Commit Graph

114 Commits

Author SHA1 Message Date
Matthieu Bouron 83cab07a4c mxfdec: set audio packets pts
Also fix playback of ntsc files.

Reviewed-by: Tomas Härdin <tomas.hardin@codemill.se>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-11-16 14:12:37 +01:00
Peter Ross a373f35272 wtvenc: produce seekable files
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-11-07 13:59:03 +01:00
Peter Ross b50759cd6b wtvenc: do not emit stream2 and DSATTRIB_TRANSPORT_PROPERTY chunks; these are not required for playback
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-11-07 13:42:01 +01:00
Michael Niedermayer d8cfa98358 Merge commit '58b619c8a226cc4564ad5af291bc99a04f89ee56'
* commit '58b619c8a226cc4564ad5af291bc99a04f89ee56':
  wav muxer: write metadata

Conflicts:
	Changelog
	libavformat/version.h

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2012-10-17 15:16:23 +02:00
Michael Niedermayer c079da5073 Merge commit '0bca0283ccded5e32da143a462168ad1988a58fd'
* commit '0bca0283ccded5e32da143a462168ad1988a58fd':
  riff: do not write empty INFO tags

Conflicts:
	tests/ref/fate/vsynth1-cljr
	tests/ref/fate/vsynth1-ffvhuff
	tests/ref/fate/vsynth1-h261
	tests/ref/fate/vsynth1-h263
	tests/ref/fate/vsynth1-h263-obmc
	tests/ref/fate/vsynth1-h263p
	tests/ref/fate/vsynth1-huffyuv
	tests/ref/fate/vsynth1-jpegls
	tests/ref/fate/vsynth1-mjpeg
	tests/ref/fate/vsynth1-mpeg4-adap
	tests/ref/fate/vsynth1-mpeg4-adv
	tests/ref/fate/vsynth1-mpeg4-error
	tests/ref/fate/vsynth1-mpeg4-nr
	tests/ref/fate/vsynth1-mpeg4-qpel
	tests/ref/fate/vsynth1-mpeg4-qprd
	tests/ref/fate/vsynth1-mpeg4-rc
	tests/ref/fate/vsynth1-mpeg4-thread
	tests/ref/fate/vsynth1-msmpeg4
	tests/ref/fate/vsynth1-msmpeg4v2
	tests/ref/fate/vsynth1-rgb
	tests/ref/fate/vsynth1-wmv1
	tests/ref/fate/vsynth1-wmv2
	tests/ref/fate/vsynth1-yuv
	tests/ref/fate/vsynth2-cljr
	tests/ref/fate/vsynth2-ffvhuff
	tests/ref/fate/vsynth2-h261
	tests/ref/fate/vsynth2-h263
	tests/ref/fate/vsynth2-h263-obmc
	tests/ref/fate/vsynth2-h263p
	tests/ref/fate/vsynth2-huffyuv
	tests/ref/fate/vsynth2-jpegls
	tests/ref/fate/vsynth2-mjpeg
	tests/ref/fate/vsynth2-mpeg4-adap
	tests/ref/fate/vsynth2-mpeg4-error
	tests/ref/fate/vsynth2-mpeg4-nr
	tests/ref/fate/vsynth2-mpeg4-qpel
	tests/ref/fate/vsynth2-mpeg4-qprd
	tests/ref/fate/vsynth2-mpeg4-rc
	tests/ref/fate/vsynth2-mpeg4-thread
	tests/ref/fate/vsynth2-msmpeg4
	tests/ref/fate/vsynth2-msmpeg4v2
	tests/ref/fate/vsynth2-rgb
	tests/ref/fate/vsynth2-wmv1
	tests/ref/fate/vsynth2-wmv2
	tests/ref/fate/vsynth2-yuv
	tests/ref/lavf/avi
	tests/ref/seek/h261_avi
	tests/ref/seek/h263_avi
	tests/ref/seek/h263p_avi
	tests/ref/seek/lavf_avi
	tests/ref/seek/mjpeg_avi
	tests/ref/seek/mpeg4_adap_avi
	tests/ref/seek/mpeg4_error_avi
	tests/ref/seek/mpeg4_nr_avi
	tests/ref/seek/mpeg4_qpel_avi
	tests/ref/seek/mpeg4_qprd_avi
	tests/ref/seek/mpeg4_rc_avi
	tests/ref/seek/mpeg4_thread_avi
	tests/ref/seek/msmpeg4_avi
	tests/ref/seek/msmpeg4v2_avi
	tests/ref/seek/wmv1_avi
	tests/ref/seek/wmv2_avi

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2012-10-17 14:43:26 +02:00
Victor Vasiliev 0bca0283cc riff: do not write empty INFO tags
Signed-off-by: Anton Khirnov <anton@khirnov.net>
2012-10-16 18:51:16 +02:00
Clément Bœsch b08273c9ca lavf/mkv: avoid negative ts by default.
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."
2012-10-15 09:19:21 +02:00
Michael Niedermayer b02493e476 movenc: force video timebase to be 0.1ms precisse at least.
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>
2012-10-04 01:33:49 +02:00
Michael Niedermayer 4abc411b97 nutenc: choose for non audio streams a timebase with finer resolution.
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>
2012-10-02 01:09:12 +02:00
Michael Niedermayer e3fb5bc147 nut: store and read the r_frame_rate
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>
2012-10-02 01:09:12 +02:00
Michael Niedermayer 4eb0f5f635 nutenc: use 1/sample rate as timebase for audio instead of framesize/sample rate
This way audio frames can be exactly stored even when they are not
aligned with timestamp 0

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-10-02 01:09:12 +02:00
Michael Niedermayer b4c753487c asfenc: avoid negative timestamps
Fixes Ticket1606

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-09-28 02:40:53 +02:00
Michael Niedermayer 9e9b5159e9 mpegvideo_enc: reduce QMAT_SHIFT to avoid overflow in dnxhd
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-09-27 19:43:31 +02:00
Clément Bœsch 00e1afd83f fate: add faststart regression test.
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.
2012-09-27 08:59:37 +02:00
Michael Niedermayer 3a621c9d99 nutenc: Support writing an index
The seek test improves in accuracy
Fixes Ticket877

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-08-26 22:15:21 +02:00
Michael Niedermayer 2cd491a47c lavf: move generic index generation code to a later point
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
2012-07-26 03:04:49 +02:00
Michael Niedermayer bacbbd2b03 vocenc: fix sample rate rounding direction
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-06-25 03:30:12 +02:00
Michael Niedermayer b0387edd5e Merge commit 'f919cc7df6ab844bc12f89fe7bef4fb915a47725'
* 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>
2012-05-30 01:40:54 +02:00
Mans Rullgard 7263cd5544 fate: convert codec-regression.sh to makefile rules
Signed-off-by: Mans Rullgard <mans@mansr.com>
2012-05-29 08:35:41 +01:00
Michael Niedermayer de2cfb744a Merge remote-tracking branch 'qatar/master'
* 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>
2012-05-17 23:45:00 +02:00
Mans Rullgard 7d7b40f48a pcmenc: set correct bitrate value
This fixes a bogus bitrate value in the header of WAV files with
alaw/ulaw audio.

Signed-off-by: Mans Rullgard <mans@mansr.com>
2012-05-17 02:34:57 +01:00
Anton Khirnov fc49f22c3b ffmpeg: add support for audio filters.
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
2012-05-17 03:29:21 +02:00
Michael Niedermayer e8339302c0 fate: update ogg seektest after all the bug fixes
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-05-04 14:40:22 +02:00
Michael Niedermayer 29ec5c1102 fate: update mmf seek checksum, change caused by av_get_packet() useage
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-05-02 01:55:04 +02:00
Justin Ruggles c5671aeb77 FATE: avoid channel mixing in lavf-dv_fmt
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.
2012-04-24 15:55:45 -04:00
Michael Niedermayer 3bbf3f7e42 Merge remote-tracking branch 'qatar/master'
* 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>
2012-04-24 02:30:41 +02:00
Dale Curtis 8336eb6f85 matroska: Add incremental parsing of clusters.
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>
2012-04-22 17:23:50 -07:00
Michael Niedermayer 3194ab78a6 Merge remote-tracking branch 'qatar/master'
* 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>
2012-04-20 22:18:26 +02:00
Justin Ruggles acb1730218 FATE: allow lavf tests to alter input parameters
Change some lavf tests to avoid resampling and channel mixing.
2012-04-20 10:23:57 -04:00
Justin Ruggles 5052980400 FATE: replace the acodec-pcm_s24daud test with an enc_dec_pcm checksum test
This avoids resampling and channel mixing by using a source with
the correct channel layout and sample rate.
2012-04-20 10:23:57 -04:00
Justin Ruggles 03caef1bed FATE: replace the acodec-g726 test with 4 new encode/decode tests
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.
2012-04-20 10:23:57 -04:00
Michael Niedermayer 4778783160 Merge commit '3b266da3d35f3f7a61258b78384dfe920d875d29'
* 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>
2012-04-17 04:03:50 +02:00
Ramiro Polla bd603494f9 asfenc: properly write index information
The index must take into account the pre-roll time and must seek backwards,
not forwards.

Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
2012-04-05 06:03:10 +02:00
Michael Niedermayer 967facb695 Merge remote-tracking branch 'qatar/master'
* 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>
2012-03-22 00:40:11 +01:00
Justin Ruggles b0f75ba272 mpegaudioenc: use AVCodec.encode2()
Update FATE references due to encoder delay.
2012-03-20 18:56:22 -04:00
Justin Ruggles aa872af5e3 ac3enc: update to AVCodec.encode2()
Update FATE references due to encoder delay.
2012-03-20 18:46:56 -04:00
Michael Niedermayer 967bdb8572 Merge remote-tracking branch 'qatar/master'
* 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>
2012-03-17 23:16:05 +01:00
Justin Ruggles 85cf49fab7 FATE: remove WMA acodec tests 2012-03-17 11:46:15 -04:00
Wolfram Gloger f8353d5fda mpegvideo: don't pretend the first frame is always a key frame
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>
2012-03-16 21:27:23 +01:00
Michael Niedermayer ad53c7f9ec lavf: Add system to seperate relative timestamps from absolute ones.
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>
2012-03-09 19:36:12 +01:00
Michael Niedermayer f095391a14 Merge remote-tracking branch 'qatar/master'
* 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>
2012-03-06 06:03:32 +01:00
Justin Ruggles 8d1a20aa7c aiffdec: do not set AVCodecContext.frame_size
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.
2012-03-05 13:08:17 -05:00
Michael Niedermayer 15c6be8c7d Merge remote-tracking branch 'qatar/master'
* 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>
2012-03-04 04:26:04 +01:00
Justin Ruggles 0883109b27 voc/avs: Do not change the sample rate mid-stream.
Also, set the time base based on the sample rate.
lavf-voc seek test updated to reflect slightly different seek points.
2012-03-03 17:03:27 -05:00
Justin Ruggles 0b8b7db01b mpegaudio_parser: do not ignore information from the first parsed frame
Update some demuxing and seeking fate tests.
2012-03-03 17:03:26 -05:00
Anton Khirnov 87d7a92b62 rawdec: set timebase to 1/fps. 2012-02-26 07:30:21 +01:00
Reimar Döffinger 048cc80292 Update mkv seek tests.
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>
2012-02-13 21:48:20 +01:00
Michael Niedermayer cd1c12b5c5 Merge remote-tracking branch 'qatar/master'
* 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>
2012-02-12 01:06:13 +01:00
Pavel Koshevoy 277e52845e Modified to generate PAT/PMT for video keyframes
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>
2012-02-11 23:57:35 +01:00
Justin Ruggles b498867d66 FATE: update reference for seek-alac_mp4
This should have been updated in b590f3a7bf.
2012-02-11 16:41:01 -05:00