- 08 Mar, 2015 3 commits
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Luca Barbato authored
The valid returned values are always at most 11bit. Remove the previous check that assumed larger values plausible and use a signed integer to check get_vlc2 return values. CC: libav-stable@libav.org
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Luca Barbato authored
Makes the code a little simpler.
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Carl Eugen Hoyos authored
Use correct context, reduce log level, don't assume it is a video stream, and print the tag of the unknown stream. Signed-off-by: Vittorio Giovara <vittorio.giovara@gmail.com>
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- 07 Mar, 2015 2 commits
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Martin Storsjö authored
Signed-off-by: Martin Storsjö <martin@martin.st>
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Martin Storsjö authored
When all the codepaths using manually set .arch/.fpu code is behind runtime detection, the elf attributes should be suppressed. This allows tools to know that the final built binary doesn't strictly require these extensions. Signed-off-by: Martin Storsjö <martin@martin.st>
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- 06 Mar, 2015 5 commits
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Jonas Lindgren authored
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
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Martin Storsjö authored
a8765852 had the unintended side effect of returning AVERROR(ENOMEM) when track->entry is zero, while the code intentionally wants to continue in that case. Signed-off-by: Martin Storsjö <martin@martin.st>
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Timo Rothenpieler authored
The mov muxer already supports picking up extradata that wasn't present during the avformat_write_header call - we just need to propagate it. Since the dash muxer uses delay_moov, we have time up until the first segment is written to get extradata filled in. Also update the codec description string when the extradata becomes available. Signed-off-by: Martin Storsjö <martin@martin.st>
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Timo Rothenpieler authored
Signed-off-by: Martin Storsjö <martin@martin.st>
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Martin Storsjö authored
Signed-off-by: Martin Storsjö <martin@martin.st>
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- 05 Mar, 2015 7 commits
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Himangi Saraogi authored
Signed-off-by: Diego Biurrun <diego@biurrun.de>
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Anton Khirnov authored
It was only relevant when data partitioning was in use.
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Andreas Cadhalpun authored
It is used in adx_read_packet, which currently depends on the decoder/parser setting this value between reading the file header and demuxing the first packet. Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com> Signed-off-by: Anton Khirnov <anton@khirnov.net>
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Andreas Cadhalpun authored
The chunk size is limited to UINT16_MAX (written by avio_wb16), so make sure that the packet size is not too large. Such large frames need to be split into slices smaller than 64 kB, but that is currently supported neither by the rv10/rv20 encoders nor the rm muxer. Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com> Signed-off-by: Anton Khirnov <anton@khirnov.net>
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Andreas Cadhalpun authored
According to the WebP Lossless Bitstream Specification the highest allowed value for a prefix code is 39. If prefix_code is too large, the calculated extra_bits has an invalid value and triggers an assertion in get_bits. Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com> Signed-off-by: Anton Khirnov <anton@khirnov.net>
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Andreas Cadhalpun authored
If it doesn't fit into 12 bits it triggers an assertion. Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com> Signed-off-by: Anton Khirnov <anton@khirnov.net>
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Thomas Volkert authored
The code was tested with live555 server. Signed-off-by: Martin Storsjö <martin@martin.st>
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- 04 Mar, 2015 10 commits
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Himangi Saraogi authored
Signed-off-by: Diego Biurrun <diego@biurrun.de>
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Martin Storsjö authored
The original flags variable contains rtpdec flags, while the rmflags variable contains RM flag bits which have a completely different definition. Signed-off-by: Martin Storsjö <martin@martin.st>
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Martin Storsjö authored
The only case where RTP_FLAG_KEY actually is needed is in RDT, where such a flag needs to be passed via the rtpdec parse function's flags parameter. Signed-off-by: Martin Storsjö <martin@martin.st>
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Martin Storsjö authored
Nothing in the framework nor in the rest of the depacketizer actually uses this flag - the chained demuxer sets the keyframe flag properly on demuxed packets already. Signed-off-by: Martin Storsjö <martin@martin.st>
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Luca Barbato authored
And provide a more informative message in case of failure. CC: libav-stable@libav.org
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Vittorio Giovara authored
Although the specification mandates this bit to zero, it may happen that software tools incorrectly flip it to one, invalidating a possibly valid stream. Relax this restriction, by failing only when AV_EF_BITSTREAM is set. This behaviour is similar to aac decoders in Firefox and Quicktime. Signed-off-by: Vittorio Giovara <vittorio.giovara@gmail.com>
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Diego Biurrun authored
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Martin Storsjö authored
This was missed in bb4a310b. This fixes a double free on close. Signed-off-by: Martin Storsjö <martin@martin.st>
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Martin Storsjö authored
This fixes fate on all architectures that have an optimized but not bitexact idct available. Signed-off-by: Martin Storsjö <martin@martin.st>
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Martin Storsjö authored
This fixes an oversight in 96084251, in a refactoring done on top of Gilles' original patch. Pointed out by Gilles Chanteperdrix. Signed-off-by: Martin Storsjö <martin@martin.st>
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- 03 Mar, 2015 4 commits
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Anton Khirnov authored
It's required for at least AVCodecContext. Fixes make checkheaders.
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Vittorio Giovara authored
Similarly to what has been done for MOV, display XMP metadata only when users explicitly require it. The Extensible Metadata Platform tag can contain various kind of data which are not strictly related to the video file, such as history of edits and saves from the project file. Signed-off-by: Vittorio Giovara <vittorio.giovara@gmail.com>
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Vittorio Giovara authored
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Vittorio Giovara authored
Normally the aic decoder finds the proper slice combination (multiple of some number less than 32) but in case of odd width, it resorts to the default values, which were actually swapped. The number of slices is modified to account for such odd width cases. CC: libav-stable@libav.org
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- 02 Mar, 2015 5 commits
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Martin Storsjö authored
Signed-off-by: Martin Storsjö <martin@martin.st>
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Martin Storsjö authored
This reverts commit 4abfa387. This commit broke playback of fragmented mp4 files with b-frames. While investigating this, it turned out that the general framework isn't ready for a PTS-based index yet. Revert this change until a better thought out solution is in place. Signed-off-by: Martin Storsjö <martin@martin.st>
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Luca Barbato authored
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Luca Barbato authored
The encoder or decoder might be disabled but the format would be supported for at least remuxing.
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Luca Barbato authored
Some files produced by the official encoder have up to 16bit of padding instead of the expected padding to the byte. Use a self-explanatory macro instead of a simple number. CC: libav-stable@libav.org Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
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- 28 Feb, 2015 4 commits
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Martin Storsjö authored
Instead check the timestamps while muxing, to avoid buffering a too long timestamp range into one single packet. This makes the AMR and AAC packetization slightly less efficient, since we set a possibly unnecessarily high max_frames_per_packet. (These packetizers end up doing a memmove of the TOC bytes if sending a packet before max_frames_per_packet is achieved, and we end up setting max_frames_per_packet to a value that should be high enough for most uses.) All packetizers that use max_frames_per_packet now set it either to a default value, or to a value calculated based on other parameters, so none of them rely on the previous default setting. For iLBC, copy one frame at a time, to allow checking the timestamp range for each of them - basically doing potentially multiple loops to simplify the code instead of trying to calculate the number of frames to buffer while honoring s1->max_delay. This is in preparation for reducing the coupling between libavformat and libavcodec, by not having the muxers use the encoder field frame_size (which may not be available during e.g. stream copy). Signed-off-by: Martin Storsjö <martin@martin.st>
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Martin Storsjö authored
Factorize out the s->num_frames check at the start of the if statements, simplifying adding more alternative causes for sending the buffered frames. Signed-off-by: Martin Storsjö <martin@martin.st>
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Martin Storsjö authored
Signed-off-by: Martin Storsjö <martin@martin.st>
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Martin Storsjö authored
This avoids having to jump to the defaultcase in the switch. Manually override the stream time base back to 90 kHz for the few audio codecs that don't use the sample rate as time base (mp2, mp3). Signed-off-by: Martin Storsjö <martin@martin.st>
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