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Author SHA1 Message Date
Justin Ruggles f1c2915ce1 lavr: remove automatic context close/open for resampling compensation 
It adds unnecessary complication for insignificant usability improvement.
The user really should know if they'll need resampling compensation before
opening the context.

Note that only the documentation has changed. The current functionality will
still work until the next major bump.
 2012-12-16 19:26:17 -05:00
Justin Ruggles f322b20735 lavr: only save/restore the mixing matrix if mixing is being done 2012-12-16 19:26:17 -05:00
Justin Ruggles 230acdde26 lavr: move AudioMix struct definition to audio_mix.c 2012-12-15 12:26:15 -05:00
Justin Ruggles 4e5a8878d5 asyncts: ignore min_delta only if first_pts is set 2012-12-13 18:51:56 -05:00
Justin Ruggles c0dc57f126 asyncts: merge two conditions 2012-12-13 18:51:56 -05:00
Justin Ruggles 0ee440fe38 asyncts: cosmetics: reindent 2012-12-13 13:02:21 -05:00
Justin Ruggles b35e5d985d doc: improve documentation for the asyncts filter first_pts option 
Note the time base for the option. Add an additional example description.
 2012-12-13 13:02:21 -05:00
Justin Ruggles c143de40c3 asyncts: fix the asyncts behavior when using the first_pts option 
Currently it will do padding, but it does not properly handle
start-of-stream trimming as documented.
 2012-12-13 13:02:21 -05:00
Justin Ruggles 8083332c2d asyncts: use clipped delta value when setting resample compensation 
The max_comp option is supposed to limit maximum compensation, but
currently the clipped value is not actually used.
 2012-12-13 13:02:21 -05:00
Justin Ruggles f266486b2e asyncts: fix flushing of final samples at EOF 2012-12-13 13:02:21 -05:00
Justin RugglesandLuca Barbato 42b5688d5e lavr: fix segfault due to overlooked change needed in 14758e3 
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
 2012-12-12 09:09:00 +01:00
Justin Ruggles 14758e3211 lavr: temporarily store custom matrix in AVAudioResampleContext 
This allows AudioMix to be treated the same way as other conversion contexts
and removes the requirement to allocate it at the same time as the
AVAudioResampleContext.

The current matrix get/set functions are split between the public interface
and AudioMix private functions.
 2012-12-11 14:00:32 -05:00
Justin Ruggles 887d4c05c9 lavr: clarify documentation for avresample_get/set_matrix() 2012-12-11 14:00:32 -05:00
Justin Ruggles d04c17c913 swfdec: cosmetics: fix indentation 2012-12-11 09:37:17 -05:00
Justin Ruggles e70c5b034c swfdec: do better validation of tag length 
Avoids trying to read a packet with 0 or negative size.
Avoids a potential infinite loop due to seeking backwards.

Partially based on a patch by Michael Niedermayer.
 2012-12-11 09:36:26 -05:00
Justin Ruggles 1c012e6bfb x86: float_dsp: fix loading of the len parameter on x86-32 2012-12-07 21:19:29 -05:00
Justin Ruggles b64ba37c4c Changelog: add an entry for deprecating the avconv -vol option 2012-12-06 11:33:38 -05:00
Justin RugglesandJanne Grunau ecc8b02194 x86: float_dsp: fix compilation of ff_vector_dmul_scalar_avx() on x86-32 
Signed-off-by: Janne Grunau <janne-libav@jannau.net>
 2012-12-06 14:11:15 +01:00
Justin Ruggles 5e1bbb8c7e alacenc: add support for multi-channel encoding 2012-12-05 16:13:37 -05:00
Justin Ruggles 9a71d362a6 avconv: deprecate the -vol option 
Remove the code for volume scaling in avconv.c and instead auto-insert a
volume filter into the beginning of the filter chain.
 2012-12-05 11:23:37 -05:00
Justin Ruggles b30a363331 x86: af_volume: add SSE2/SSSE3/AVX-optimized s32 volume scaling 2012-12-05 11:23:37 -05:00
Justin Ruggles f96f1e06a4 x86: af_volume: add SSE2-optimized s16 volume scaling 2012-12-05 11:23:37 -05:00
Justin Ruggles b384e031da lavfi: add volume filter 
Based on the volume filter in FFmpeg written by Stefano Sabatini
<stefasab@gmail.com>.
 2012-12-05 11:23:37 -05:00
Justin Ruggles 9d5c62ba5b lavu/opt: do not filter out the initial sign character except for flags 
This allows parsing of special-case negative numbers like decibels.
 2012-12-05 11:23:36 -05:00
Justin Ruggles 5312268b34 eval: treat dB as decibels instead of decibytes 2012-12-05 11:23:36 -05:00
Justin Ruggles ac7eb4cb20 float_dsp: add vector_dmul_scalar() to multiply a vector of doubles 
Include x86-optimized versions for SSE2 and AVX.
 2012-12-05 11:23:36 -05:00
Justin Ruggles 95682d8cd2 avconv: fix variable shadowing in configure_input_audio_filter() 2012-11-28 18:13:21 -05:00
Justin Ruggles ab87d9b667 nuv: check for malloc failure when allocating extradata 
Also make sure extradata is freed in the case where multiple
NUV_EXTRADATA frame types are found. This may not happen in practice,
but it could happen in a malformed stream, which would lead to a memleak
if not handled.
 2012-11-28 11:18:50 -05:00
Justin Ruggles 838ed296df nuv: use the stream indices generated by avformat_new_stream() 2012-11-28 11:18:50 -05:00
Justin Ruggles c74f81786d nuv: cosmetics: pretty-printing 2012-11-28 11:18:50 -05:00
Justin Ruggles 5c7bf2ddde lavf: move nuv fourcc audio tags from riff to nuv 2012-11-28 11:18:50 -05:00
Justin Ruggles 261e9348ef lavf: add a common function for selecting a pcm codec from parameters 2012-11-28 11:18:50 -05:00
Justin Ruggles bfe5454cd2 lavf: move ff_codec_get_tag() and ff_codec_get_id() definitions to internal.h 2012-11-28 11:18:49 -05:00
Justin Ruggles cdaa1f84fb lavf: move "MP3 " fourcc from riff to nut 
Original commit, 7b24f9b, says it was added because it is used in libnut.
 2012-11-28 11:18:49 -05:00
Justin Ruggles 1d86aa8b0f lavr: do not pass consumed samples as a parameter to ff_audio_resample() 
Since the resampler handles buffering of unconsumed samples internally, the
caller does not need this information.
 2012-11-27 16:49:19 -05:00
Justin Ruggles d2f9f8e094 lavr: correct the documentation for the ff_audio_resample() return value 2012-11-27 16:49:18 -05:00
Justin Ruggles 7f534d11ed lavr: do not pass sample count as a parameter to ff_audio_convert() 
It will always be the number of samples in the input buffer, so just use that
directly instead of passing it as a separate parameter.
 2012-11-27 16:49:18 -05:00
Justin Ruggles 259d8c5647 riff: do not add empty metadata tags in INFO chunk 2012-11-26 19:08:51 -05:00
Justin Ruggles 3c370f5abc riff: only warn on a bad INFO chunk code size instead of failing 
fixes Bug 392
 2012-11-26 19:08:51 -05:00
Justin Ruggles 947f933687 x86: float_dsp: add SSE version of vector_fmul_scalar() 2012-11-26 11:30:19 -05:00
Justin Ruggles 284ea790d8 dsputil: move vector_fmul_scalar() to AVFloatDSPContext in libavutil 2012-11-26 11:29:06 -05:00
Justin Ruggles 3d3cf6745e aacdec: use float planar sample format for output 2012-11-25 19:06:36 -05:00
Justin Ruggles 8e134e5104 lavc: clarify get_buffer() documentation 
This is needed for the AAC decoder, which may need to call get_buffer()
more than once if the channel configuration changes.
 2012-11-25 19:06:36 -05:00
Justin Ruggles 3ffed68c2a mpegaudiodec: use planar sample format for output unless packed is requested 2012-11-25 19:05:58 -05:00
Justin Ruggles 2d3993ce8c x86: h264 qpel: use the correct number of utilized xmm regs in cglobal 
Fixes xmm register clobbering on win64.
 2012-11-25 18:48:43 -05:00
Justin Ruggles 00dd9a6d6a pcm: fix decoding of pcm_s16le_planar on big-endian 
The sample count is decremented by the DECODE() macro and needs to be reset
in each loop iteration. Also, DECODE() increments the src pointer so that does
not need to be done separately.
 2012-11-20 10:52:49 -05:00
Justin Ruggles b353321caa FATE: add 24-bit ALAC tests 2012-11-20 00:26:46 -05:00
Justin Ruggles 3e6c2a67a1 FATE: rename ALAC tests from alac-* to alac-16-* 
This will distinguish ALAC 16-bit tests from ALAC tests with other bit
depths.
 2012-11-20 00:26:45 -05:00
Justin Ruggles 7c278d2ae4 alacenc: support 24-bit encoding 2012-11-20 00:26:45 -05:00
Justin Ruggles c9d0f4506f pcmdec: use planar sample format for pcm_s16le_planar 2012-11-19 23:48:37 -05:00
Justin Ruggles 79b7747556 vorbisdec: use float planar sample format 2012-11-19 23:48:37 -05:00
Justin Ruggles 02738792c5 lavr: fix the decision for writing directly to the output buffer 
If there are any samples remaining in the output fifo from previous conversion
calls, we have to output those samples first instead of doing direct output
of the current samples.
 2012-11-17 20:09:40 -05:00
Justin Ruggles b9629acb6b yop: set channel layout 2012-11-12 10:33:23 -05:00
Justin Ruggles 935fbb66ef wtv: set channel layout for mpeg audio 2012-11-12 10:33:23 -05:00
Justin Ruggles 7b48d93e8a westwood_aud: set channel layout 2012-11-12 10:33:22 -05:00
Justin Ruggles 2ce7f820d4 wc3movie: set channel layout 2012-11-12 10:33:22 -05:00
Justin Ruggles a94b0267f2 tmv: set channel layout 2012-11-12 10:33:22 -05:00
Justin Ruggles 7f348bd764 tiertexseq: set channel layout 2012-11-12 10:33:22 -05:00
Justin Ruggles a3949fe11f swfdec: set channel layout 2012-11-12 10:33:22 -05:00
Justin Ruggles 1c7587728c sol: set channel layout 2012-11-12 10:33:22 -05:00
Justin Ruggles bfccd76adb smacker: set channel layout 2012-11-12 10:33:22 -05:00
Justin Ruggles 57e590e4b8 siff: set channel layout 2012-11-12 10:33:22 -05:00
Justin Ruggles a634896cf8 sierravmd: set channel layout 2012-11-12 10:33:21 -05:00
Justin Ruggles c1ac602d53 rtpdec_amr: set channel layout 2012-11-12 10:33:21 -05:00
Justin Ruggles ce842029ce rsodec: set channel layout 2012-11-12 10:33:21 -05:00
Justin Ruggles 60a585304c rmdec: set channel layout for RA version 3 2012-11-12 10:33:21 -05:00
Justin Ruggles 444b79c18a qcp: set channel layout 2012-11-12 10:33:21 -05:00
Justin Ruggles b5e3e77711 psxstr: set channel layout 2012-11-12 10:33:21 -05:00
Justin Ruggles c9759eb426 omadec: set channel layout 2012-11-12 10:33:21 -05:00
Justin Ruggles d4088efbe2 oggparsespeex: validate channel count and set channel layout 2012-11-12 10:33:20 -05:00
Justin Ruggles 232e9c4c4b nuv: set channel layout 2012-11-12 10:33:20 -05:00
Justin Ruggles cc57228e31 mxg: set channel layout 2012-11-12 10:33:20 -05:00
Justin Ruggles 66d7ceb4aa mvi: set channel layout 2012-11-12 10:33:20 -05:00
Justin Ruggles f6c6e5aac1 mpc7: set channel layout 2012-11-12 10:33:20 -05:00
Justin Ruggles f24b0b1b6c mmf: set channel layout 2012-11-12 10:33:20 -05:00
Justin Ruggles 87199d34db mm: set channel layout 2012-11-12 10:33:20 -05:00
Justin Ruggles ef1b23ad21 jvdec: set channel layout 2012-11-12 10:33:20 -05:00
Justin Ruggles 4371131283 iss: set channel layout 2012-11-12 10:33:20 -05:00
Justin Ruggles 41a2d9590d ipmovie: set channel layout 2012-11-12 10:33:19 -05:00
Justin Ruggles 024e03701c iff: set channel layout 2012-11-12 10:33:19 -05:00
Justin Ruggles 73e2007f3d idroqdec: set channel layout 2012-11-12 10:33:19 -05:00
Justin Ruggles f6c3adde41 gxfdec: set channel layout when applicable 2012-11-12 10:33:19 -05:00
Justin Ruggles d4a105ae5c gsmdec: set channel layout 2012-11-12 10:33:19 -05:00
Justin Ruggles 644d8d2e5a flvdec: set channel layout 2012-11-12 10:33:19 -05:00
Justin Ruggles d5ca70b103 dv: set channel layout 2012-11-12 10:33:19 -05:00
Justin Ruggles 49e7af06f2 dsicin: set channel layout 2012-11-12 10:33:19 -05:00
Justin Ruggles a05a63785c daud: set channel layout 
Update FATE reference due to WAVE channel mask now being set correctly.
 2012-11-12 10:33:18 -05:00
Justin Ruggles 0d09a5848f cdxl: set channel layout 2012-11-12 10:33:18 -05:00
Justin Ruggles e8088d6e4b bmv: set channel layout 2012-11-12 10:33:18 -05:00
Justin Ruggles 2fe804f316 bink: set channel layout 2012-11-12 10:33:18 -05:00
Justin Ruggles ff50d27a63 bfi: set channel layout 2012-11-12 10:33:18 -05:00
Justin Ruggles b5d1a15d1b bethsoftvid: set channel layout 2012-11-12 10:33:18 -05:00
Justin Ruggles 39f0e9b8c6 apc: set channel layout 2012-11-12 10:33:18 -05:00
Justin Ruggles 5595368bcc amr: set channel_layout 2012-11-12 10:33:18 -05:00
Justin Ruggles faf340f60c binkaudio: set channel layout 2012-11-12 00:41:02 -05:00
Justin Ruggles a903f8f087 Include libavutil/channel_layout.h instead of libavutil/audioconvert.h 
Also reorder some other #include when applicable.
 2012-11-11 13:35:12 -05:00
Justin Ruggles 5980f5dd18 lavu: rename audioconvert.* to channel_layout.* and deprecate audioconvert.h 2012-11-11 13:35:12 -05:00
Justin Ruggles 3a2731cbd3 flacenc: ensure the order is within the min/max range in LPC order search 
This fixes use of uninitialized values when the FLAC encoder uses the
2-level, 4-level, and 8-level search methods. Fixes failure of the
fate-flac-24-comp-8 test when run using valgrind.
 2012-11-08 13:57:34 -05:00
Justin Ruggles 3ba416408a avconv: rescale packet duration to muxer time base when flushing encoders 
Fixes Bug 385
 2012-11-08 13:57:34 -05:00
Justin Ruggles 00f8ad41c7 add 24-bit FLAC encoding to Changelog 
Also move a misplaced line from the 9_beta2 section to <next>
 2012-11-08 13:57:34 -05:00