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* avformat: Drop pointless "format" from container long namesDiego Biurrun2012-07-3086-135/+135
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* wv: K&R formatting cosmeticsAnton Khirnov2012-07-301-79/+96
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* flvdec: remove spurious use of stream idLuca Barbato2012-07-291-8/+7
| | | | We match streams by codec id now.
* lavf: deprecate r_frame_rate.Anton Khirnov2012-07-2914-31/+69
| | | | | | | | | | | | | | | | | | | | According to its description, it is supposed to be the LCM of all the frame durations. The usability of such a thing is vanishingly small, especially since we cannot determine it with any amount of reliability. Therefore get rid of it after the next bump. Replace it with the average framerate where it makes sense. FATE results for the wtv and xmv demux tests change. In the wtv case this is caused by the file being corrupted (or possibly badly cut) and containing invalid timestamps. This results in lavf estimating the framerate wrong and making up wrong frame durations. In the xmv case the file contains pts jumps, so again the estimated framerate is far from anything sane and lavf again makes up different frame durations. In some other tests lavf starts making up frame durations from different frame.
* lavf: round estimated average fps to a "standard" fps.Anton Khirnov2012-07-291-0/+18
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* lavf: use dts difference instead of AVPacket.duration in find_stream_info()Anton Khirnov2012-07-292-7/+41
| | | | | | | | | | AVPacket.duration is mostly made up and thus completely useless, this is especially true for video streams. Therefore use dts difference for framerate estimation and the max_analyze_duration check. The asyncts test now needs -analyzeduration, because the default is 5 seconds and the audio stream in the sample appears at ~10 seconds.
* avf: introduce nobuffer optionLuca Barbato2012-07-294-5/+12
| | | | | | | | | | | | | Useful in cases where a significant analyzeduration is still needed, while minimizing buffering before output. An example is processing low-latency streams where all media types won't necessarily come in if the analyzeduration is small. Additional changes by Josh Allmann <joshua.allmann@gmail.com> Signed-off-by: Anton Khirnov <anton@khirnov.net>
* wv: return meaningful error codes.Anton Khirnov2012-07-281-22/+23
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* wv: return AVERROR_EOF on EOF, not EIO.Anton Khirnov2012-07-281-1/+1
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* mp3dec: forward errors for av_get_packet().Anton Khirnov2012-07-281-3/+2
| | | | | | | Don't invent a bogus EIO error. The code now doesn't check for ret == 0, but that check is redundant, av_get_packet() never returns 0.
* mp3dec: remove a pointless local variable.Anton Khirnov2012-07-281-4/+2
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* mp3dec: remove commented out cruft.Anton Khirnov2012-07-281-1/+0
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* lavf: remove commented out cruft in avformat_find_stream_info()Anton Khirnov2012-07-281-7/+1
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* Add missing libavutil/time.h includes.Anton Khirnov2012-07-286-0/+6
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* hls: Proceed to the next segment at any error codeMartin Storsjö2012-07-281-2/+0
| | | | | | | | | | | | | Previously, we returned any error code except AVERROR_EOF to the caller - only if AVERROR_EOF or 0 was returned, we proceeded to the next segment. With some setups of web servers, using Connection: close in https and GnuTLS, we don't get a clean error code at the end of segments. In those cases, just proceed to the next segment. Tested-by: Antti Seppälä <a.seppala@gmail.com> Signed-off-by: Martin Storsjö <martin@martin.st>
* tls: Return AVERROR_EOF if the TLS_read/write functions return 0Martin Storsjö2012-07-281-2/+2
| | | | | | | | | | OpenSSL returns 0 when the peer has closed the connection. GnuTLS doesn't return that though, but returns GNUTLS_E_UNEXPECTED_PACKET_LENGTH if the connection simply is closed without a clean close notify packet. Tested-by: Antti Seppälä <a.seppala@gmail.com> Signed-off-by: Martin Storsjö <martin@martin.st>
* rtmp: Check the buffer length of ping packetsSamuel Pitoiset2012-07-261-0/+12
| | | | Signed-off-by: Martin Storsjö <martin@martin.st>
* rtmp: Allow having more unknown data at the end of a chunk size packet ↵Samuel Pitoiset2012-07-261-2/+2
| | | | | | without failing Signed-off-by: Martin Storsjö <martin@martin.st>
* rtmp: Prevent reading outside of an allocate buffer when receiving server ↵Samuel Pitoiset2012-07-261-0/+7
| | | | | | bandwidth packets Signed-off-by: Martin Storsjö <martin@martin.st>
* Support urlencoded http authentication credentialsAntti Seppälä2012-07-265-12/+146
| | | | | | | | | It should be possible to specify usernames in http requests containing urlencoded characters. This patch adds support for decoding the auth strings. Signed-off-by: Antti Seppälä <a.seppala@gmail.com> Signed-off-by: Martin Storsjö <martin@martin.st>
* rtmp: Return an error when the client bandwidth is incorrectSamuel Pitoiset2012-07-251-2/+10
| | | | Signed-off-by: Martin Storsjö <martin@martin.st>
* rtmp: Return proper error code in handle_server_bwSamuel Pitoiset2012-07-251-1/+1
| | | | Signed-off-by: Martin Storsjö <martin@martin.st>
* rtmp: Return proper error code in handle_client_bwSamuel Pitoiset2012-07-251-1/+1
| | | | Signed-off-by: Martin Storsjö <martin@martin.st>
* rtmp: Return proper error codes in handle_chunk_sizeSamuel Pitoiset2012-07-251-2/+2
| | | | Signed-off-by: Martin Storsjö <martin@martin.st>
* rtmp: Factorize the code by adding handle_invokeSamuel Pitoiset2012-07-251-66/+77
| | | | Signed-off-by: Martin Storsjö <martin@martin.st>
* rtmp: Factorize the code by adding handle_chunk_sizeSamuel Pitoiset2012-07-251-15/+30
| | | | Signed-off-by: Martin Storsjö <martin@martin.st>
* rtmp: Factorize the code by adding handle_pingSamuel Pitoiset2012-07-251-4/+16
| | | | Signed-off-by: Martin Storsjö <martin@martin.st>
* rtmp: Factorize the code by adding handle_client_bwSamuel Pitoiset2012-07-251-8/+18
| | | | Signed-off-by: Martin Storsjö <martin@martin.st>
* rtmp: Factorize the code by adding handle_server_bwSamuel Pitoiset2012-07-251-6/+17
| | | | Signed-off-by: Martin Storsjö <martin@martin.st>
* rtmp: Add a new option 'rtmp_pageurl'Samuel Pitoiset2012-07-251-0/+7
| | | | | | | This option specifies the URL of the web page in which the media was embedded. Signed-off-by: Martin Storsjö <martin@martin.st>
* rtmp: Make the description of the rtmp_tcurl option more genericSamuel Pitoiset2012-07-251-1/+1
| | | | Signed-off-by: Martin Storsjö <martin@martin.st>
* sctp: add port missing error messageJordi Ortiz2012-07-251-1/+5
| | | | | | | Without this patch a user a bit absent-minded may not notice that the connection doesn't work because the port is missing. Signed-off-by: Martin Storsjö <martin@martin.st>
* tcp: add port missing error messageJordi Ortiz2012-07-251-2/+5
| | | | | | | Without this patch a user a bit absent-minded may not notice that the connection doesn't work because the port is missing. Signed-off-by: Martin Storsjö <martin@martin.st>
* rtmp: Add credit/copyright to librtmp authors for parts of the RTMPE codeMartin Storsjö2012-07-242-0/+4
| | | | | | Our implementation of RTMPE is heavily based on librtmp. Signed-off-by: Martin Storsjö <martin@martin.st>
* rtmp: Move the CONFIG_ condition into the if conditionsSamuel Pitoiset2012-07-241-8/+7
| | | | | | | | This makes sure these calls are removed by dead code elimination even if optimization is disabled. This fixes building without crypto libraries without optimization. Signed-off-by: Martin Storsjö <martin@martin.st>
* build: Skip compiling rtmpdh.h if ffrtmpcrypt protocol is not enabledDiego Biurrun2012-07-241-0/+1
| | | | | The ffrtmpcrypt protocol depends on external libraries, which are also required to compile the header file.
* rtp: Only choose static payload types if the sample rate and channels are rightAdriano Pallavicino2012-07-241-3/+11
| | | | | | | | | If using a different sample rate or number of channels, use a dynamic payload type instead, where the parameters are passed in the SDP. G722 is a special case where the normal rules don't apply. Signed-off-by: Martin Storsjö <martin@martin.st>
* wav: init st to NULL to avoid a false-positive warning.Clément Bœsch2012-07-231-1/+1
| | | | | | | | | | | | If st is NULL, it means no 'fmt ' tag is found, but 'data' tag (which needs a previous 'fmt ' tag to be parsed correctly and st initialized) check will make sure st is never dereferenced in that case. Fixes warning: libavformat/wav.c: In function ‘wav_read_header’: libavformat/wav.c:499:44: warning: ‘st’ may be used uninitialized in this function [-Wmaybe-uninitialized] Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
* RTMPTE protocol supportSamuel Pitoiset2012-07-235-6/+54
| | | | Signed-off-by: Martin Storsjö <martin@martin.st>
* RTMPE protocol supportSamuel Pitoiset2012-07-238-8/+913
| | | | | | | | | | This adds two protocols, but one of them is an internal implementation detail just used as an abstraction layer/generalization in the code. The RTMPE protocol implementation uses ffrtmpcrypt:// as an alternative to the tcp:// protocol. This allows moving most of the lower level logic out from the higher level generic rtmp code. Signed-off-by: Martin Storsjö <martin@martin.st>
* rtmp: Add ff_rtmp_calc_digest_pos()Samuel Pitoiset2012-07-232-10/+27
| | | | | | | This function is used for calculating digest position for RTMP handshake packets. Signed-off-by: Martin Storsjö <martin@martin.st>
* rtmp: Rename rtmp_calc_digest to ff_rtmp_calc_digest and make it globalSamuel Pitoiset2012-07-232-33/+36
| | | | Signed-off-by: Martin Storsjö <martin@martin.st>
* lavf: use conditional notation for default codec in muxer declarations.Ronald S. Bultje2012-07-226-57/+18
| | | | | This removes the use of macro nesting in these code constructs, which makes it easier to parse in pre-processors.
* matroskadec: return more correct error code on read error.Anton Khirnov2012-07-221-1/+2
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* Bump libavcodec and libavformat minor versions for G.723.1 decoder and demuxerKostya Shishkov2012-07-221-1/+1
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* G.723.1 demuxer and decoderMohamed Naufal Basheer2012-07-224-1/+88
| | | | Signed-off-by: Kostya Shishkov <kostya.shishkov@gmail.com>
* rtsp: remove terminal comma in FF_RTP_FLAG_OPTS macro.Ronald S. Bultje2012-07-214-4/+4
| | | | | | | This makes usage of the macro look more natural when used with array entries. Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
* mpegenc: remove disabled codeDiego Biurrun2012-07-211-116/+0
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* cafdec: allow larger ALAC magic cookieJustin Ruggles2012-07-191-1/+1
| | | | | It already skips any extra bytes at the end, and apparently there are some samples in the wild with larger 'kuki' chunks.
* avformat: move 'chan' tag parsing to mov_chan.c to share with the CAF demuxerJustin Ruggles2012-07-194-47/+63
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