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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.
* Add avcodec_decode_audio4().Justin Ruggles2011-12-02
| | | | | | Deprecate avcodec_decode_audio3(). Implement audio support in avcodec_default_get_buffer(). Implement the new audio decoder API in all audio decoders.
* avcodec: change number of plane pointers from 4 to 8 at next major bump.Justin Ruggles2011-12-02
| | | | | | Add AV_NUM_DATA_POINTERS to simplify the bump transition. This will allow for supporting more planar audio channels without having to allocate separate pointer arrays.
* Update developers documentation with coding conventions.Victor Vasiliev2011-12-02
| | | | Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
* svq1dec: avoid undefined get_bits(0) callMans Rullgard2011-12-02
| | | | Signed-off-by: Mans Rullgard <mans@mansr.com>
* ARM: h264dsp_neon cosmeticsMans Rullgard2011-12-02
| | | | | | | | - Replace 'ip' with 'r12'. - Use correct size designators for vld1/vst1. - Whitespace fixes. Signed-off-by: Mans Rullgard <mans@mansr.com>
* ARM: make some NEON macros reusableJanne Grunau2011-12-02
| | | | Signed-off-by: Mans Rullgard <mans@mansr.com>
* 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>
* fate: update asf seektestLuca Barbato2011-12-02
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* vp8: flush buffers on size changes.Aaron Colwell2011-12-02
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* doc: improve general documentation for MacOSXLuca Barbato2011-12-02
| | | | Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
* 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>
* Cleaned up alacenc.cNathan Adil Maxson2011-12-01
| | | | Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
* 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>
* h264: cap max has_b_frames at MAX_DELAYED_PIC_COUNT - 1.Ronald S. Bultje2011-12-01
| | | | | This prevents frame allocation overflows, and fixed fate-h264-conformance-mr3_tandberg_b with 2 threads.
* ARM: fix indentation in ff_dsputil_init_neon()Mans Rullgard2011-12-01
| | | | Signed-off-by: Mans Rullgard <mans@mansr.com>
* ARM: NEON put/avg_pixels8/16 cosmeticsMans Rullgard2011-12-01
| | | | | | | This makes whitespace and register names consistent with the style used in more recent code. Signed-off-by: Mans Rullgard <mans@mansr.com>
* ARM: add remaining NEON avg_pixels8/16 functionsMans Rullgard2011-12-01
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* ARM: clean up NEON put/avg_pixels macrosMans Rullgard2011-12-01
| | | | | | Although this adds a few lines, the macro calls are less convoluted. Signed-off-by: Mans Rullgard <mans@mansr.com>
* fate: split acodec-pcm into individual testsJustin Ruggles2011-12-01
| | | | | this removes 2 redundant tests for pcm in mkv. we can add the coverage back in later as fate-lavf tests if needed.
* swscale: #include "libavutil/mathematics.h"Reinhard Tartler2011-12-01
| | | | | | | | this file uses the M_PI macro since 4e74187db2f5db52f88729efc662df9d6bc763e1, so include the correct header directly. Signed-off-by: Reinhard Tartler <siretart@tauware.de>
* pmpdec: don't use deprecated av_set_pts_info.Anton Khirnov2011-12-01
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* rv34: align temporary block of "dct" coefsMans Rullgard2011-12-01
| | | | | | This is needed for optimised transforms. Signed-off-by: Mans Rullgard <mans@mansr.com>
* 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>
* vble: use dsp.add_hfyu_median_prediction() this allows asm optimizations to ↵Michael Niedermayer2011-11-30
| | | | | | | be used. Signed-off-by: Michael Niedermayer <michaelni@gmx.at> Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
* vble: remove flags copy, its not used in any speed relevant code.Michael Niedermayer2011-11-30
| | | | | Signed-off-by: Michael Niedermayer <michaelni@gmx.at> Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
* vble: merge len and val arraysDerek Buitenhuis2011-11-30
| | | | | | | | | | There's no reason to use two arrays for this. Based off commit 2fea60c60084c4e70d7cef128ea3bca5690ce465 to FFmpeg by Michael Niedermayer. Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com> Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
* vble: remove unused variable len.Michael Niedermayer2011-11-30
| | | | | Signed-off-by: Michael Niedermayer <michaelni@gmx.at> Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
* rtpdec: Templatize the code for different g726 bitrate variantsMartin Storsjö2011-11-30
| | | | Signed-off-by: Martin Storsjö <martin@martin.st>
* rv40: move loop filter to rv34dsp contextJanne Grunau2011-11-30
| | | | Signed-off-by: Mans Rullgard <mans@mansr.com>
* 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>
* avconv: make copy_tb on by default.Anton Khirnov2011-11-30
| | | | | | | I.e. on streamcopy set output codec timebase from input stream timebase (as opposed to input codec timebase). This should be more sane, because since the stream is not decoded, the input codec tb has no relation to the timestamps of the copied packets.
* 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.
* avconv: compute next_pts from input packet duration when possible.Anton Khirnov2011-11-30
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* 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.
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