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* latmdec: fix audio specific config parsingJanne Grunau2011-12-03
| | | | | | | | | Pass the correct size in bits to mpeg4audio_get_config and add a flag to disable parsing of the sync extension when the size is not known. Latm with AudioMuxVersion 0 does not specify the size of the audio specific config. Data after the audio specific config can be misinterpreted as sync extension resulting in random and wrong configs.
* Do not memcpy raw video frames when using null muxerMans Rullgard2011-12-02
| | | | | | | | | | | Commit 035af99 made avconv always call an encoder when using the null muxer. While useful for 2-pass encodes, it inadvertently caused an extra memcpy of raw frames when decoding only. This hack restores the old behaviour when only decoding while allowing use of the null muxer with encoded streams as well. Signed-off-by: Mans Rullgard <mans@mansr.com>
* asf: use packet dts as approximation of ptsJohn Stebbins2011-12-02
| | | | | | Having a somehow off seeking is better than having none at all. Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
* asf: do not call av_read_frameLuca Barbato2011-12-02
| | | | The asf_read_pts should read the bitstream directly.
* rtsp: Initialize the media_type_mask in the rtp guessing demuxerMartin Storsjö2011-12-02
| | | | | | | | | | | The media_type_mask is initialized via AVOptions for the rtsp and sdp demuxers, but it isn't available as an option for the rtp guessing demuxer (since it doesn't really make sense there). Therefore, it must be manually initialized instead, since a zero value means no media types at all are accepted. Signed-off-by: Martin Storsjö <martin@martin.st>
* rtpenc: Add support for G726 audioMartin Storsjö2011-12-01
| | | | Signed-off-by: Martin Storsjö <martin@martin.st>
* rtpdec: Interpret the different G726 names as bits_per_coded_sampleMartin Storsjö2011-12-01
| | | | | | | | | | | | | | For the standardized 8 kHz sample rate, this works exactly the same. For nonstandard sample rates, the different predefined G726 names (G726-16, G726-24, G726-32, G726-40) are interpreted as an indication of the bits per coded sample, even though their actual bitrates aren't what the name specifies. This feels more sane than using free-form names for nonstandard sample rate/bitrate combinations, e.g like G726-22, G726-33 for 11025 Hz. Signed-off-by: Martin Storsjö <martin@martin.st>
* rtpenc: Change rtp_send_samples to handle sample sizes other than even bytesMartin Storsjö2011-12-01
| | | | Signed-off-by: Martin Storsjö <martin@martin.st>
* rtpenc: Cast a rescaling parameter to int64_tMartin Storsjö2011-12-01
| | | | | | | This avoids overflow if frame_size is over 2147, since both frame_size and AV_TIME_BASE are plain integers. Signed-off-by: Martin Storsjö <martin@martin.st>
* pmpdec: don't use deprecated av_set_pts_info.Anton Khirnov2011-12-01
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* Add PlayStation Portable PMP format demuxerReimar Döffinger2011-12-01
| | | | | | | | | Not yet complete, for demuxing AAC the AAC header must be generated manually. Possibly the decoder could accept the header as extradata to simplify this. Signed-off-by: Anton Khirnov <anton@khirnov.net>
* proto: Realign struct initializersMartin Storsjö2011-12-01
| | | | Signed-off-by: Martin Storsjö <martin@martin.st>
* proto: Use .priv_data_size to allocate the private contextMartin Storsjö2011-12-01
| | | | | | | | This simplifies the open functions by avoiding one function call that needs error checking, reducing the amount of extra bulk code. Signed-off-by: Martin Storsjö <martin@martin.st>
* mmsh: Properly clean up if the second ffurl_alloc failedMartin Storsjö2011-12-01
| | | | Signed-off-by: Martin Storsjö <martin@martin.st>
* rtmp: Clean up properly if the handshake failedMartin Storsjö2011-12-01
| | | | | | This prevents memory leaks if this function returns an error. Signed-off-by: Martin Storsjö <martin@martin.st>
* md5proto: Remove the get_file_handle functionMartin Storsjö2011-12-01
| | | | | | | The private data pointer isn't a file handle, this protocol doesn't have any file handle to return. Signed-off-by: Martin Storsjö <martin@martin.st>
* applehttpproto: Use the close function if the open function failsMartin Storsjö2011-12-01
| | | | | | This should clean up leaked memory. Signed-off-by: Martin Storsjö <martin@martin.st>
* http: Make sure proxyauth is initializedMartin Storsjö2011-12-01
| | | | | | | | | This string will be passed to ff_http_auth_create_response even if no proxy is used, resulting in reading uninitialized memory. The other auth string is always initialized by av_url_split. Signed-off-by: Martin Storsjö <martin@martin.st>
* rtpdec: Templatize the code for different g726 bitrate variantsMartin Storsjö2011-11-30
| | | | Signed-off-by: Martin Storsjö <martin@martin.st>
* lavf: make av_set_pts_info private.Anton Khirnov2011-11-30
| | | | It's supposed to be called only from (de)muxers.
* rtpdec: Add support for G726 audioMiroslav Slugeň2011-11-30
| | | | | | | | | | | This requires using a separate init function, since there isn't necessarily any fmtp lines for this codec, so parse_sdp_a_line won't be called. Incorporating it with the alloc function wouldn't do either, since it is called before the full rtpmap line is parsed (where the sample rate is extracted). Signed-off-by: Martin Storsjö <martin@martin.st>
* rtpdec: Add an init function that can do custom codec context initializationMartin Storsjö2011-11-30
| | | | Signed-off-by: Martin Storsjö <martin@martin.st>
* matroskadec: don't set codec timebase.Anton Khirnov2011-11-30
| | | | It's not supposed to be set outside of lavc.
* rmdec: don't set codec timebase.Anton Khirnov2011-11-30
| | | | | It's not supposed to be set outside of lavc. Set r_frame_rate instead.
* lavf: estimate frame duration from r_frame_rate.Anton Khirnov2011-11-30
| | | | | If r_frame_rate is set, it should be more reliable for this than either codec or stream timebase.
* adtsenc: Check frame size.Alex Converse2011-11-28
| | | | | | Inspired by work from: Michael Niedermayer <michaelni@gmx.at>. Signed-off-by: Alex Converse <alex.converse@gmail.com>
* Eliminate pointless 0/NULL initializers in AVCodec and similar declarations.Diego Biurrun2011-11-28
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* md5proto: Fix order of operations.Alex Converse2011-11-27
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* musepack: fix signed shift overflow in mpc_read_packet()Mans Rullgard2011-11-27
| | | | | | | Using an unsigned variable avoids problems with overflows. There is further no need for a 64-bit intermediate here. Signed-off-by: Mans Rullgard <mans@mansr.com>
* mov: Make format string match variable type.Alex Converse2011-11-26
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* Generalize RIFF INFO tag support; support reading INFO tag in wavVictor Vasiliev2011-11-26
| | | | Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
* adx: rename ff_adx_decode_header() to avpriv_adx_decode_header()Justin Ruggles2011-11-26
| | | | | It is used by the ADX decoder, and therefore needs to be exported in order to work with shared libs.
* avformat: add CRI ADX format demuxerJustin Ruggles2011-11-26
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* adx: add an ADX parser.Justin Ruggles2011-11-26
| | | | | | This simplifies the decoder so it doesn't have to process an in-packet header or handle arbitrary-sized packets. It also fixes decoding of files with large headers.
* sol: use AV_RL32 instead of raw pointer castMans Rullgard2011-11-26
| | | | Signed-off-by: Mans Rullgard <mans@mansr.com>
* nullenc: drop AVFMT_RAWPICTURE from the flagsLuca Barbato2011-11-25
| | | | | This makes the two pass encoding with x264 working with -f null - as first pass.
* cdg: skip packets that don't contain a cdg command.Justin Ruggles2011-11-23
| | | | Signed-off-by: Anton Khirnov <anton@khirnov.net>
* crcenc: add flagsAnton Khirnov2011-11-23
| | | | | | | | | | AVFMT_NOTIMESTAMPS for crc, as it ignores the timestamps. AVFMT_VARIABLE_FPS for framecrc, as it prints dts. Many FATE changes, because avconv is no longer duplicating frames in those tests. Also added -vsync 0 for some tests to prevent avconv from dropping frames until it can be fixed more properly.
* md5enc: add flagsAnton Khirnov2011-11-23
| | | | | | | | | AVFMT_NOTIMESTAMPS for md5, as it ignores the timestamps. AVFMT_VARIABLE_FPS for framemd5, as it prints dts. -vsync 0 for the vp8 test is needed because with vsync 2 the timestamp guessing code gets confused by an altref frame that is never displayed and drops a frame later.
* mov: Remove some redundant and obsolete comments.Alex Converse2011-11-22
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* doxy: structure libavformat groupsLuca Barbato2011-11-22
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* pthread: do not touch has_b_framesJanne Grunau2011-11-22
| | | | | | | | | | | | | | | | | | Adding the thread count in frame level multithreading to has_b_frames as an additional delay causes more problems than it solves. For example inconsistent behaviour during timestamp calculation in libavformat. Thread count and frame level multithreading are both set by the user. If the additional delay caused by frame level multithreading needs to be considered in the calling code it has all information to take it into account. Should it become necessary to calculate a maximum delay inside libavcodec it should be exported as its own field and not reusing an existing field. Based on a patch by Michael Niedermayer. Signed-off-by: Janne Grunau <janne-libav@jannau.net>
* httpproxy: Specify the object files needed in the MakefileMartin Storsjö2011-11-20
| | | | Signed-off-by: Martin Storsjö <martin@martin.st>
* rtmp: do not hardcode invoke numbersJosh Allmann2011-11-18
| | | | | | | | Note: FCPublish/FCUnpublish are adobe server specific and not described in the rtmp specification. Some servers might not cope with them at all. Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
* avio: Mark the old interrupt callback mechanism as deprecatedMartin Storsjö2011-11-18
| | | | Prepare for removing it at an upcoming major bump.
* applehttp: Fix seeking in streams not starting at DTS=0Panagiotis H.M. Issaris2011-11-18
| | | | | | | | | | | | | | | | | | | | | | | | | The Apple HTTP Live Streaming demuxer's implementation of seeking searches for the MPEG TS segment which contains the requested timestamp. In its current implementation it assumes that the first segment will start from 0. But, MPEG TS streams do not necessarily start with timestamp (near) 0, causing seeking to fail for those streams. This also occurs when using live streaming of HTTP Live Streams. In this case sliding playlists may be used, which means that in that case only the last x encoded segments are stored, the earlier segments get deleted from disk and removed from the playlist. Because of this, when starting playback of a stream in the middle of such a broadcast, the initial segment fetched after parsing the m3u8 playlist will not start from timestamp (near) 0, causing (the admittedly limited live) seeking to fail. This patch changes this demuxers seeking implementation to use the initial DTS as an offset for searching the segments containing the requested timestamp. Signed-off-by: Martin Storsjö <martin@martin.st>
* http: Don't use the normal http proxy mechanism for httpsMartin Storsjö2011-11-18
| | | | | | | | | | The tls protocol handles connections via proxies internally. With TLS/SSL, the peer verification requires that the client speaks directly with the server, since the proxy doesn't have the remote server's private key. Signed-off-by: Martin Storsjö <martin@martin.st>
* tls: Handle connection via a http proxyMartin Storsjö2011-11-18
| | | | Signed-off-by: Martin Storsjö <martin@martin.st>
* http: Reorder two code blocksMartin Storsjö2011-11-18
| | | | | | This is in preparation for a later commit. Signed-off-by: Martin Storsjö <martin@martin.st>
* http: Add a new protocol for opening connections via http proxiesMartin Storsjö2011-11-18
| | | | | | | | | | This opens a plain TCP connection through the proxy via the CONNECT HTTP method. Normally, this is allowed for connections on port 443, but can in general be used to allow connections to any port (depending on proxy configuration), and could thus be used to tunnel any TCP connection via a HTTP proxy. Signed-off-by: Martin Storsjö <martin@martin.st>
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