If 384k is too high for the samplerate, choose the closest
possible
Idea to increase the bitrate from: 46439e1562
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Extrapolate audio timestamps based on the number of samples demuxed.
Deal with some MXF nastiness involving fractional number of
samples per EditUnit when seeking (the specs handwave this away).
Further fixes from Tomas Härdin.
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
Fixes sync in some samples (e.g. bugs 7581 and 8374 in VLC).
Based on a commit by Matthieu Bouron <matthieu.bouron@gmail.com>
Reported-by: Jean-Baptiste Kempf <jb@videolan.org>
CC: libav-stable@libav.org
This may improve compatibility of lgpegs generated by libavcodec
also encoded ljpegs become slightly smaller
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Files won't validate with mkvalidtor if these two elements are missing.
Use a const "Lavf" string that wont change with library version bumps.
Signed-off-by: James Almer <jamrial@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
The element was only being written when the value == 1. But the default
value of this element is 1, so this has no useful effect. This element
needs to be written when the value == 0.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
The fate tests change as they used 1.2 previously
The increased size is due to:
32bit CRCs per slice by default (can be disabled),
it adds slice headers to allow decoding one slice without the others
an additional slice size field is added to make it possible to find
slices within corrupted surroundings.
these add up to about 57bit per slice more
at 50 frames and 4 slices thats 1425 byte
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
The bug it was working seems to have been fixed.
This change causes ffmpeg to use the trim filter to implement
the -t option.
FATE tests are updated due to the more accurate handling of
the last packets.
Tags must have at least one SimpleTag element to be spec conformant.
Updated lavf-mkv and seek-lavf-mkv FATE references as the tests were affected by
this.
Fixes ticket #2785
Signed-off-by: James Almer <jamrial@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This makes -t sample-accurate for audio and will allow further
simplication in the future.
Most of the FATE changes are due to audio now being sample accurate. In
some cases a video frame was incorrectly passed with the old code, while
its was over the limit.
Most formats do not support negative timestamps, shift them to avoid
unexpected behaviour and a number of bad crashes.
CC:libav-stable@libav.org
Signed-off-by: Anton Khirnov <anton@khirnov.net>
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
This commit removes the badly duplicated code between the encoder and
the muxer. That may sound surprising, but the encoder is now responsible
from the encoding of the picture when muxing to a .gif file. It also
does not require anymore a manual user intervention such as a -pix_fmt
rgb24 to work properly. To summarize, output gif are now easier to
generate, code is saner and simpler, and files are smaller (thanks to
the lzw encoding which was unused so far with the default .gif output).
We can certainly make things even better, but this is the first step.
FATE is updated because of the output being produced by the encoder and
not the muxer (no lzw in the muxer), and in the seek test only the size
mismatches.
Fixes Ticket #2262
* commit '3e2f200237af977b9253b0aff121eee27bcedb44':
roqvideodec: fix a potential infinite loop in roqvideo_decode_frame().
xxan: fix invalid memory access in xan_decode_frame_type0()
tty: set avg_frame_rate.
FATE: enable multiple slices in the ffv1 vsynth test
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Other software does not store it in this case, and the information
is provided by the codec stream
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
The QuickTime specification does not contain any hint that the atom
must not be written in some cases and both the QuickTime and the
AVID decoders do not fail if the atom is present.
This change allows to signal (visually) interlaced streams with
a codec different from uncompressed video.
As a side-effect, this fixes ticket #2202
* commit 'e6bc38fd49c94726b45d5d5cc2b756ad8ec49ee0':
wmv2: move IDCT to its own DSP context.
Conflicts:
libavcodec/dsputil.h
tests/ref/seek/vsynth2-wmv2
tests/ref/vsynth/vsynth1-wmv2
tests/ref/vsynth/vsynth2-wmv2
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This allows us to remove FF_IDCT_WMV2, which serves no practical purpose
other than to be able to select the WMV2 IDCT for MPEG (or vice versa)
and get corrupt output.
Fate tests for all wmv2-related tests change, because (for some obscure
reason) they forced use of the MPEG IDCT. You would get the same changes
previously by not using -idct simple in the fate test (or replacing it
with -idct auto).
Since 83cab07 audio stream time bases are based on SampleRate, not EditRate.
This fixes trac ticket #2029 and a few seeking issues.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* commit 'e816034a5fa131b13c4ad87bb0b5065b4f5697c6':
fate-seek: remove use of gnu make 3.82 only private modifier
fate: move vsynth reference files to their own directory
fate: move fate-acodec reference files to their own dir
configure: avplay now depends on avresample
fate: split dependencies for fate-seek tests
Conflicts:
configure
tests/fate/seek.mak
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Each fate-seek test depends now only on the corresponding fate-acodec,
fate-vsynth2 or fate-lavf test which creates the file seek-tests
operates on. The tests and references are renamed to match the test they
depend on.
Without this exception files with ".gif" extension by default
recognized as input suitable for image2 demuxer rather than gif.
In order to pass image through gif demuxer it was necessary
to use -f gif option.
This change affected 'make fate' test results because previously
image2 demuxer and gif decoder took only first frame of multiframe
test data, which is no longer true with gif demuxer.
Signed-off-by: Vitaliy E Sugrobov <vsugrob@hotmail.com>
This fixes playback in some circumstances (like webm in firefox).
Regression after 2c34367b.
It is also matching the Matroska specifications:
http://matroska.org/technical/specs/notes.html, "The quick eye will
notice that if a Cluster's Timecode is set to zero, it is possible to
have Blocks with a negative Raw Timecode. Blocks with a negative Raw
Timecode are not valid."
The timebases before where only guranteed to be 1/fps precisse
and could cause AV sync errors on low fps
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
While a 25 fps stream can in general store frame durations in 1/25
units, this is not true for the timestamps. For example a 25fps
and a 25000/1001 fps stream when they are stored together might have
a matching 0 timestamp point but when for example a chapter from
this is cut the new start is no longer aligned. The issue gets
MUCH worse when the streams are lower fps, like 1 or 2 fps.
This commit thus makes the muxer choose a multiple of the
framerate as timebase that is at least about 20 micro seconds precise
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
With this, when we use a finer timebase than neccessary to store
durations the demuxer still knows what the original timebase was.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Also factorize the common options for the different mov-based tests.
Since the header is now on top in the last generated file, the data
offset in the seek test needed some updates as well.
By moving it to a later point relative and unknown timestamps
are more likely to have been corrected
similar patch reviewed-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Conflicts:
libavformat/utils.c
* commit 'f919cc7df6ab844bc12f89fe7bef4fb915a47725':
fate: fix acodec/vsynth tests for make 3.81
pcm_mpeg: fix number of consumed bytes to include the header.
avfilter: include required header file avfilter.h in video.h
x86: Avoid movs on BUTTERFLYPS when in AVX mode
x86: use new schema for ASM macros
fate: convert codec-regression.sh to makefile rules
fate: allow tests to specify unit size for psnr comparison
fate: teach videogen/rotozoom to output a single raw video stream
http: Add support for reusing the http socket for subsequent requests
http: Add support for using persistent connections
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
pcmenc: set correct bitrate value
avprobe: don't print format entry name when only one was requested
Conflicts:
ffprobe.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Some of the FATE changes are due to off-by-one different rounding being used
(lrintf vs av_rescale_q).
Some fate changes are due to 1 audio frame less being encoded (the new variant seems
matching what qatar does and according to ffprobe its closer to the requested duration)
the mapchan feature sadly is lost in this commit because it depends on resampling
being done in ffmpeg.c which is now moved completely into the av filter layer
-async is broken after this commit, this will be fixed in subsequent commits
the new filter reconfiguration system is flawed and will drop a frame on each
parameter change which is why the nelly moser checksums need updating.
Conflicts:
ffmpeg.c
tests/ref/fate/smjpeg
This partially reverts acb1730218
which would only have needed to change the checksums if channel mixing had
been properly avoided. This changes the output file size reference and the
seek test reference back to the previous values.
* qatar/master:
matroska: Clear prev_pkt between seeks.
avutil: change default buffer size alignment for sample buffer functions
audemux: Add a sanity check for the number of channels
Remove libdirac decoder.
matroska: Add incremental parsing of clusters.
avconv: fix off by one check in complex_filter
mpegts: Try seeking back even for nonseekable protocols
swscale: K&R formatting cosmetics (part III)
Conflicts:
configure
doc/general.texi
doc/platform.texi
ffmpeg.c
libavcodec/Makefile
libavcodec/allcodecs.c
libavcodec/libdirac.h
libavcodec/libdiracdec.c
libavformat/au.c
libavformat/mpegts.c
libswscale/input.c
tests/ref/seek/lavf_mkv
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Reduces the amount of upfront data required for cluster parsing
thus decreasing latency on seek and startup.
The change in the seek-lavf_mkv FATE test is due to incremental
parsing no longer reading as much data as the old parser and
thus not having that additional data to generate index entries
based on keyframes. Index entries are added correctly as the
file is parsed.
All FATE tests pass and Chrome has been using this patch for ~6
months without issue.
Currently incremental parsing is not supported for files with
SSA tracks since they require merging packets between clusters.
In this case the code falls back to non-incremental parsing.
Signed-off-by: Aaron Colwell <acolwell@chromium.org>
Signed-off-by: Dale Curtis <dalecurtis@chromium.org>
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
* qatar/master:
avcodec: add a cook parser to get subpacket duration
FATE: allow lavf tests to alter input parameters
FATE: replace the acodec-pcm_s24daud test with an enc_dec_pcm checksum test
FATE: replace the acodec-g726 test with 4 new encode/decode tests
FATE: replace current g722 encoding tests with an encode/decode test
FATE: add a pattern rule for generating asynth wav files
FATE: optionally write a WAVE header in audiogen
avutil: add audio fifo buffer
Conflicts:
doc/APIchanges
libavcodec/version.h
libavutil/avutil.h
tests/Makefile
tests/codec-regression.sh
tests/fate/voice.mak
tests/lavf-regression.sh
tests/ref/acodec/g722
tests/ref/acodec/g726
tests/ref/acodec/pcm_s24daud
tests/ref/lavf/dv_fmt
tests/ref/lavf/gxf
tests/ref/lavf/mxf
tests/ref/lavf/mxf_d10
tests/ref/seek/lavf_dv
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Avoids resampling and channel mixing. This only tests the behavior
with respect to input and output audio rather than also testing changes
to the encoder or muxer that do not affect the resulting decoded output.
* commit '3b266da3d35f3f7a61258b78384dfe920d875d29':
avconv: add support for complex filtergraphs.
avconv: make filtergraphs global.
avconv: move filtered_frame from InputStream to OutputStream.
avconv: don't set output width/height directly from input value.
avconv: move resample_{width,height,pix_fmt} to InputStream.
avconv: remove a useless variable from OutputStream.
avconv: get output pixel format from lavfi.
graphparser: fix the order in which unlabeled input links are returned.
avconv: change {input,output}_{streams,files} into arrays of pointers.
avconv: don't pass input/output streams to some functions.
Conflicts:
cmdutils.c
cmdutils.h
doc/ffmpeg.texi
ffmpeg.c
ffplay.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master: (26 commits)
adxenc: use AVCodec.encode2()
adxenc: Use the AVFrame in ADXContext for coded_frame
indeo4: fix out-of-bounds function call.
configure: Restructure help output.
configure: Internal-only components should not be command-line selectable.
vorbisenc: use AVCodec.encode2()
libvorbis: use AVCodec.encode2()
libopencore-amrnbenc: use AVCodec.encode2()
ra144enc: use AVCodec.encode2()
nellymoserenc: use AVCodec.encode2()
roqaudioenc: use AVCodec.encode2()
libspeex: use AVCodec.encode2()
libvo_amrwbenc: use AVCodec.encode2()
libvo_aacenc: use AVCodec.encode2()
wmaenc: use AVCodec.encode2()
mpegaudioenc: use AVCodec.encode2()
libmp3lame: use AVCodec.encode2()
libgsmenc: use AVCodec.encode2()
libfaac: use AVCodec.encode2()
g726enc: use AVCodec.encode2()
...
Conflicts:
configure
libavcodec/Makefile
libavcodec/ac3enc.c
libavcodec/adxenc.c
libavcodec/libgsm.c
libavcodec/libvorbis.c
libavcodec/vorbisenc.c
libavcodec/wmaenc.c
tests/ref/acodec/g722
tests/ref/lavf/asf
tests/ref/lavf/ffm
tests/ref/lavf/mkv
tests/ref/lavf/mpg
tests/ref/lavf/rm
tests/ref/lavf/ts
tests/ref/seek/lavf_asf
tests/ref/seek/lavf_ffm
tests/ref/seek/lavf_mkv
tests/ref/seek/lavf_mpg
tests/ref/seek/lavf_rm
tests/ref/seek/lavf_ts
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
resample: allocate a large enough output buffer
fate: fix enc_dec_pcm tests with remote target
wmaenc: remove bit-exact hack
FATE: remove WMA acodec tests
FATE: add WMAv1 and WMAv2 encode/decode tests with fuzzy comparison
FATE: add AC-3 and E-AC-3 encode/decode tests with fuzzy comparison
qtrle: Use bytestream2 functions to prevent buffer overreads.
vqavideo: check malloc return values
x11grab: fix a memory leak exposed by valgrind
threads: fix old frames returned after avcodec_flush_buffers()
MPV: always mark dummy frames as reference
h264: fix deadlocks on incomplete reference frame decoding.
mpeg4: report frame decoding completion at ff_MPV_frame_end().
mimic: don't use self as reference, and report completion at end of decode().
Conflicts:
libavcodec/h264.c
libavcodec/qtrle.c
libavcodec/resample.c
libavcodec/vqavideo.c
libavdevice/x11grab.c
tests/ref/seek/wmav1_asf
tests/ref/seek/wmav2_asf
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Signed-off-by: Wolfram Gloger <wmglo@dent.med.uni-muenchen.de>
Modify the parser initialization so that parsers can
set pict_type themselves. Use this in the mpegvideo parser
so that initial frames are not unconditionally I frames.
I have had this in my tree for several years.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
With this we can always know if a timestamp is based on added durations
from an unknown origin or if it is based on a correct timestamp (and possibly
added durations)
This should fix some bugs where this distinction was mixed up.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master: (31 commits)
cdxl demux: do not create packets with uninitialized data at EOF.
Replace computations of remaining bits with calls to get_bits_left().
amrnb/amrwb: Remove get_bits usage.
cosmetics: reindent
avformat: do not require a pixel/sample format if there is no decoder
avformat: do not fill-in audio packet duration in compute_pkt_fields()
lavf: Use av_get_audio_frame_duration() in get_audio_frame_size()
dca_parser: parse the sample rate and frame durations
libspeexdec: do not set AVCodecContext.frame_size
libopencore-amr: do not set AVCodecContext.frame_size
alsdec: do not set AVCodecContext.frame_size
siff: do not set AVCodecContext.frame_size
amr demuxer: do not set AVCodecContext.frame_size.
aiffdec: do not set AVCodecContext.frame_size
mov: do not set AVCodecContext.frame_size
ape: do not set AVCodecContext.frame_size.
rdt: remove workaround for infinite loop with aac
avformat: do not require frame_size in avformat_find_stream_info() for CELT
avformat: do not require frame_size in avformat_find_stream_info() for MP1/2/3
avformat: do not require frame_size in avformat_find_stream_info() for AAC
...
Conflicts:
doc/APIchanges
libavcodec/Makefile
libavcodec/avcodec.h
libavcodec/h264.c
libavcodec/h264_ps.c
libavcodec/utils.c
libavcodec/version.h
libavcodec/x86/dsputil_mmx.c
libavformat/utils.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
It is unnecessary. Also, for some codecs we're reading more than 1 frame per
packet. Instead we use a private context variable to calculate the bit rate,
stream duration, and packet durations.
Updated FATE seek test, which has slightly different timestamps due to a
more accurate bit rate calculation.
* qatar/master:
tiertexseq: set correct block_align for audio
tiertexseq: set audio stream start time to 0
voc/avs: Do not change the sample rate mid-stream.
segafilm: use the sample rate as the time base for audio streams
ea: fix audio pts
psx-str: fix audio pts
vqf: set packet duration
tta demuxer: set packet duration
mpegaudio_parser: do not ignore information from the first parsed frame
mpegaudio_parser: be less picky about the start position
thp: set audio packet durations
avcodec: add a Vorbis parser to get packet duration
vorbisdec: read the previous window flag for long windows
lavc: free the output packet when encoding failed or produced no output.
lavc: preserve avpkt->destruct in ff_alloc_packet().
lavc: clarify the meaning of AVCodecContext.frame_number.
mpegts: Pad the packet buffer in handle_packet().
mpegts: Do not call read_sl_header() when no bytes remain in the buffer.
Conflicts:
libavcodec/mpegaudio_parser.c
libavcodec/version.h
libavformat/mpegts.c
tests/ref/fate/pva-demux
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Seek beyond the end will now directly return an error instead
of claiming to succeed and then return EOF immediately on next read.
This change is because before 47e015e6f1
mkv seek incorrectly never failed.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
* qatar/master:
FATE: update reference for seek-alac_mp4
sunrast: Return AVERROR values instead of -1.
sunrast: Add support for gray8 decoding.
swscale: enforce a minimum filtersize.
alacenc: use AVCodec.encode2()
alacenc: cosmetics: indentation
alacenc: consolidate bitstream writing into a single function.
alacenc: only encode frame size in header for a final smaller frame
alacenc: store current frame size in AlacEncodeContext.
alacenc: return AVERROR codes in alac_encode_frame()
alacenc: calculate a new max frame size for the final small frame
alacenc: pretty-printing and other cosmetics
alacenc: fix error handling and potential memleaks in alac_encode_init()
alacenc: do not set coded_frame->key_frame
alacenc: do not set bits_per_coded_sample
alacenc: remove unneeded frame_size check in alac_encode_frame()
tta: error out if samplerate is zero.
ttadec: fix invalid free when an error occurs while decoding 24-bit tta
wavpack: add needed braces for 2 statements inside an if block
Conflicts:
tests/ref/acodec/alac
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This is so that TS fragments produced by
http://code.google.com/p/httpsegmenter/
would be compatible with JW Player.
A new member variable prev_payload_key was added to MpegTSWriteStream
to help detect transition from non-key to key frame, so that
PAT/PMT would not be produced for every keyframe in intra-only videos.
Signed-off-by: Pavel Koshevoy <pkoshevoy@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Current code compares the desired recording time with InputStream.pts,
which has a very unclear meaning. Change the code to use actual
timestamps of the frames passed to the encoder.
In several tests, one less frame is encoded, which is more correct.
In the idroq test one more frame is encoded, which is again more
correct.
Behavior with stream copy should be unchanged.
* qatar/master:
libx264: fix indentation.
vorbis: fix overflows in floor1[] vector and inverse db table index.
win64: add a XMM clobber test configure option.
movdec: Parse the dvc1 atom
ARM: ac3: fix ac3_bit_alloc_calc_bap_armv6
swscale: K&R formatting cosmetics for Blackfin code
frwu: lowercase the FRWU codec name
movdec: fix dts generation in fragmented files
fate: make acodec-ac3_fixed test output raw AC3
APIchanges: add missing commit hashes
swscale: implement MMX, SSE2 and AVX functions for RGB32 input.
ra144enc: drop pointless "encoder" from .long_name
bethsoftvideo: fix palette reading.
mpc7: use av_fast_padded_malloc()
mpc7: simplify handling of packet sizes that are not a multiple of 4 bytes
doc: decoding Forward Uncompressed is supported
Fix a typo in the x86 asm version of ff_vector_clip_int32()
pcmenc: Do not set avpkt->size.
ff_alloc_packet: modify the size of the packet to match the requested size
Conflicts:
doc/APIchanges
libavcodec/libx264.c
libavcodec/mpc7.c
libavformat/isom.h
libswscale/Makefile
libswscale/bfin/yuv2rgb_bfin.c
tests/ref/fate/bethsoft-vid
tests/ref/seek/ac3_ac3
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This uses the old demuxing code for OP1a and separate demuxing code for OPAtom.
Timestamp output is added to the old demuxing code.
The seeking code is made to seek to the start of the desired EditUnit only,
from which the normal demuxing code takes over (if OP1a). This means we
do not use delta entries or slices, only StreamOffsets. OPAtom seeking
basically works like before.
This also makes D-10 seeking behave the same way as OP1a and OPAtom. In other
words, we allow seeking before the start or past the end for D-10 too.
Based on several patches by Tomas Härdin <tomas.hardin@codemill.se> and
Reimar Döffinger <Reimar.Doeffinger@gmx.de>.
Changed av_calloc to av_mallocz, added overflow checks.
This uses the old demuxing code for OP1a and separate demuxing code for OPAtom.
Timestamp output is added to the old demuxing code.
The seeking code is made to seek to the start of the desired EditUnit only,
from which the normal demuxing code takes over (if OP1a). This means we don't
use delta entries or slices, only StreamOffsets.
OPAtom seeking basically works like before.
This also makes D-10 seeking behave the same way as OP1a and OPAtom. In other
words, we allow seeking before the start or past the end for D-10 too.
This fixes ticket #746.
This changes mxf_compute_ptses() to be used for MXFIndexTable, and also adds
code for computing the fake index to it.
This also temporarily disables PTS computation. A future patch will restore it.
As far as I could see the only change is increased pos values,
which is as expected with additional metadata in the files.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
