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* jv demux: set video stream durationPaul B Mahol2012-03-14
| | | | | Signed-off-by: Paul B Mahol <onemda@gmail.com> Signed-off-by: Janne Grunau <janne-libav@jannau.net>
* http: Clear the auth state on redirectsMartin Storsjö2012-03-13
| | | | | | | Currently we only try continuing with the same auth mechanism as the initial request. Signed-off-by: Martin Storsjö <martin@martin.st>
* http: Retry auth if it failed due to being staleMartin Storsjö2012-03-13
| | | | | | | Allow up to 4 retries for normal requests, where both the proxy and the target server might need to authenticate. Signed-off-by: Martin Storsjö <martin@martin.st>
* rtsp: Resend new keepalive commands if they used stale authMartin Storsjö2012-03-13
| | | | | | | | | | | | | These commands are sent asynchronously, not waiting for the reply. This reply is parsed later by ff_rtsp_tcp_read_packet or udp_read_packet. If the reply indicates that we used stale authentication and need to use a new nonce, resend a new keepalive command immediately. This is the only request sent asynchronously, so currently there's no other command that needs to be resent in the same way. Signed-off-by: Martin Storsjö <martin@martin.st>
* rtsp: Retry authentication if failed due to being staleMartin Storsjö2012-03-13
| | | | Signed-off-by: Martin Storsjö <martin@martin.st>
* httpauth: Parse the stale field in digest authMartin Storsjö2012-03-13
| | | | Signed-off-by: Martin Storsjö <martin@martin.st>
* iff: make .long_name more descriptivePaul B Mahol2012-03-12
| | | | | Signed-off-by: Paul B Mahol <onemda@gmail.com> Signed-off-by: Martin Storsjö <martin@martin.st>
* rtsp: Fix a typoMartin Storsjö2012-03-12
| | | | Signed-off-by: Martin Storsjö <martin@martin.st>
* Fix a bunch of common typos.Diego Biurrun2012-03-09
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* mpegts: Always honor a registration descriptor if present and there is no ↵Alex Converse2012-03-09
| | | | other codec information.
* rtsp: Only set the ttl parameter if the server actually gave a valueMartin Storsjö2012-03-09
| | | | | | | Passing ttl=0 to the rtp/udp url contexts makes packets never leave the host machine. Signed-off-by: Martin Storsjö <martin@martin.st>
* udp: Set ttl for read-write streams, too, not only for write-only onesMartin Storsjö2012-03-09
| | | | Signed-off-by: Martin Storsjö <martin@martin.st>
* udp: Only bind to the multicast address if in read-only modeMartin Storsjö2012-03-09
| | | | | | | | | | | | | | | This fixes sending back RTCP RR packets if receiving RTP over multicast. If the multicast stream is sent on demand (set up and signalled via RTSP), the sender might depend on getting RTCP RR packets knowing that there are listeners, otherwise the stream can be closed after a certain timeout. This fixes receiving RTSP streams over multicast on unix, from certain Axis cameras. Signed-off-by: Martin Storsjö <martin@martin.st>
* udp: Clarify the comment about binding the multicast addressMartin Storsjö2012-03-09
| | | | Signed-off-by: Martin Storsjö <martin@martin.st>
* udp: Reorder commentsMartin Storsjö2012-03-09
| | | | | | | | | When this code was added in 36b532815cb83, the new code was added between the existing comment and the existing line of code, making the old comment seem to refer to the new code. This makes it read correctly. Signed-off-by: Martin Storsjö <martin@martin.st>
* Fix uninitialized reads on malformed ogg files.Dale Curtis2012-03-08
| | | | | | | | | | The ogg decoder wasn't padding the input buffer with the appropriate FF_INPUT_BUFFER_PADDING_SIZE bytes. Which led to uninitialized reads in various pieces of parsing code when they thought they had more data than they actually did. Signed-off-by: Dale Curtis <dalecurtis@chromium.org> Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
* rtpenc: Fix the AVRational used for av_rescale_q_rndMartin Storsjö2012-03-08
| | | | | | | The current one has a zero denominator - this is what was intended in 14aecc50fae6. Signed-off-by: Martin Storsjö <martin@martin.st>
* udp: Print an error message if bind failsMartin Storsjö2012-03-07
| | | | Signed-off-by: Martin Storsjö <martin@martin.st>
* smacker: error out if palette copy-with-offset overruns palette size.Ronald S. Bultje2012-03-07
| | | | | Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind CC: libav-stable@libav.org
* mov: Allow last chunk to have an arbitrary number of samples.Carl Eugen Hoyos2012-03-06
| | | | | | | Fixes ticket #673. (cherry picked from commit 8dcd2a41ecff8cc1e9b20cc267df54c59878ab3b) Signed-off-by: Alex Converse <alex.converse@gmail.com>
* 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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