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>
I am not entirely sure the seek functionality is working correctly,
there are some strange cases like successful seek but no dts value.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
* qatar/master: (25 commits)
rtpenc: Add support for G726 audio
rtpdec: Interpret the different G726 names as bits_per_coded_sample
rtpenc: Change rtp_send_samples to handle sample sizes other than even bytes
rtpenc: Cast a rescaling parameter to int64_t
h264: cap max has_b_frames at MAX_DELAYED_PIC_COUNT - 1.
ARM: fix indentation in ff_dsputil_init_neon()
ARM: NEON put/avg_pixels8/16 cosmetics
ARM: add remaining NEON avg_pixels8/16 functions
ARM: clean up NEON put/avg_pixels macros
fate: split acodec-pcm into individual tests
swscale: #include "libavutil/mathematics.h"
pmpdec: don't use deprecated av_set_pts_info.
rv34: align temporary block of "dct" coefs
Add PlayStation Portable PMP format demuxer
proto: Realign struct initializers
proto: Use .priv_data_size to allocate the private context
mmsh: Properly clean up if the second ffurl_alloc failed
rtmp: Clean up properly if the handshake failed
md5proto: Remove the get_file_handle function
applehttpproto: Use the close function if the open function fails
...
Conflicts:
libavcodec/vble.c
libavformat/mmsh.c
libavformat/pmpdec.c
libavformat/udp.c
tests/ref/acodec/pcm
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master: (22 commits)
aacdec: Fix PS in ADTS.
avconv: Consistently use PIX_FMT_NONE.
dsputil: use cpuflags in x86 emu_edge_core
dsputil: use movups instead of movdqu in ff_emu_edge_core_sse()
wma: initialize prev_block_len_bits, next_block_len_bits, and block_len_bits.
mov: Remove some redundant and obsolete comments.
Add libavutil/mathematics.h #includes for INFINITY
doxy: structure libavformat groups
doxy: introduce an empty structure in libavcodec
doxy: provide a start page and document libavutil
doxy: cleanup pixfmt.h
regtest: split video encode/decode tests into individual targets
ARM: add explicit .arch and .fpu directives to asm.S
pthread: do not touch has_b_frames
avconv: cleanup the transcoding loop in output_packet().
avconv: split subtitle transcoding out of output_packet().
avconv: split video transcoding out of output_packet().
avconv: split audio transcoding out of output_packet().
avconv: reindent.
avconv: move streamcopy-only code out of decoding loop.
...
Conflicts:
avconv.c
libavcodec/aaccoder.c
libavcodec/pthread.c
libavcodec/version.h
libavutil/audioconvert.h
libavutil/avutil.h
libavutil/mem.h
tests/ref/vsynth1/dv
tests/ref/vsynth1/mpeg2thread
tests/ref/vsynth2/dv
tests/ref/vsynth2/mpeg2thread
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master: (35 commits)
libopencore-amr: check output buffer size before decoding
libopencore-amr: remove unneeded buf_size==0 check.
libopencore-amr: remove unneeded frame_count field.
aac_latm: remove unneeded check for zero-size packet.
pcmdec: fix output buffer size check by calculating the actual output size prior to decoding.
pcmdec: move codec-specific variable declarations to the corresponding codec blocks.
pcmdec: return buf_size instead of src-buf.
avcodec: remove the Zork PCM encoder.
pcm_zork: use AV_SAMPLE_FMT_U8 instead of shifting all samples by 8.
pcmenc: remove unneeded sample_fmt check.
pcmdec: move number of channels check to pcm_decode_init()
pcmdec: remove unnecessary check for sample_fmt change
pcmdec: move DVD PCM bits_per_coded_sample check near to the code that sets the sample size.
pcmdec: do not needlessly set *data_size to 0
alacdec: remove unneeded NULL or zero-size packet checks.
alacdec: simplify buffer allocation by using FF_ALLOC_OR_GOTO()
alacdec: ask for a sample for unsupported sample depths.
alacdec: cosmetics: use 'ch' instead of 'chan' to iterate channels
alacdec: move some declarations to the top of the function
alacdec: always use get_sbits_long() for uncompressed samples
...
Conflicts:
libavcodec/pcm.c
tests/ref/acodec/pcm
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
proresdsp: fix function prototypes.
prores-idct: fix overflow in c code.
fate: update prores-alpha ref after changing pix_fmt to yuv444p10le
prores: add missing feature warning for alpha
mov: 10l: Terminate string with 0 not '0'
mov: Prevent illegal writes when chapter titles are very short.
prores: add appropriate -fix_fmt parameter to FATE command
riff: always generate a proper WAVEFORMATEX structure in ff_put_wav_header
lavc: add a flag-based error_recognition field to AVCodecContext and deprecate non-flag-based ER field
lavc: rename deprecation symbol FF_API_VERY_AGGRESSIVE to FF_API_ER
Conflicts:
libavcodec/avcodec.h
libavformat/mov.c
tests/fate/prores.mak
tests/ref/acodec/g726
tests/ref/fate/prores-alpha
Merged-by: Michael Niedermayer <michaelni@gmx.at>
The cbSize field should be included in all cases, even with PCM where
its value is ignored.
Fixes encoding PCM audio in Matroska for some players which insist on
a full WAVEFORMATEX structure for A_MS/ACM audio.
Since fate uses wav files for the audio test a larger number of tests
has changed checksums or shifted positions due to the 2 byte longer
wave header.
Signed-off-by: Janne Grunau <janne-libav@jannau.net>
* qatar/master:
ac3enc: Add channel coupling support for the fixed-point AC-3 encoder.
ac3enc: scale floating-point coupling channel coefficients in scale_coefficients() rather than in apply_channel_coupling()
ac3enc: fix encoding of stereo ac3 files when rematrixing is disabled.
wavpack: fix wrong return value in wavpack_decode_block()
avconv: fix parsing metadata specifiers.
fate: use +frame+slice named constants instead of '3'
mpeg12: propagate more real return values through chunk decode error return and fix some indentation
wavpack: use context reset in appropriate places
avconv: move mux_preload and mux_max_delay to options context
avconv: move bitstream filters to options context.
avconv: move rate_emu to options context.
avconv: move max_frames to options context.
avconv: move metadata to options context.
avconv: move ts scale to options context.
avconv: move chapter maps to options context.
avconv: move metadata maps to options context.
avconv: move codec_names to options context.
Conflicts:
avconv.c
tests/fate-run.sh
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master: (28 commits)
mp3enc: write a xing frame containing number of frames in the file
lavf: update AVStream.nb_frames when muxing.
ffmpeg: remove unused variables from InputStream.
doc: update ffmpeg -ar and -ac documentation to reflect reality.
ffmpeg: remove pointless if (nb_input_files)
ffmpeg: merge input_files_ts_offset into input_files.
ffmpeg: merge input_codecs into input_streams.
ffmpeg: drop AV prefixes from struct names.
ffmpeg: deprecate loop_input and loop_output options
gif: add loop private option.
img2: add loop private option.
AVOptions: in av_opt_find() don't return named constants unless unit is specified.
x11grab: replace undocumented nomouse hackery with a private option.
dict: extend documentation.
lls: whitespace cosmetics
docs: Use proper markup for a literal command line option
docs: Remove a remark that isn't relevant any longer
docs: Explain how to regenerate import libraries with MSVC tools
docs: Mention that libraries for MSVC can be built with a cross compiler
docs: Remove old docs that mention setting up a build environment with lib.exe
...
Conflicts:
doc/ffmpeg.texi
doc/general.texi
ffmpeg.c
libavcodec/Makefile
libavcodec/dnxhddata.c
libavformat/mp3enc.c
libavformat/utils.c
libavutil/Makefile
tests/copycooker.sh
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This should fix behavior introduced by commit
96573c0d76. Av_rescale_rnd() is not
lossless so if two timestamps are equal after being rescaled they are
not always actually identical. This patch use av_compare_ts() to get
always a correct result.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
This updates the seek test reference to match de11ee9. Before this
change, most of the seeks requested positions before the supposed
start of the file (the preroll time), resulting in the first packet
being returned. With the preroll subtracted, some of these seeks
will land within the file and some beyond the end, thus returning
a different set of packets.
Signed-off-by: Mans Rullgard <mans@mansr.com>
This increases the accuracy of coefficients, leading to improved quality.
Rescaling of the coefficients to full 25-bit accuracy is done rather than
offsetting the exponent values. This requires coefficient scaling to be done
before determining the rematrixing strategy. Also, the rematrixing strategy
calculation must use 64-bit math to prevent overflow due to the higher
precision coefficients.
The rematrixing strategy reuse flags are not reset between frames, so they
need to be initialized for all blocks, not just block 0.
Signed-off-by: Mans Rullgard <mans@mansr.com>
This improves the audio quality significantly for stereo source with both the
fixed-point and floating-point AC-3 encoders.
Update acodec-ac3_fixed and seek-ac3_rm test references.
Originally committed as revision 26271 to svn://svn.ffmpeg.org/ffmpeg/trunk
This gives slightly better quality in PEAQ tests.
Code 3 gives a dBpb value of 2816 = -132dB (128 psd units = -6dB), which
corresponds to 22 bits. Since the exponents have an offset applied, the
16-bit source looks like 24-bit source to the bit allocation routine.
So using dBpb code=3 is a closer match to the exponent range.
Regression test refs updated for acodec-ac3, lavf-rm, and seek-ac3_rm.
Originally committed as revision 26144 to svn://svn.ffmpeg.org/ffmpeg/trunk
This avoids a 16-bit overflow in mdct512() due to a -32768 value in costab.
References updated for acodec-ac3, lavf-rm, and seek-ac3_rm tests.
Thanks to Måns Rullgård for finding the bug.
Originally committed as revision 26071 to svn://svn.ffmpeg.org/ffmpeg/trunk
Seek test reference updated because FLAC seeking now works properly.
Fixes roundup issue 1150.
Patch by Michael Chinen [mchinen at gmail]
Originally committed as revision 25914 to svn://svn.ffmpeg.org/ffmpeg/trunk
Fixes a scr issue reported with dvdauthor ([FFmpeg-user] FFMPEG encoded MPEG-2 video causes error in DVDAuthor)
Originally committed as revision 25512 to svn://svn.ffmpeg.org/ffmpeg/trunk
Start them on keyframes when reasonable, and delay writing audio packets
to help ensure that there's audio samples available for the first frame in
clusters.
Patch by James Zern <jzern at google>
Originally committed as revision 23473 to svn://svn.ffmpeg.org/ffmpeg/trunk
This isn't exactly semantically equivalent, but the field has already been
long abused to mean this, and writing it helps in determining a decent cfr
time base when transcoding from a mkv where the video codec stores none (VP8).
Originally committed as revision 23284 to svn://svn.ffmpeg.org/ffmpeg/trunk
This takes into account whether the granule defines the start or end times
of packets, and sets the correct file offset of the associated page.
Originally committed as revision 22462 to svn://svn.ffmpeg.org/ffmpeg/trunk
Otherwise it gets set automatically to a page midstream and prevents seeking
to the first page.
Originally committed as revision 22454 to svn://svn.ffmpeg.org/ffmpeg/trunk
Seeking on image sequences doesn't actually work, so this
test isn't very useful until that capability is added.
Originally committed as revision 22286 to svn://svn.ffmpeg.org/ffmpeg/trunk
