Use external asm (nasm/yasm) 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)
and external asm (.s or .asm files, handled by an assembler such as nasm)
and external asm (.s or .asm files, handled by an assembler such as nasm/yasm)
are accepted in FFmpeg. Which one to use differs per specific case.
- if your code is intended to be inlined in a C function, inline asm is always
+111
View File
@@ -301,6 +301,45 @@ ffmpeg -re -i INPUT -c:v rawvideo -pix_fmt bgra -f fbdev /dev/fb0
See also @url{http://linux-fbdev.sourceforge.net/}, and fbset(1).
@section opengl
OpenGL output device. Deprecated and will be removed.
To enable this output device you need to configure FFmpeg with @code{--enable-opengl}.
This output device allows one to render to OpenGL context.
Context may be provided by application or default SDL window is created.
When device renders to external context, application must implement handlers for following messages:
@code{AV_DEV_TO_APP_CREATE_WINDOW_BUFFER} - create OpenGL context on current thread.
@code{AV_DEV_TO_APP_PREPARE_WINDOW_BUFFER} - make OpenGL context current.
@code{AV_DEV_TO_APP_DISPLAY_WINDOW_BUFFER} - swap buffers.
@code{AV_DEV_TO_APP_DESTROY_WINDOW_BUFFER} - destroy OpenGL context.
Application is also required to inform a device about current resolution by sending @code{AV_APP_TO_DEV_WINDOW_SIZE} message.
@subsection Options
@table @option
@item background
Set background color. Black is a default.
@item no_window
Disables default SDL window when set to non-zero value.
Application must provide OpenGL context and both @code{window_size_cb} and @code{window_swap_buffers_cb} callbacks when set.
@item window_title
Set the SDL window title, if not specified default to the filename specified for the output device.
Ignored when @option{no_window} is set.
@item window_size
Set preferred window size, can be a string of the form widthxheight or a video size abbreviation.
If not specified it defaults to the size of the input video, downscaled according to the aspect ratio.
Mostly usable when @option{no_window} is not set.
@end table
@subsection Examples
Play a file on SDL window using OpenGL rendering:
@example
ffmpeg -i INPUT -f opengl "window title"
@end example
@section oss
OSS (Open Sound System) output device.
@@ -367,6 +406,78 @@ Play a file on default device on default server:
ffmpeg -i INPUT -f pulse "stream name"
@end example
@section sdl
SDL (Simple DirectMedia Layer) output device. Deprecated and will be removed.
For monitoring purposes in FFmpeg, pipes and a video player such as ffplay can be used:
@example
ffmpeg -i INPUT -f nut -c:v rawvideo - | ffplay -
@end example
"sdl2" can be used as alias for "sdl".
This output device allows one to show a video stream in an SDL
window. Only one SDL window is allowed per application, so you can
have only one instance of this output device in an application.
To enable this output device you need libsdl installed on your system
when configuring your build.
For more information about SDL, check:
@url{http://www.libsdl.org/}
@subsection Options
@table @option
@item window_borderless
Set SDL window border off.
Default value is 0 (enable window border).
@item window_enable_quit
Enable quit action (using window button or keyboard key)
when non-zero value is provided.
Default value is 1 (enable quit action).
@item window_fullscreen
Set fullscreen mode when non-zero value is provided.
Default value is zero.
@item window_size
Set the SDL window size, can be a string of the form
@var{width}x@var{height} or a video size abbreviation.
If not specified it defaults to the size of the input video,
downscaled according to the aspect ratio.
@item window_title
Set the SDL window title, if not specified default to the filename
specified for the output device.
@item window_x
@item window_y
Set the position of the window on the screen.
@end table
@subsection Interactive commands
The window created by the device can be controlled through the
following interactive commands.
@table @key
@item q, ESC
Quit the device immediately.
@end table
@subsection Examples
The following command shows the @command{ffmpeg} output is an
SDL window, forcing its size to the qcif format:
@example
ffmpeg -i INPUT -c:v rawvideo -pix_fmt yuv420p -window_size qcif -f sdl "SDL output"
@end example
@section sndio
sndio audio output device.
+2 -18
View File
@@ -158,7 +158,7 @@ You will need the following prerequisites:
To set up a proper environment in MSYS2, you need to run @code{msys_shell.bat} from
the Visual Studio or Intel Compiler command prompt.
Place @code{nasm.exe} somewhere in your @code{PATH}.
Place @code{yasm.exe} somewhere in your @code{PATH}.
Next, make sure any other headers and libs you want to use, such as zlib, are
located in a spot that the compiler can see. Do so by modifying the @code{LIB}
@@ -301,7 +301,7 @@ These library packages are only available from
@uref{http://sourceware.org/cygwinports/, Cygwin Ports}:
@example
libSDL-devel, libgsm-devel, libmp3lame-devel,
yasm, libSDL-devel, libgsm-devel, libmp3lame-devel,
speex-devel, libtheora-devel, libxvidcore-devel
@end example
@@ -330,20 +330,4 @@ and for a build with shared libraries
./configure --target-os=mingw32 --enable-shared --disable-static --extra-cflags=-mno-cygwin --extra-libs=-mno-cygwin
@end example
@section ARM64EC
FFmpeg does not intend to support the Windows ARM64EC build configuration;
patches for changing the individual libraries for the purposes of ARM64EC will
not be accepted.
It may still be possible to build FFmpeg in this build configuration;
such a build may seem to work to some extent. Such a build may have some
amounts of ABI inconsistencies though - which we are not willing to fix.
(Changing aarch64 assembly code to strictly support ARM64EC would require
ifdeffing/changing essentially all such assembly code, and would be a
huge maintenance burden. Additionally, changing the library ABI
surfaces to fix ABI inconsistencies could require extensive and intrusive
changes.)
@bye
+53 -263
View File
@@ -71,7 +71,7 @@ client may also set a user/password for authentication. The default for both
fields is "guest". Name of virtual host on broker can be set with vhost. The
default value is "/".
Multiple subscribers may stream from the broker using the command:
Muliple subscribers may stream from the broker using the command:
@example
ffplay amqp://[[user]:[password]@@]hostname[:port][/vhost]
@end example
@@ -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
@@ -354,17 +352,6 @@ means auto (seekable for normal files, non-seekable for named pipes).
Many demuxers handle seekable and non-seekable resources differently,
overriding this might speed up opening certain files at the cost of losing some
features (e.g. accurate seeking).
@item pkt_size
Set the maximum packet size used for file I/O. A smaller value may reduce
memory usage. A higher value may increase throughput especially with networked
filesystems.
For reading, if explicitly set, it overrides the default internal buffer size
(32 KB) and limits the maximum amount of data read per operation.
For writing, this sets the size of each write operation. The default is 256 KB
for regular files, 32 KB otherwise.
@end table
@section ftp
@@ -419,6 +406,26 @@ Gophers protocol.
The Gopher protocol with TLS encapsulation.
@section hls
Read Apple HTTP Live Streaming compliant segmented stream as
a uniform one. The M3U8 playlists describing the segments can be
remote HTTP resources or local files, accessed using the standard
file protocol.
The nested protocol is declared by specifying
"+@var{proto}" after the hls URI scheme name, where @var{proto}
is either "file" or "http".
@example
hls+http://host/path/to/remote/resource.m3u8
hls+file://path/to/local/resource.m3u8
@end example
Using this protocol is discouraged - the hls demuxer should work
just as well (if not, please report the issues) and is more complete.
To use the hls demuxer instead, simply use the direct URLs to the
m3u8 files.
@section http
HTTP (Hyper Text Transfer Protocol).
@@ -453,30 +460,7 @@ 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).
@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
@option{short_seek_size} to the same value or higher. Doing so allows FFmpeg
to reuse a single HTTP connection wherever possible.
@item initial_request_size
Limit the size of initial requests. Similar to @code{request_size}, but only
used during initial format parsing. Useful for formats like MXF or MOV that
require frequent seeks during header parsing. Lasts until the demuxer makes a
read request larger than this size (without a seek in between), after which
the implementation will continue using requests as usual. Disabled (set to 0)
by default.
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.
Use persistent connections if set to 1, default is 0.
@item post_data
Set custom HTTP post data.
@@ -623,7 +607,7 @@ The resource requested by a client, when the experimental HTTP server is in use.
The HTTP code returned to the client, when the experimental HTTP server is in use.
@item short_seek_size
Set the threshold, in bytes, for when a readahead should be preferred over a seek and
Set the threshold, in bytes, for when a readahead should be prefered over a seek and
new HTTP request. This is useful, for example, to make sure the same connection
is used for reading large video packets with small audio packets in between.
@@ -1002,11 +986,6 @@ Set TCP_NODELAY to disable Nagle's algorithm. Default value is 0.
@emph{Remark: Writing to the socket is currently not optimized to minimize system calls and reduces the efficiency / effect of TCP_NODELAY.}
@item tcp_keepalive=@var{1|0}
Enable the TCP keepalive mechanism to detect dead peers and help maintain long-lived idle connections. Default value is 0.
Only the basic keepalive option (SO_KEEPALIVE) can be enabled or disabled. Platform-specific tuning parameters such as TCP_KEEPIDLE, TCP_KEEPINTVL, or TCP_KEEPCNT are not configurable and will use the operating system's default values.
@end table
For example to read with @command{ffplay} a multimedia resource named
@@ -1171,20 +1150,11 @@ ffplay "rtmp://myserver/live/mystream live=1"
Real-time Transport Protocol.
The required syntax for an RTP URL is:
@example
rtp://@var{hostname}[:@var{port}][?@var{options}]
@end example
rtp://@var{hostname}[:@var{port}][?@var{option}=@var{val}...]
@var{port} specifies the RTP port to use.
@var{options} contains a list of &-separated options of the form
@var{key}=@var{val}. Standard percent-encoding (and using the plus sign for
space) can be used to escape keys and values.
Options can also can be specified via command line options (or in code via
@code{AVOption}s).
The list of supported options follows.
The following URL options are supported:
@table @option
@@ -1194,12 +1164,10 @@ Set the TTL (Time-To-Live) value (for multicast only).
@item rtcpport=@var{n}
Set the remote RTCP port to @var{n}.
@item localport, local_rtpport, localrtpport=@var{n}
@item localrtpport=@var{n}
Set the local RTP port to @var{n}.
Using the localport option name is deprecated and should not be used.
@item local_rtcpport, localrtcpport=@var{n}'
@item localrtcpport=@var{n}'
Set the local RTCP port to @var{n}.
@item pkt_size=@var{n}
@@ -1222,12 +1190,19 @@ List disallowed (blocked) source IP addresses.
Send packets to the source address of the latest received packet (if
set to 1) or to a default remote address (if set to 0).
@item localport=@var{n}
Set the local RTP port to @var{n}.
@item localaddr=@var{addr}
Local IP address of a network interface used for sending packets or joining
multicast groups.
@item timeout=@var{n}
Set timeout (in microseconds) of socket I/O operations to @var{n}.
This is a deprecated option. Instead, @option{localrtpport} should be
used.
@end table
Important notes:
@@ -1555,15 +1530,7 @@ The accepted URL syntax is:
sctp://@var{host}:@var{port}[?@var{options}]
@end example
@var{options} contains a list of &-separated options of the form
@var{key}=@var{val}. Standard percent-encoding (and using the plus sign for
space) can be used to escape keys and values.
Options can also can be specified via command line options (or in code via
@code{AVOption}s).
The list of supported options follows.
The protocol accepts the following options:
@table @option
@item listen
If set to any value, listen for an incoming connection. Outgoing connection is done by default.
@@ -1572,67 +1539,6 @@ 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.
@@ -1643,13 +1549,18 @@ srt://@var{hostname}:@var{port}[?@var{options}]
@end example
@var{options} contains a list of &-separated options of the form
@var{key}=@var{val}. Standard percent-encoding (and using the plus sign for
space) can be used to escape keys and values.
@var{key}=@var{val}.
Options can also can be specified via command line options (or in code via
@code{AVOption}s).
or
The list of supported options follows.
@example
@var{options} srt://@var{hostname}:@var{port}
@end example
@var{options} contains a list of '-@var{key} @var{val}'
options.
This protocol accepts the following options.
@table @option
@item connect_timeout=@var{milliseconds}
@@ -1921,10 +1832,6 @@ 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}.
@@ -2006,11 +1913,7 @@ tcp://@var{hostname}:@var{port}[?@var{options}]
@end example
@var{options} contains a list of &-separated options of the form
@var{key}=@var{val}. Standard percent-encoding (and using the plus sign for
space) can be used to escape keys and values.
Options can also can be specified via command line options (or in code via
@code{AVOption}s).
@var{key}=@var{val}.
The list of supported options follows.
@@ -2065,14 +1968,8 @@ The required syntax for a TLS/SSL url is:
tls://@var{hostname}:@var{port}[?@var{options}]
@end example
@var{options} contains a list of &-separated options of the form
@var{key}=@var{val}. Standard percent-encoding (and using the plus sign for
space) can be used to escape keys and values.
Options can also can be specified via command line options (or in code via
@code{AVOption}s).
The list of supported options follows.
The following parameters can be set via command line options
(or in code via @code{AVOption}s):
@table @option
@@ -2083,7 +1980,7 @@ need to be specified for verification to work, but not all libraries and
setups have defaults built in.
The file must be in OpenSSL PEM format.
@item tls_verify, verify=@var{1|0}
@item tls_verify=@var{1|0}
If enabled, try to verify the peer that we are communicating with.
Note, if using OpenSSL, this currently only makes sure that the
peer certificate is signed by one of the root certificates in the CA
@@ -2091,8 +1988,8 @@ 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.
@@ -2127,94 +2024,6 @@ To play back a stream from the TLS/SSL server using @command{ffplay}:
ffplay tls://@var{hostname}:@var{port}
@end example
@section dtls
Datagram Transport Layer Security (DTLS)
The required syntax for a DTLS URL is:
@example
dtls://@var{hostname}:@var{port}[?@var{options}]
@end example
@var{options} contains a list of &-separated options of the form
@var{key}=@var{val}. Standard percent-encoding (and using the plus sign for
space) can be used to escape keys and values.
Options can also can be specified via command line options (or in code via
@code{AVOption}s).
DTLS shares most options with TLS, but operates over UDP instead of TCP.
The list of supported options follows.
@table @option
@item ca_file, cafile=@var{filename}
A file containing certificate authority (CA) root certificates to treat
as trusted. If the linked TLS library contains a default this might not
need to be specified for verification to work, but not all libraries and
setups have defaults built in.
The file must be in OpenSSL PEM format.
@item tls_verify, verify=@var{1|0}
If enabled, try to verify the peer that we are communicating with.
Note, if using OpenSSL, this currently only makes sure that the
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.
@item cert_file, cert=@var{filename}
A file containing a certificate to use in the handshake with the peer.
(When operating as server, in listen mode, this is more often required
by the peer, while client certificates only are mandated in certain
setups.)
@item key_file, key=@var{filename}
A file containing the private key for the certificate.
@item cert_pem=@var{string}
Certificate PEM string
@item key_pem=@var{string}
Private key PEM string
@item listen=@var{1|0}
If enabled, listen for connections on the provided port, and assume
the server role in the handshake instead of the client role.
@item mtu=@var{size}
Set the Maximum Transmission Unit (MTU) for DTLS packets.
@item use_srtp=@var{1|0}
Enable the use_srtp DTLS extension.
This is used in WebRTC applications to establish SRTP encryption keys
through the DTLS handshake. Default is disabled.
@item external_sock=@var{1|0}
Use an external socket instead of creating a new one.
This option only makes sense to pass when interacting with the code via
API, enabling this from CLI will cause immediate failure.
Default is disabled.
@end table
Example command lines:
To create a DTLS server:
@example
ffmpeg -listen 1 -i dtls://@var{hostname}:@var{port} @var{output}
@end example
To create a DTLS client and send data to server:
@example
ffmpeg -i @var{input} -f @var{format} dtls://@var{hostname}:@var{port}
@end example
@section udp
User Datagram Protocol.
@@ -2224,12 +2033,7 @@ The required syntax for an UDP URL is:
udp://@var{hostname}:@var{port}[?@var{options}]
@end example
@var{options} contains a list of &-separated options of the form
@var{key}=@var{val}. Standard percent-encoding (and using the plus sign for
space) can be used to escape keys and values.
Options can also can be specified via command line options (or in code via
@code{AVOption}s).
@var{options} contains a list of &-separated options of the form @var{key}=@var{val}.
In case threading is enabled on the system, a circular buffer is used
to store the incoming data, which allows one to reduce loss of data due to
@@ -2268,9 +2072,6 @@ Explicitly allow or disallow reusing UDP sockets.
@item ttl=@var{ttl}
Set the time to live value (for multicast only).
@item dscp=@var{dscp}
Set the 6-bit DSCP field for outgoing packets.
@item connect=@var{1|0}
Initialize the UDP socket with @code{connect()}. In this case, the
destination address can't be changed with ff_udp_set_remote_url later.
@@ -2352,17 +2153,6 @@ The following parameters can be set via command line options
Timeout in ms.
@item listen
Create the Unix socket in listening mode.
@item type
Select socket type.
@table @samp
@item stream Corresponds to SOCK_STREAM (reliable stream-oriented)
@item datagram Corresponds to SOCK_DGRAM (unreliable packet-oriented)
@item seqpacket Corresponds to SOCK_SEQPACKET (reliable packet-oriented)
@end table
@item pkt_size
Maximum packet size for packet-oriented sockets (SOCK_DGRAM and
SOCK_SEQPACKET). If greater than zero, this value is used as
@code{max_packet_size}. Ignored for SOCK_STREAM. Default is @code{0}.
@end table
@section zmq
+13 -101
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@@ -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.
@@ -108,10 +68,6 @@ Select full chroma input.
@item bitexact
Enable bitexact output.
@item unstable
Allow the use of experimental new code. May subtly affect the output or even
produce wrong results. For testing only.
@end table
@item srcw @var{(API only)}
@@ -140,9 +96,6 @@ If value is set to @code{1}, indicates source is full range. Default value is
If value is set to @code{1}, enable full range for destination. Default value
is @code{0}, which enables limited range.
@item gamma @var{(boolean)}
If value is set to @code{1}, enable gamma correct scaling. Default value is @code{0}.
@anchor{sws_params}
@item param0, param1
Set scaling algorithm parameters. The specified values are specific of
@@ -191,47 +144,6 @@ 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
+1 -1
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-345
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@@ -1,345 +0,0 @@
New swscale design to change everything (tm)
============================================
SwsGraph
--------
The entry point to the new architecture, SwsGraph is what coordinates
multiple "passes". These can include cascaded scaling passes, error diffusion
dithering, and so on. Or we could have separate passes for the vertical and
horizontal scaling. In between each SwsPass lies a fully allocated image buffer.
Graph passes may have different levels of threading, e.g. we can have a single
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.
From the point of view of SwsGraph itself, all inputs are progressive.
SwsOp / SwsOpList
-----------------
This is the newly introduced abstraction layer between the high-level format
handling logic and the low-level backing implementation. Each SwsOp is designed
to be as small and atomic as possible, with the possible exception of the
read / write operations due to their numerous variants.
The basic idea is to split logic between three major components:
1. The high-level format "business logic", which generates in a very
naive way a sequence of operations guaranteed to get you from point A
to point B. This logic is written with correctness in mind only, and
ignoring any performance concerns or low-level implementation decisions.
Semantically, everything is always decoded from the input format to
normalized (real valued) RGB, and then encoded back to output format.
This code lives in libswscale/format.c
2. The optimizer. This is where the "magic" happens, so to speak. The
optimizer's job is to take the abstract sequence of operations
produced by the high-level format analysis code and incrementally
optimize it. Each optimization step is designed to be minute and provably
lossless, or otherwise guarded behind the BITEXACT flag. This ensures that
the resulting output is always identical, no matter how many layers of
optimization we add.
This code lives in libswscale/ops.c
3. The compiler. Once we have a sequence of operations as output by the
optimizer, we "compile" this down to a callable function. This is then
applied by the dispatch wrapper by striping it over the input image.
See libswscale/ops_backend.c for the reference backend, or
libswscale/x86/ops.c for a more complex SIMD example.
This overall approach has a considerable number of benefits:
1. It allows us to verify correctness of logic and spot semantic errors at a
very high level, by simply looking at the sequence of operations (available
by default at debug / verbose log level), without having to dig through the
multiple levels of complicated, interwoven format handling code that is
legacy swscale.
2. Because most of the brains lives inside the powerful optimizer, we get
fast paths "for free" for any suitable format conversion, rather than having
to enumerate them one by one. SIMD code itself can be written in a very
general way and does need to be tied to specific pixel formats - subsequent
low-level implementations can be strung together without much overhead.
3. We can in the future, with relative ease, compile these operations
down to SPIR-V (or even LLVM IR) and generate efficient GPU or
target-machine specific implementations. This also opens the window for
adding hardware frame support to libswscale, and even transparently using
GPU acceleration for CPU frames.
4. Platform-specific SIMD can be reduced down to a comparatively small set of
optimized routines, while still providing 100% coverage for all possible
pixel formats and operations. (As of writing, the x86 example backend has
about 60 unique implementations, of which 20 are trivial swizzles, 10 are
read/write ops, 10 are pixel type conversions and the remaining 20 are the
various logic/arithmetic ops).
5. Backends hide behind a layer of abstraction offering them a considerable
deal of flexibility in how they want to implement their operations. For
example, the x86 backend has a dedicated function for compiling compatible
operations down to a single in-place pshufb instruction.
Platform specific low level data is self-contained within its own setup()
function and private data structure, eliminating all reads into SwsContext
or the possibility of conflicts between platforms.
6. We can compute an exact reference result for each operation with fixed
precision (ff_sws_op_apply_q), and use that to e.g. measure the amount of
error introduced by dithering, or even catch bugs in the reference C
implementation. (In theory - currently checkasm just compares against C)
Examples of SwsOp in action
---------------------------
For illustration, here is the sequence of operations currently generated by
my prototype, for a conversion from RGB24 to YUV444P:
Unoptimized operation list:
[ u8 .... -> ....] SWS_OP_READ : 3 elem(s) packed >> 0
[ u8 .... -> ....] SWS_OP_SWIZZLE : 0123
[ u8 .... -> ....] SWS_OP_RSHIFT : >> 0
[ u8 .... -> ....] SWS_OP_CLEAR : {_ _ _ 0}
[ u8 .... -> ....] SWS_OP_CONVERT : u8 -> f32
[f32 .... -> ....] SWS_OP_LINEAR : diag3+alpha [[1/255 0 0 0 0] [0 1/255 0 0 0] [0 0 1/255 0 0] [0 0 0 1 1]]
[f32 .... -> ....] SWS_OP_LINEAR : matrix3 [[0.299000 0.587000 0.114000 0 0] [-0.168736 -0.331264 1/2 0 0] [1/2 -0.418688 -57/701 0 0] [0 0 0 1 0]]
[f32 .... -> ....] SWS_OP_LINEAR : diag3+off3 [[219 0 0 0 16] [0 224 0 0 128] [0 0 224 0 128] [0 0 0 1 0]]
[f32 .... -> ....] SWS_OP_DITHER : 16x16 matrix
[f32 .... -> ....] SWS_OP_MAX : {0 0 0 0} <= x
[f32 .... -> ....] SWS_OP_MIN : x <= {255 255 255 _}
[f32 .... -> ....] SWS_OP_CONVERT : f32 -> u8
[ u8 .... -> ....] SWS_OP_LSHIFT : << 0
[ u8 .... -> ....] SWS_OP_SWIZZLE : 0123
[ u8 .... -> ....] SWS_OP_WRITE : 3 elem(s) planar >> 0
This is optimized into the following sequence:
Optimized operation list:
[ u8 XXXX -> +++X] SWS_OP_READ : 3 elem(s) packed >> 0
[ u8 ...X -> +++X] SWS_OP_CONVERT : u8 -> f32
[f32 ...X -> ...X] SWS_OP_LINEAR : matrix3+off3 [[0.256788 0.504129 0.097906 0 16] [-0.148223 -0.290993 112/255 0 128] [112/255 -0.367788 -0.071427 0 128] [0 0 0 1 0]]
[f32 ...X -> ...X] SWS_OP_DITHER : 16x16 matrix
[f32 ...X -> +++X] SWS_OP_CONVERT : f32 -> u8
[ u8 ...X -> +++X] SWS_OP_WRITE : 3 elem(s) planar >> 0
(X = unused, + = exact, 0 = zero)
The extra metadata on the left of the operation list is just a dump of the
internal state used by the optimizer during optimization. It keeps track of
knowledge about the pixel values, such as their value range, whether or not
they're exact integers, and so on.
In this example, you can see that the input values are exact (except for
the alpha channel, which is undefined), until the first SWS_OP_LINEAR
multiplies them by a noninteger constant. They regain their exact integer
status only after the (truncating) conversion to U8 in the output step.
Example of more aggressive optimization
---------------------------------------
Conversion pass for gray -> rgb48:
Unoptimized operation list:
[ u8 .... -> ....] SWS_OP_READ : 1 elem(s) planar >> 0
[ u8 .... -> ....] SWS_OP_SWIZZLE : 0123
[ u8 .... -> ....] SWS_OP_RSHIFT : >> 0
[ u8 .... -> ....] SWS_OP_CLEAR : {_ 0 0 0}
[ u8 .... -> ....] SWS_OP_CONVERT : u8 -> f32
[f32 .... -> ....] SWS_OP_LINEAR : luma+alpha [[1/255 0 0 0 0] [0 1 0 0 0] [0 0 1 0 0] [0 0 0 1 1]]
[f32 .... -> ....] SWS_OP_LINEAR : matrix3 [[1 0 701/500 0 0] [1 -0.344136 -0.714136 0 0] [1 443/250 0 0 0] [0 0 0 1 0]]
[f32 .... -> ....] SWS_OP_LINEAR : diag3 [[65535 0 0 0 0] [0 65535 0 0 0] [0 0 65535 0 0] [0 0 0 1 0]]
[f32 .... -> ....] SWS_OP_MAX : {0 0 0 0} <= x
[f32 .... -> ....] SWS_OP_MIN : x <= {65535 65535 65535 _}
[f32 .... -> ....] SWS_OP_CONVERT : f32 -> u16
[u16 .... -> ....] SWS_OP_LSHIFT : << 0
[u16 .... -> ....] SWS_OP_SWIZZLE : 0123
[u16 .... -> ....] SWS_OP_WRITE : 3 elem(s) packed >> 0
Optimized operation list:
[ u8 XXXX -> +XXX] SWS_OP_READ : 1 elem(s) planar >> 0
[ u8 .XXX -> +XXX] SWS_OP_CONVERT : u8 -> u16 (expand)
[u16 .XXX -> +++X] SWS_OP_SWIZZLE : 0003
[u16 ...X -> +++X] SWS_OP_WRITE : 3 elem(s) packed >> 0
(X = unused, + = exact, 0 = zero)
Here, the optimizer has managed to eliminate all of the unnecessary linear
operations on previously zero'd values, turn the resulting column matrix into
a swizzle operation, avoid the unnecessary dither (and round trip via float)
because the pixel values are guaranteed to be bit exact, and finally, turns
the multiplication by 65535 / 255 = 257 into a simple integer expand operation.
As a final bonus, the x86 backend further optimizes this into a 12-byte shuffle:
pshufb = {0, 0, 0, 0, 0, 0, 1, 1, 1, 1, 1, 1, -1, -1, -1, -1}
time=208 us, ref=4212 us, speedup=20.236x faster (single thread)
time=57 us, ref=472 us, speedup=8.160x faster (multi thread)
Compiler and underlying implementation layer (SwsOpChain)
---------------------------------------------------------
While the backend API is flexible enough to permit more exotic implementations
(e.g. using JIT code generation), we establish a common set of helpers for use
in "traditional" SIMD implementations.
The basic idea is to have one "kernel" (or implementation) per operation,
and then just chain a list of these kernels together as separate function
calls. For best performance, we want to keep data in vector registers in
between function calls using a custom calling convention, thus avoiding any
unnecessary memory accesses. Additionally, we want the per-kernel overhead to
be as low as possible, with each kernel ideally just jumping directly into
the next kernel.
As a result, we arrive at a design where we first divide the image into small
chunks, or "blocks", and then dispatch the "chain" of kernels on each chunk in
sequence. Each kernel processes a fixed number of pixels, with the overall
entry point taking care of looping. Remaining pixels (the "tail") are handled
generically by the backend-invariant dispatch code (located in ops.c), using a
partial memcpy into a suitably sized temporary buffer.
To minimize the per-kernel function call overhead, we use a "continuation
passing style" for chaining kernels. Each operation computes its result and
then directly calls the next operation in the sequence, with the appropriate
internal function signature.
The C reference backend reads data into the stack and then passes the array
pointers to the next continuation as regular function arguments:
void process(GlobalContext *ctx, OpContext *op,
block_t x, block_t y, block_t z, block_t w)
{
for (int i = 0; i < SWS_BLOCK_SIZE; i++)
// do something with x[i], y[i], z[i], w[i]
op->next(ctx, &op[1], x, y, z, w);
}
With type conversions pushing the new data onto the stack as well:
void convert8to16(GlobalContext *ctx, OpContext *op,
block_t x, block_t y, block_t z, block_t w)
{
/* Pseudo-code */
u16block_t x16 = (u16block_t) x;
u16block_t y16 = (u16block_t) y;
u16block_t z16 = (u16block_t) z;
u16block_t w16 = (u16block_t) w;
op->next(ctx, &op[1], x16, y16, z16, w16);
}
By contrast, the x86 backend always keeps the X/Y/Z/W values pinned in specific
vector registers (ymm0-ymm3 for the lower half, and ymm4-ymm7 for the second
half).
Each kernel additionally has access to a 32 byte per-op context storing the
pointer to the next kernel plus 16 bytes of arbitrary private data. This is
used during construction of the function chain to place things like small
constants.
In assembly, the per-kernel overhead looks like this:
load $tmp, $arg1
...
add $arg1, 32
jump $tmp
This design gives vastly better performance than the alternative of returning
out to a central loop or "trampoline". This is partly because the order of
kernels within a chain is always the same, so the branch predictor can easily
remember the target address of each "jump" instruction.
The only way to realistically improve on this design would be to directly
stitch the kernel body together using runtime code generation.
Future considerations and limitations
-------------------------------------
My current prototype has a number of severe limitations and opportunities
for improvements:
1. It does not handle scaling at all. I am not yet entirely sure on how I want
to handle scaling; this includes handling of subsampled content. I have a
number of vague ideas in my head, but nothing where I can say with certainty
that it will work out well.
It's possible that we won't come up with a perfect solution here, and will
need to decide on which set of compromises we are comfortable accepting:
1. Do we need the ability to scale YUV -> YUV by handling luma and chroma
independently? When downscaling 100x100 4:2:0 to 50x50 4:4:4, should we
support the option of reusing the chroma plane directly (even though
this would introduce a subpixel shift for typical chroma siting)?
Looking towards zimg, I am also thinking that we probably also want to do
scaling on floating point values, since this is best for both performance
and accuracy, especially given that we need to go up to 32-bit intermediates
during scaling anyway.
So far, the most promising approach seems to be to handle subsampled
input/output as a dedicated read/write operation type; perhaps even with a
fixed/static subsampling kernel. To avoid compromising on performance when
chroma resampling is not necessary, the optimizer could then relax the
pipeline to use non-interpolating read/writes when all intermediate
operations are component-independent.
2. Since each operation is conceptually defined on 4-component pixels, we end
up defining a lot of variants of each implementation for each possible
*subset*. For example, we have four different implementations for
SWS_OP_SCALE in my current templates:
- op_scale_1000
- op_scale_1001
- op_scale_1110
- op_scale_1111
This reflects the four different arrangements of pixel components that are
typically present (or absent). While best for performance, it does turn into
a bit of a chore when implementing these kernels.
The only real alternative would be to either branch inside the kernel (bad),
or to use separate kernels for each individual component and chain them all
together. I have not yet tested whether the latter approach would be faster
after the latest round of refactors to the kernel glue code.
3. I do not yet have any support for LUTs. But when I add them, something we
could do is have the optimized pass automatically "promote" a sequence of
operations to LUTs. For example, any sequence that looks like:
1. [u8] SWS_OP_CONVERT -> X
2. [X] ... // only per-component operations
4. [X] SWS_OP_CONVERT -> Y
3. [Y] SWS_OP_WRITE
could be replaced by a LUT with 256 entries. This is especially important
for anything involving packed 8-bit input (e.g. rgb8, rgb4_byte).
We also definitely want to hook this up to the existing CMS code for
transformations between different primaries.
4. Because we rely on AVRational math to generate the coefficients for
operations, we need to be able to represent all pixel values as an
AVRational. However, this presents a challenge for 32-bit formats (e.g.
GRAY32, RGBA128), because their size exceeds INT_MAX, which is the maximum
value representable by an AVRational.
It's possible we may want to introduce an AVRational64 for this, or
perhaps more flexibly, extend AVRational to an AVFloating type which is
represented as { AVRational n; int exp; }, representing n/d * 2^exp. This
would preserve our ability to represent all pixel values exactly, while
opening up the range arbitrarily.
5. Is there ever a situation where the use of floats introduces the risk of
non bit-exact output? For this reason, and possible performance advantages,
we may want to explore the use of a fixed-point 16 bit path as an alternative
to the floating point math.
So far, I have managed to avoid any bit exactness issues inside the x86
backend by ensuring that the order of linear operations is identical
between the C backend and the x86 backend, but this may not be practical
to guarantee on all backends. The x86 float code is also dramatically
faster than the old fixed point code, so I'm tentatively optimistic about
the lack of a need for a fixed point path.
+82 -151
View File
@@ -54,25 +54,12 @@ sub get_formatting_function($$) {
}
# determine texinfo version
my $package_version = ff_get_conf('PACKAGE_VERSION');
$package_version =~ s/\+nc$//;
$package_version =~ s/\+?dev$//;
my $program_version_num = version->declare($package_version)->numify;
my $program_version_num = version->declare(ff_get_conf('PACKAGE_VERSION'))->numify;
my $program_version_6_8 = $program_version_num >= 6.008000;
# no navigation elements
ff_set_from_init_file('HEADERS', 0);
my %sectioning_commands = %Texinfo::Common::sectioning_commands;
if (scalar(keys(%sectioning_commands)) == 0) {
%sectioning_commands = %Texinfo::Commands::sectioning_heading_commands;
}
my %root_commands = %Texinfo::Common::root_commands;
if (scalar(keys(%root_commands)) == 0) {
%root_commands = %Texinfo::Commands::root_commands;
}
sub ffmpeg_heading_command($$$$$)
{
my $self = shift;
@@ -90,9 +77,6 @@ sub ffmpeg_heading_command($$$$$)
return $result;
}
# no need to set it as the $element_id is output unconditionally
my $heading_id;
my $element_id = $self->command_id($command);
$result .= "<a name=\"$element_id\"></a>\n"
if (defined($element_id) and $element_id ne '');
@@ -100,99 +84,86 @@ sub ffmpeg_heading_command($$$$$)
print STDERR "Process $command "
.Texinfo::Structuring::_print_root_command_texi($command)."\n"
if ($self->get_conf('DEBUG'));
my $output_unit;
if ($root_commands{$command->{'cmdname'}}) {
if ($command->{'associated_unit'}) {
$output_unit = $command->{'associated_unit'};
} elsif ($command->{'structure'}
and $command->{'structure'}->{'associated_unit'}) {
$output_unit = $command->{'structure'}->{'associated_unit'};
} elsif ($command->{'parent'}
and $command->{'parent'}->{'type'}
and $command->{'parent'}->{'type'} eq 'element') {
$output_unit = $command->{'parent'};
}
my $element;
if ($Texinfo::Common::root_commands{$command->{'cmdname'}}
and $command->{'parent'}
and $command->{'parent'}->{'type'}
and $command->{'parent'}->{'type'} eq 'element') {
$element = $command->{'parent'};
}
if ($output_unit) {
if ($element) {
$result .= &{get_formatting_function($self, 'format_element_header')}($self, $cmdname,
$command, $output_unit);
$command, $element);
}
my $heading_level;
# Never use node for heading
if ($cmdname ne 'node') {
if (defined($command->{'extra'})
and defined($command->{'extra'}->{'section_level'})) {
$heading_level = $command->{'extra'}->{'section_level'};
} elsif ($command->{'structure'}
and defined($command->{'structure'}->{'section_level'})) {
$heading_level = $command->{'structure'}->{'section_level'};
} else {
$heading_level = $command->{'level'};
# node is used as heading if there is nothing else.
if ($cmdname eq 'node') {
if (!$element or (!$element->{'extra'}->{'section'}
and $element->{'extra'}->{'node'}
and $element->{'extra'}->{'node'} eq $command
# 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 {
$heading_level = $command->{'level'};
}
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 ($Texinfo::Common::root_commands{$cmdname}
and $Texinfo::Common::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);
}
if ($self->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];
}
# format_heading_text expects an array of headings for texinfo >= 7.0
if ($program_version_num >= 7.000000) {
$heading = [$heading];
}
$result .= &{get_formatting_function($self,'format_heading_text')}(
$self, $cmdname, $heading,
$heading_level +
$self->get_conf('CHAPTER_HEADER_LEVEL') - 1, $command);
}
}
$result .= $content if (defined($content));
return $result;
}
foreach my $command (keys(%sectioning_commands), 'node') {
foreach my $command (keys(%Texinfo::Common::sectioning_commands), 'node') {
texinfo_register_command_formatting($command, \&ffmpeg_heading_command);
}
@@ -217,56 +188,28 @@ sub ffmpeg_begin_file($$$)
my $filename = shift;
my $element = shift;
my ($element_command, $node_command, $command_for_title);
if ($element) {
if ($element->{'unit_command'}) {
$element_command = $element->{'unit_command'};
} elsif ($self->can('tree_unit_element_command')) {
$element_command = $self->tree_unit_element_command($element);
} elsif ($self->can('tree_unit_element_command')) {
$element_command = $self->element_command($element);
}
$node_command = $element_command;
if ($element_command and $element_command->{'cmdname'}
and $element_command->{'cmdname'} ne 'node'
and $element_command->{'extra'}
and $element_command->{'extra'}->{'associated_node'}) {
$node_command = $element_command->{'extra'}->{'associated_node'};
}
$command_for_title = $element_command if ($self->get_conf('SPLIT'));
my $command;
if ($element and $self->get_conf('SPLIT')) {
$command = $self->element_command($element);
}
my ($title, $description, $keywords, $encoding, $date, $css_lines, $doctype,
$root_html_element_attributes, $body_attributes, $copying_comment,
$after_body_open, $extra_head, $program_and_version, $program_homepage,
my ($title, $description, $encoding, $date, $css_lines,
$doctype, $bodytext, $copying_comment, $after_body_open,
$extra_head, $program_and_version, $program_homepage,
$program, $generator);
if ($program_version_num >= 7.001090) {
($title, $description, $keywords, $encoding, $date, $css_lines, $doctype,
$root_html_element_attributes, $body_attributes, $copying_comment,
$after_body_open, $extra_head, $program_and_version, $program_homepage,
$program, $generator) = $self->_file_header_information($command_for_title,
$filename);
} elsif ($program_version_num >= 7.000000) {
($title, $description, $encoding, $date, $css_lines, $doctype,
$root_html_element_attributes, $copying_comment, $after_body_open,
if ($program_version_num >= 7.000000) {
($title, $description, $encoding, $date, $css_lines,
$doctype, $bodytext, $copying_comment, $after_body_open,
$extra_head, $program_and_version, $program_homepage,
$program, $generator) = $self->_file_header_information($command_for_title,
$filename);
$program, $generator) = $self->_file_header_information($command);
} else {
($title, $description, $encoding, $date, $css_lines,
$doctype, $root_html_element_attributes, $copying_comment,
$after_body_open, $extra_head, $program_and_version, $program_homepage,
$program, $generator) = $self->_file_header_informations($command_for_title);
$doctype, $bodytext, $copying_comment, $after_body_open,
$extra_head, $program_and_version, $program_homepage,
$program, $generator) = $self->_file_header_informations($command);
}
my $links;
if ($program_version_num >= 7.000000) {
$links = $self->_get_links($filename, $element, $node_command);
} else {
$links = $self->_get_links ($filename, $element);
}
my $links = $self->_get_links ($filename, $element);
my $head1 = $ENV{"FFMPEG_HEADER1"} || <<EOT;
<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN" "http://www.w3.org/TR/html4/loose.dtd">
@@ -309,25 +252,13 @@ sub ffmpeg_program_string($)
if (defined($self->get_conf('PROGRAM'))
and $self->get_conf('PROGRAM') ne ''
and defined($self->get_conf('PACKAGE_URL'))) {
if ($program_version_num >= 7.001090) {
return $self->convert_tree(
$self->cdt('This document was generated using @uref{{program_homepage}, @emph{{program}}}.',
{ 'program_homepage' => {'text' => $self->get_conf('PACKAGE_URL')},
'program' => {'text' => $self->get_conf('PROGRAM') }}));
} else {
return $self->convert_tree(
return $self->convert_tree(
$self->gdt('This document was generated using @uref{{program_homepage}, @emph{{program}}}.',
{ 'program_homepage' => {'text' => $self->get_conf('PACKAGE_URL')},
'program' => {'text' => $self->get_conf('PROGRAM') }}));
}
{ 'program_homepage' => $self->get_conf('PACKAGE_URL'),
'program' => $self->get_conf('PROGRAM') }));
} else {
if ($program_version_num >= 7.001090) {
return $self->convert_tree(
$self->cdt('This document was generated automatically.'));
} else {
return $self->convert_tree(
$self->gdt('This document was generated automatically.'));
}
return $self->convert_tree(
$self->gdt('This document was generated automatically.'));
}
}
if ($program_version_6_8) {
Executable → Regular
View File
Executable → Regular
View File
+9 -17
View File
@@ -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:
+1
View File
@@ -44,3 +44,4 @@ a+b*c;
here the reader knows that a,b,c are meant to be signed integers but for C
standard compliance / to avoid undefined behavior they are stored in unsigned
ints.
-4
View File
@@ -731,12 +731,8 @@ FL+FR+FC+LFE+BL+BR+SL+SR+TFL+TFR+TBL+TBR
FL+FR+FC+LFE+BL+BR+SL+SR+TFL+TFR+TBC+LFE2
@item 9.1.4
FL+FR+FC+LFE+BL+BR+FLC+FRC+SL+SR+TFL+TFR+TBL+TBR
@item 9.1.6
FL+FR+FC+LFE+BL+BR+FLC+FRC+SL+SR+TFL+TFR+TBL+TBR+TSL+TSR
@item hexadecagonal
FL+FR+FC+BL+BR+BC+SL+SR+WL+WR+TBL+TBR+TBC+TFC+TFL+TFR
@item binaural
BIL+BIR
@item downmix
DL+DR
@item 22.2
+1 -6
View File
@@ -3,10 +3,6 @@ OBJS-$(HAVE_ARMV6) += $(ARMV6-OBJS) $(ARMV6-OBJS-yes)
OBJS-$(HAVE_ARMV8) += $(ARMV8-OBJS) $(ARMV8-OBJS-yes)
OBJS-$(HAVE_VFP) += $(VFP-OBJS) $(VFP-OBJS-yes)
OBJS-$(HAVE_NEON) += $(NEON-OBJS) $(NEON-OBJS-yes)
OBJS-$(HAVE_SVE) += $(SVE-OBJS) $(SVE-OBJS-yes)
OBJS-$(HAVE_SVE2) += $(SVE2-OBJS) $(SVE2-OBJS-yes)
OBJS-$(HAVE_SME) += $(SME-OBJS) $(SME-OBJS-yes)
OBJS-$(HAVE_SME2) += $(SME2-OBJS) $(SME2-OBJS-yes)
OBJS-$(HAVE_MIPSFPU) += $(MIPSFPU-OBJS) $(MIPSFPU-OBJS-yes)
OBJS-$(HAVE_MIPSDSP) += $(MIPSDSP-OBJS) $(MIPSDSP-OBJS-yes)
@@ -23,6 +19,5 @@ OBJS-$(HAVE_RV) += $(RV-OBJS) $(RV-OBJS-yes)
OBJS-$(HAVE_RVV) += $(RVV-OBJS) $(RVV-OBJS-yes)
OBJS-$(HAVE_RV_ZVBB) += $(RVVB-OBJS) $(RVVB-OBJS-yes)
OBJS-$(HAVE_SIMD128) += $(SIMD128-OBJS) $(SIMD128-OBJS-yes)
OBJS-$(HAVE_MMX) += $(MMX-OBJS) $(MMX-OBJS-yes)
OBJS-$(HAVE_X86ASM) += $(X86ASM-OBJS) $(X86ASM-OBJS-yes)
+2 -6
View File
@@ -38,10 +38,8 @@ int main(int argc, char **argv)
return -1;
output = fopen(argv[2], "wb");
if (!output) {
fclose(input);
if (!output)
return -1;
}
if (argc == 4) {
name = argv[3];
@@ -69,10 +67,8 @@ int main(int argc, char **argv)
fclose(output);
if (ferror(input) || !feof(input)) {
fclose(input);
if (ferror(input) || !feof(input))
return -1;
}
fclose(input);
+33 -75
View File
@@ -2,32 +2,31 @@
# common bits used by all libraries
#
DEFAULT_X86ASMD=.dbg
ifeq ($(DBG),1)
X86ASMD=$(DEFAULT_X86ASMD)
else
X86ASMD=
endif
ifndef SUBDIR
LINK = $(LD) $(1)
ifeq ($(LD),$(CC))
ifneq ($(CXX),)
LDXX := $(CXX)
LINK = $(if $(filter -lstdc++,$(1)),$(LDXX) $(filter-out -lstdc++,$(1)),$(LD) $(1))
endif
endif
BIN2CEXE = ffbuild/bin2c$(HOSTEXESUF)
BIN2C = $(BIN2CEXE)
ifndef V
Q = @
ECHO = printf "$(1)\t%s\n" $(2)
BRIEF = CC CXX OBJCC HOSTCC HOSTLD AS X86ASM AR LD LDXX STRIP CP WINDRES GLSLC NVCC BIN2C METALCC METALLIB
SILENT = DEPCC DEPCXX DEPHOSTCC DEPAS DEPX86ASM RANLIB RM
BRIEF = CC CXX OBJCC HOSTCC HOSTLD AS X86ASM AR LD STRIP CP WINDRES NVCC BIN2C
SILENT = DEPCC DEPHOSTCC DEPAS DEPX86ASM RANLIB RM
MSG = $@
M = @$(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.
@@ -43,7 +42,7 @@ OBJCCFLAGS = $(CPPFLAGS) $(CFLAGS) $(OBJCFLAGS)
ASFLAGS := $(CPPFLAGS) $(ASFLAGS)
# Use PREPEND here so that later (target-dependent) additions to CPPFLAGS
# end up in CXXFLAGS.
$(call PREPEND,CXXFLAGS, CPPFLAGS)
$(call PREPEND,CXXFLAGS, CPPFLAGS CFLAGS)
X86ASMFLAGS += $(IFLAGS:%=%/) -I$(<D)/ -Pconfig.asm
HOSTCCFLAGS = $(IFLAGS) $(HOSTCPPFLAGS) $(HOSTCFLAGS)
@@ -60,7 +59,6 @@ COMPILE_S = $(call COMPILE,AS)
COMPILE_M = $(call COMPILE,OBJCC)
COMPILE_X86ASM = $(call COMPILE,X86ASM)
COMPILE_HOSTC = $(call COMPILE,HOSTCC)
COMPILE_GLSLC = $(call COMPILE,GLSLC)
COMPILE_NVCC = $(call COMPILE,NVCC)
COMPILE_MMI = $(call COMPILE,CC,MMIFLAGS)
COMPILE_MSA = $(call COMPILE,CC,MSAFLAGS)
@@ -97,6 +95,10 @@ 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;/^$$/d;' > $@
%.o: %.asm
$(COMPILE_X86ASM)
-$(if $(ASMSTRIPFLAGS), $(STRIP) $(ASMSTRIPFLAGS) $@)
@@ -113,26 +115,6 @@ COMPILE_LASX = $(call COMPILE,CC,LASXFLAGS)
$(BIN2CEXE): ffbuild/bin2c_host.o
$(HOSTLD) $(HOSTLDFLAGS) $(HOSTLD_O) $^ $(HOSTEXTRALIBS)
RUN_BIN2C = $(BIN2C) $(patsubst $(SRC_PATH)/%,$(SRC_LINK)/%,$<) $@ $(subst .,_,$(basename $(notdir $@)))
RUN_GZIP = $(M)mkdir -p $(dir $@) && gzip -nc9 $(patsubst $(SRC_PATH)/%,$(SRC_LINK)/%,$<) >$@
RUN_MINIFY = $(M)mkdir -p $(dir $@) && sed 's!/\\*.*\\*/!!g' $< | tr '\n' ' ' | tr -s ' ' | sed 's/^ //; s/ $$//' > $@
%.gz: TAG = GZIP
%.min: TAG = MINIFY
%.spv: %.glsl
$(COMPILE_GLSLC)
ifdef CONFIG_SHADER_COMPRESSION
%.spv.gz: %.spv
$(RUN_GZIP)
%.spv.c: %.spv.gz $(BIN2CEXE)
$(RUN_BIN2C)
else
%.spv.c: %.spv $(BIN2CEXE)
$(RUN_BIN2C)
endif
%.metal.air: %.metal
$(METALCC) $< -o $@
@@ -140,52 +122,27 @@ endif
$(METALLIB) --split-module-without-linking $< -o $@
%.metallib.c: %.metallib $(BIN2CEXE)
$(RUN_BIN2C)
$(BIN2C) $< $@ $(subst .,_,$(basename $(notdir $@)))
%.ptx: %.cu $(SRC_PATH)/compat/cuda/cuda_runtime.h
$(COMPILE_NVCC)
ifdef CONFIG_SHADER_COMPRESSION
ifdef CONFIG_PTX_COMPRESSION
%.ptx.gz: TAG = GZIP
%.ptx.gz: %.ptx
$(RUN_GZIP)
$(M)gzip -nc9 $(patsubst $(SRC_PATH)/%,$(SRC_LINK)/%,$<) >$@
%.ptx.c: %.ptx.gz $(BIN2CEXE)
$(RUN_BIN2C)
$(BIN2C) $(patsubst $(SRC_PATH)/%,$(SRC_LINK)/%,$<) $@ $(subst .,_,$(basename $(notdir $@)))
else
%.ptx.c: %.ptx $(BIN2CEXE)
$(RUN_BIN2C)
endif
%.css.min: %.css
$(RUN_MINIFY)
ifdef CONFIG_RESOURCE_COMPRESSION
%.css.min.gz: %.css.min
$(RUN_GZIP)
%.css.c: %.css.min.gz $(BIN2CEXE)
$(RUN_BIN2C)
%.html.gz: %.html
$(RUN_GZIP)
%.html.c: %.html.gz $(BIN2CEXE)
$(RUN_BIN2C)
else # NO COMPRESSION
%.css.c: %.css.min $(BIN2CEXE)
$(RUN_BIN2C)
%.html.c: %.html $(BIN2CEXE)
$(RUN_BIN2C)
$(BIN2C) $(patsubst $(SRC_PATH)/%,$(SRC_LINK)/%,$<) $@ $(subst .,_,$(basename $(notdir $@)))
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:
@@ -202,20 +159,20 @@ endif
include $(SRC_PATH)/ffbuild/arch.mak
OBJS += $(OBJS-yes)
SLIBOBJS += $(SLIBOBJS-yes)
SHLIBOBJS += $(SHLIBOBJS-yes)
STLIBOBJS += $(STLIBOBJS-yes)
FFLIBS := $($(NAME)_FFLIBS) $(FFLIBS-yes) $(FFLIBS)
DEVPROGS += $(DEVPROGS-yes)
TESTPROGS += $(TESTPROGS-yes)
LDLIBS = $(FFLIBS:%=%$(BUILDSUF))
FFEXTRALIBS := $(LDLIBS:%=$(LD_LIB)) $(foreach lib,EXTRALIBS-$(NAME) $(FFLIBS:%=EXTRALIBS-%),$($(lib))) $(EXTRALIBS)
OBJS := $(sort $(OBJS:%=$(SUBDIR)%))
SLIBOBJS := $(sort $(SLIBOBJS:%=$(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))
@@ -233,11 +190,11 @@ ALLHEADERS := $(subst $(SRC_DIR)/,$(SUBDIR),$(wildcard $(SRC_DIR)/*.h $(SRC_DIR)
SKIPHEADERS += $(ARCH_HEADERS:%=$(ARCH)/%) $(SKIPHEADERS-)
SKIPHEADERS := $(SKIPHEADERS:%=$(SUBDIR)%)
HOBJS = $(filter-out $(SKIPHEADERS:.h=.h.o),$(ALLHEADERS:.h=.h.o))
SPVOBJS = $(filter %.spv.o,$(OBJS))
PTXOBJS = $(filter %.ptx.o,$(OBJS))
$(HOBJS): CCFLAGS += $(CFLAGS_HEADERS)
checkheaders: $(HOBJS)
.SECONDARY: $(HOBJS:.o=.c) $(SPVOBJS:.o=.c) $(SPVOBJS:.o=.gz) $(SPVOBJS:.o=) $(PTXOBJS:.o=.c) $(PTXOBJS:.o=.gz) $(PTXOBJS:.o=)
.SECONDARY: $(HOBJS:.o=.c) $(PTXOBJS:.o=.c) $(PTXOBJS:.o=.gz) $(PTXOBJS:.o=)
alltools: $(TOOLS)
$(HOSTOBJS): %.o: %.c
@@ -249,21 +206,22 @@ $(HOSTPROGS): %$(HOSTEXESUF): %.o
$(OBJS): | $(sort $(dir $(OBJS)))
$(HOBJS): | $(sort $(dir $(HOBJS)))
$(HOSTOBJS): | $(sort $(dir $(HOSTOBJS)))
$(SLIBOBJS): | $(sort $(dir $(SLIBOBJS)))
$(SHLIBOBJS): | $(sort $(dir $(SHLIBOBJS)))
$(STLIBOBJS): | $(sort $(dir $(STLIBOBJS)))
$(TESTOBJS): | $(sort $(dir $(TESTOBJS)))
$(TOOLOBJS): | tools
OUTDIRS := $(OUTDIRS) $(dir $(OBJS) $(HOBJS) $(HOSTOBJS) $(SHLIBOBJS) $(STLIBOBJS) $(TESTOBJS))
OUTDIRS := $(OUTDIRS) $(dir $(OBJS) $(HOBJS) $(HOSTOBJS) $(SLIBOBJS) $(SHLIBOBJS) $(STLIBOBJS) $(TESTOBJS))
CLEANSUFFIXES = *.d *.gcda *.gcno *.h.c *.ho *.map *.o *.objs *.pc *.ptx *.ptx.gz *.ptx.c *.spv *.spv.gz *.spv.c *.gen.asm *.gen.c *.gen.S *.ver *.version *.html.gz *.html.c *.css.min.gz *.css.min *.css.c *~ *.ilk *.pdb
CLEANSUFFIXES = *.d *.gcda *.gcno *.h.c *.ho *.map *.o *.pc *.ptx *.ptx.gz *.ptx.c *.ver *.version *$(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) $(SLIBOBJS:.o=.d)) $(OBJS:.o=$(DEFAULT_X86ASMD).d)

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