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* cdxl demux: do not create packets with uninitialized data at EOF.Reimar Döffinger2012-03-05
| | | | | | Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de> Signed-off-by: Paul B Mahol <onemda@gmail.com> Signed-off-by: Justin Ruggles <justin.ruggles@gmail.com>
* cosmetics: reindentJustin Ruggles2012-03-05
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* avformat: do not require a pixel/sample format if there is no decoderJustin Ruggles2012-03-05
| | | | | | | | Also, do not keep trying to find and open a decoder in try_decode_frame() if we already tried and failed once. Fixes always searching until max_analyze_duration in avformat_find_stream_info() when demuxing codecs without a decoder.
* avformat: do not fill-in audio packet duration in compute_pkt_fields()Justin Ruggles2012-03-05
| | | | Use the estimated duration only to calculate missing timestamps if needed.
* lavf: Use av_get_audio_frame_duration() in get_audio_frame_size()Justin Ruggles2012-03-05
| | | | | | | | | | Also, do not give AVCodecContext.frame_size priority for muxing. Updated 2 FATE references: dxa-feeble - adds 1 audio frame that is still within 2 seconds as specified by -t 2 in the FATE test wmv8-drm-nodec - durations are not needed. previously they were estimated using the packet size and average bit rate.
* siff: do not set AVCodecContext.frame_sizeJustin Ruggles2012-03-05
| | | | | also, properly set AVCodecContext.bits_per_coded_sample, AVStreasm.start_time, and AVPacket.duration.
* amr demuxer: do not set AVCodecContext.frame_size.Justin Ruggles2012-03-05
| | | | it is not necessary.
* aiffdec: do not set AVCodecContext.frame_sizeJustin Ruggles2012-03-05
| | | | | | | | | 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.
* mov: do not set AVCodecContext.frame_sizeJustin Ruggles2012-03-05
| | | | It is not necessary.
* ape: do not set AVCodecContext.frame_size.Justin Ruggles2012-03-05
| | | | prevents lavf from setting incorrect packet durations.
* rdt: remove workaround for infinite loop with aacJustin Ruggles2012-03-05
| | | | avformat_find_stream_info() no longer hangs while waiting for AAC frame_size
* avformat: do not require frame_size in avformat_find_stream_info() for CELTJustin Ruggles2012-03-05
| | | | | In Ogg/CELT, frame_size is found in the same place as the sample_rate and channels, so we do not need to force the frame_size to be parsed.
* avformat: do not require frame_size in avformat_find_stream_info() for MP1/2/3Justin Ruggles2012-03-05
| | | | | It was only needed to avoid a bad time base (and thus non-monotone timestamps) for stream copy to avi.
* avformat: do not require frame_size in avformat_find_stream_info() for AACJustin Ruggles2012-03-05
| | | | We already will get the needed info because of CODEC_CAP_CHANNEL_CONF
* swfenc: use av_get_audio_frame_duration() instead of AVCodecContext.frame_sizeJustin Ruggles2012-03-05
| | | | | This way we can do stream copy without having the demuxer wait until frame_size has been set.
* rtpenc: use av_get_audio_frame_duration() for max_frames_per_packetJustin Ruggles2012-03-05
| | | | | It is more reliable than AVCodecContext.frame_size for codecs with constant packet duration.
* riffenc: use av_get_audio_frame_duration()Justin Ruggles2012-03-05
| | | | | | For encoding, frame_size is not a reliable indicator of packet duration. Also, we don't want to have to force the demuxer to find frame_size for stream copy to work.
* lavf: deobfuscate read_frame_internal().Anton Khirnov2012-03-05
| | | | | | | | | | | | | | Split off packet parsing into a separate function. Parse full packets at once and store them in a queue, eliminating the need for tracking parsing state in AVStream. The horrible unreadable loop in read_frame_internal() now isn't weirdly ordered and doesn't contain evil gotos, so it should be much easier to understand. compute_pkt_fields() now invents slightly different timestamps for two raw vc1 tests, due to has_b_frames being set a bit later. They shouldn't be more wrong (or right) than previous ones.
* lavf: make read_from_packet_buffer() more flexible.Anton Khirnov2012-03-05
| | | | | | | | Make packet buffer a parameter, don't hardcode it to be AVFormatContext.packet_buffer. Also move the function higher in the file, since it will be called from read_frame_internal().
* lavf: factorize freeing a packet buffer.Anton Khirnov2012-03-05
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* dv: Do not redundantly initialize struct members to zero.Diego Biurrun2012-03-05
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* tiertexseq: set correct block_align for audioJustin Ruggles2012-03-03
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* tiertexseq: set audio stream start time to 0Justin Ruggles2012-03-03
| | | | | Update FATE test to reflect delayed video due to the file having audio-only frames prior to the first frame with video.
* voc/avs: Do not change the sample rate mid-stream.Justin Ruggles2012-03-03
| | | | | Also, set the time base based on the sample rate. lavf-voc seek test updated to reflect slightly different seek points.
* segafilm: use the sample rate as the time base for audio streamsJustin Ruggles2012-03-03
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* ea: fix audio ptsJustin Ruggles2012-03-03
| | | | | | | The time base is 1 / sample_rate, not 90000. Several more codecs encode the sample count in the first 4 bytes of the chunk, so we set the durations accordingly. Also, we can set start_time and packet duration instead of keeping track of the sample count in the demuxer.
* psx-str: fix audio ptsJustin Ruggles2012-03-03
| | | | Each packet has 18 sectors with 224/channels samples in each sector.
* vqf: set packet durationJustin Ruggles2012-03-03
| | | | | | Fixes timestamp calculation. The FATE reference is updated because timestamp calculations are now more accurate. Previous timestamps were based on average bit rate.
* tta demuxer: set packet durationJustin Ruggles2012-03-03
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* thp: set audio packet durationsJustin Ruggles2012-03-03
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* avcodec: add a Vorbis parser to get packet durationJustin Ruggles2012-03-03
| | | | This also allows for removing some of the Vorbis-related hacks.
* mpegts: Pad the packet buffer in handle_packet().Alex Converse2012-03-02
| | | | | | | This allows it to be used with get_bits without the thread of overreads. Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind CC: libav-stable@libav.org
* mpegts: Do not call read_sl_header() when no bytes remain in the buffer.Alex Converse2012-03-02
| | | | | Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind CC: libav-stable@libav.org
* matroska: check buffer size for RM-style byte reordering.Ronald S. Bultje2012-03-02
| | | | | Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind CC: libav-stable@libav.org
* rmdec: Honor .RMF tag size rather than assuming 18.Alex Converse2012-03-02
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* r3d: don't set codec timebase.Anton Khirnov2012-03-02
| | | | | | It's not supposed to be set by demuxers. Set avg_frame_rate and r_frame_rate instead.
* electronicarts: set timebase for tgv video.Anton Khirnov2012-03-02
| | | | | | | | The container has no timestamps and the framerate isn't stored in the data either. The decoder sets codec timebase to experimentally found value 1/15. Do the same for the demuxer too, it should at least be better than the default 1/90000.
* electronicarts: parse the framerate for cmv video.Anton Khirnov2012-03-02
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* ogg: don't set codec timebaseAnton Khirnov2012-03-02
| | | | Demuxers are not supposed to set it.
* electronicarts: don't set codec timebaseAnton Khirnov2012-03-02
| | | | | | Demuxers are not supposed to set it. Set stream timebase and framerates instead (this is a cfr container with no timestamps).
* avs: don't set codec timebaseAnton Khirnov2012-03-02
| | | | | Demuxers are not supposed to set it. Set r_frame_rate and avg_frame_rate instead.
* mp3dec: Fix reading file size and frames in VBRI headersIngo Brückl2012-03-01
| | | | | | | | The fields "Number of Bytes" and "Number of Frames" are mixed up. "Bytes" come first, "Frames" behind. Signed-off-by: Michael Niedermayer <michaelni@gmx.at> Signed-off-by: Alex Converse <alex.converse@gmail.com>
* rmdec: adjust printf format string specifier to fix warningDiego Biurrun2012-03-01
| | | | libavformat/rmdec.c:383: warning: format ‘%d’ expects type ‘int’, but argument 7 has type ‘int64_t’
* rtpenc: Use MB info side data for splitting H263 packets for RFC 2190Martin Storsjö2012-03-01
| | | | | | | This makes the packetization spec compliant for cases where one single GOB doesn't fit into an RTP packet. Signed-off-by: Martin Storsjö <martin@martin.st>
* dxa: set audio stream time base using the sample rateJustin Ruggles2012-02-29
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* psx-str: do not allow seeking by bytesJustin Ruggles2012-02-29
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* asfdec: Do not set AVCodecContext.frame_sizeJustin Ruggles2012-02-29
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* vqf: set packet parameters after av_new_packet()Justin Ruggles2012-02-29
| | | | Otherwise the values are overwritten.
* rtpenc: Fix setting the max packet sizeMartin Storsjö2012-02-29
| | | | | | | This fixes cases where the user had specified one desired MTU via an option, and the protocol indicates another one. Signed-off-by: Martin Storsjö <martin@martin.st>
* Add a minor bump, changelog/APIchanges entry and some documentation for APIC ↵Anton Khirnov2012-02-29
| | | | support.
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