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Author SHA1 Message Date
Justin Ruggles acd9948e74 lavr: x86: fix ff_conv_fltp_to_flt_6ch function prototypes 
Changed to match the number of parameters in conv_func_interleave(), which is
how they are called. The change isn't strictly necessary because the 4th
parameter is not used, but the code is clearer if they match.
 2012-06-26 12:29:35 -04:00
Justin Ruggles 14a34d90ad lavr: x86: merge some branches 2012-06-25 13:49:18 -04:00
Justin Ruggles ae46fbee1d af_amix: allow float planar sample format as input 2012-06-18 18:01:14 -04:00
Justin Ruggles f0ece49e74 af_amix: use AVFloatDSPContext.vector_fmac_scalar() 2012-06-18 18:01:14 -04:00
Justin Ruggles 82b2df9790 float_dsp: add x86-optimized functions for vector_fmac_scalar() 2012-06-18 18:01:14 -04:00
Justin Ruggles cb5042d02c float_dsp: Move vector_fmac_scalar() from libavcodec to libavutil 2012-06-18 18:01:14 -04:00
Justin Ruggles 4e4dd71730 lavr: Add x86-optimized function for flt to s32 conversion 2012-06-18 16:16:59 -04:00
Justin Ruggles 6c63cbfe7a lavr: Add x86-optimized function for flt to s16 conversion 2012-06-18 16:16:59 -04:00
Justin Ruggles 97ce1ba867 lavr: Add x86-optimized functions for s32 to flt conversion 2012-06-18 16:16:59 -04:00
Justin Ruggles 5904f25b9f lavr: Add x86-optimized functions for s32 to s16 conversion 2012-06-18 16:16:59 -04:00
Justin Ruggles d721f67d0a lavr: Add x86-optimized functions for s16 to flt conversion 2012-06-18 16:16:59 -04:00
Justin Ruggles 1168e29df1 lavr: Add x86-optimized function for s16 to s32 conversion 2012-06-18 16:16:59 -04:00
Justin Ruggles f61ce90caa lavr: add x86-optimized functions for mixing 1-to-2 s16p with flt coeffs 2012-06-18 11:24:10 -04:00
Justin Ruggles 29f7490c46 lavr: add x86-optimized functions for mixing 1-to-2 fltp with flt coeffs 2012-06-18 11:24:10 -04:00
Justin Ruggles 4795362660 golomb: check remaining bits during unary decoding in get_ur_golomb_jpegls() 
Fixes infinite loop in FLAC decoding in case of a truncated bitstream due to
the safe bitstream reader returning 0's at the end.

Fixes Bug 310.

CC:libav-stable@libav.org
 2012-06-12 14:33:39 -04:00
Justin Ruggles 03e098e7a0 lavu: change versioning script to include all av* prefixed symbols 
Needed to properly export avpriv_* symbols.
 2012-06-08 15:47:59 -04:00
Justin Ruggles 3b81a18ba3 float_dsp: ppc: add a separate header for Altivec function prototypes 
Also include config.h so that HAVE_ALTIVEC will be defined.
Fixes compilation on PPC with Altivec enabled.
 2012-06-08 14:52:01 -04:00
Justin Ruggles d5a7229ba4 Add a float DSP framework to libavutil 
Move vector_fmul() from DSPContext to AVFloatDSPContext.
 2012-06-08 13:14:38 -04:00
Justin Ruggles 98db4e2a4e PPC: Move types_altivec.h and util_altivec.h from libavcodec to libavutil 
This will allow for easier implementation of Altivec functions in libraries
other than libavcodec.
 2012-06-08 13:14:38 -04:00
Justin Ruggles 94d2b0d2fd ARM: Move asm.S from libavcodec to libavutil 
This will allow for easier implementation of ARM-optimized functions in
libraries other than libavcodec.
 2012-06-08 13:14:38 -04:00
Justin Ruggles c7bd556d48 lavfi: amix: check active input count before calling request_samples 
fixes use of the amix filter with only 1 input
 2012-05-30 16:20:03 -04:00
Justin Ruggles b75726cb79 lavr: add x86-optimized function for mixing 2 to 1 s16p with q8 coeffs 2012-05-29 15:33:25 -04:00
Justin Ruggles c140fb2cbc lavr: add x86-optimized functions for mixing 2 to 1 s16p with float coeffs 2012-05-29 15:33:18 -04:00
Justin Ruggles 8dfc122719 lavr: add C functions for mixing 2 to 1 channels with s16p format 2012-05-29 15:16:20 -04:00
Justin Ruggles c7448c182a lavfi: add audio mix filter 2012-05-23 16:37:34 -04:00
Justin RugglesandDiego Biurrun 713548cbad x86: lavc: use %if HAVE_AVX guards around AVX functions in yasm code. 
This is needed for older versions of yasm/nasm that do not support AVX.

Signed-off-by: Diego Biurrun <diego@biurrun.de>
 2012-05-22 20:46:02 +02:00
Justin Ruggles e0d8427dce af_resample: remove an extra space in the log output 2012-05-21 18:26:51 -04:00
Justin Ruggles afeb3590fc lavfi: add an audio split filter 
Based on current version of the asplit filter in FFmpeg written by
Stefano Sabatini and others.
 2012-05-21 14:49:52 -04:00
Justin Ruggles cc30080b3f lavfi: rename vf_split.c to split.c 
This is in preparation for adding an audio split filter.
 2012-05-21 14:49:52 -04:00
Justin Ruggles 95a98ab3f0 ac3dsp: simplify x86 versions of ac3_max_msb_abs_int16 
Simplifies the code by using cpuflags and a new macro.
Also fixes the invalid use of the MMX2 pshufw operation in the MMX-only
function.
 2012-05-15 15:23:59 -04:00
Justin Ruggles 5cc6d5244d lavr: replace the SSE version of ff_conv_fltp_to_flt_6ch() with SSE4 and AVX 
The current SSE version is slower than the MMX version on Athlon64 and Sandy
Bridge, but the SSE4 and AVX versions are faster on Sandy Bridge.
 2012-05-09 16:17:59 -04:00
Justin Ruggles 85e5b866dc avplay: properly close/reopen AVAudioResampleContext on channel layout change 
fixes Bug#280
 2012-05-07 12:42:32 -04:00
Justin Ruggles f132248028 avcodec: do not needlessly set packet size to 0 in avcodec_encode_audio2() 
It is already set to 0 by av_free_packet()
 2012-05-07 12:14:05 -04:00
Justin Ruggles 74e10b6204 avcodec: for audio encoding, reset output packet when it is not valid 2012-05-07 12:14:05 -04:00
Justin Ruggles fa0319b4fd avcodec: refactor avcodec_encode_audio2() to merge common branches 2012-05-07 12:14:05 -04:00
Justin Ruggles b461cd4deb avcodec: remove fallbacks for AVCodec.encode() in avcodec_encode_audio2() 
We no longer have any audio encoders using AVCodec.encode().
 2012-05-07 12:14:04 -04:00
Justin Ruggles e5356ebf22 cosmetics: indentation 2012-05-03 16:28:08 -04:00
Justin Ruggles 8916f1fbcb avformat: only fill-in interpolated timestamps if duration is non-zero 
This avoids returning duplicate timestamps for multiple packets when the
demuxer does not provide all timestamps and packet duration is not known.
 2012-05-03 16:28:08 -04:00
Justin Ruggles ff499157a1 avformat: remove a workaround for broken timestamps 
This modifies pts in situations other than what was intended, leading to
invalid timestamps.

Reverts commit 90bb394dcc
 2012-05-03 16:28:08 -04:00
Justin Ruggles 4010d724e1 lavr: use 8.8 instead of 10.6 as the 16-bit fixed-point mixing coeff type 2012-05-01 13:52:59 -04:00
Justin Ruggles f1ffb01ee9 avplay: use libavresample for sample format conversion and channel mixing 
SDL only supports s16 sample format and a limited number of channel layouts.
Some versions of SDL on some systems support 4-channel and 6-channel output,
but it's safer overall to downmix any layout with more than 2 channels to
stereo.
 2012-05-01 13:38:23 -04:00
Justin Ruggles f7ff099997 APIchanges: fill in some dates and commit hashes 2012-04-30 13:51:41 -04:00
Justin Ruggles 394dbde548 FATE: use updated reference for aac-latm_stereo_to_51 
This uses correct stereo to 5.1 upmixing via libavresample.
 2012-04-24 23:38:54 -04:00
Justin Ruggles bcb82fe1f4 avconv: use libavresample 2012-04-24 23:38:54 -04:00
Justin Ruggles c8af852b97 Add libavresample 
This is a new library for audio sample format, channel layout, and sample rate
conversion.
 2012-04-24 21:28:27 -04:00
Justin Ruggles c5671aeb77 FATE: avoid channel mixing in lavf-dv_fmt 
This partially reverts acb1730218
which would only have needed to change the checksums if channel mixing had
been properly avoided. This changes the output file size reference and the
seek test reference back to the previous values.
 2012-04-24 15:55:45 -04:00
Justin Ruggles 83b26046fc avutil: change default buffer size alignment for sample buffer functions 
Aligning nb_samples will give both correct plane pointer alignment and enough
padding for SIMD-optimized functions.
 2012-04-23 14:06:45 -04:00
Justin Ruggles b0e9edc44f avcodec: add a cook parser to get subpacket duration 
Fixes jittery video playback of rm files with cook audio.
 2012-04-20 12:11:20 -04:00
Justin Ruggles acb1730218 FATE: allow lavf tests to alter input parameters 
Change some lavf tests to avoid resampling and channel mixing.
 2012-04-20 10:23:57 -04:00
Justin Ruggles 5052980400 FATE: replace the acodec-pcm_s24daud test with an enc_dec_pcm checksum test 
This avoids resampling and channel mixing by using a source with
the correct channel layout and sample rate.
 2012-04-20 10:23:57 -04:00
Justin Ruggles 03caef1bed FATE: replace the acodec-g726 test with 4 new encode/decode tests 
Avoids resampling and channel mixing. This only tests the behavior
with respect to input and output audio rather than also testing changes
to the encoder or muxer that do not affect the resulting decoded output.
 2012-04-20 10:23:57 -04:00
Justin Ruggles a6c8cca2a8 FATE: replace current g722 encoding tests with an encode/decode test 
Avoids resampling and channel mixing. This only tests the behavior
with respect to input and output audio rather than also testing changes
to the encoder or muxer that do not affect the resulting decoded output.
 2012-04-20 10:23:57 -04:00
Justin Ruggles 870165e217 FATE: add a pattern rule for generating asynth wav files 2012-04-20 10:23:57 -04:00
Justin Ruggles 010943c6ce FATE: optionally write a WAVE header in audiogen 2012-04-20 10:23:57 -04:00
Justin Ruggles 0c0d1bce7c avutil: add audio fifo buffer 
The functions operate on the sample level rather than the byte level and work
with all audio sample formats.
 2012-04-20 09:43:11 -04:00
Justin Ruggles 9cc338b1de FATE: pass the decoded output format and audio source file to enc_dec_pcm 
This will allow decoding to md5 and doing a diff comparison to a reference
checksum instead of a fuzzy stddev or oneoff comparison.

Signed-off-by: Mans Rullgard <mans@mansr.com>
 2012-04-18 22:26:56 -04:00
Justin Ruggles c755b1fbbc FATE: specify the input format when decoding in enc_dec_pcm() 
The output format is not always the same as the file extension,
which is sometimes required for correct probing. We can avoid
probing by specifying the format since it is already known.
 2012-04-18 22:26:49 -04:00
Justin Ruggles d8b06521a9 avconv: check for an incompatible changing channel layout 
The decoder can change the layout and channel count during decoding,
but currently we only validate that the two are compatible when opening
the codec. This checks for incompatibilities after each decoded frame.
 2012-04-18 21:02:16 -04:00
Justin Ruggles 8099fc763b riff: use bps instead of bits_per_coded_sample in the WAVEFORMATEXTENSIBLE header 
This matches the value for the plain WAVEFORMATEX header.
Also fixes stream copy to WAVE for non-16-bit raw pcm.
 2012-04-17 00:09:19 -04:00
Justin Ruggles b1041f8048 avconv: only set the "channels" option when it exists for the specified input format 
This allows the user to specify an input channel layout without avconv aborting because the
"channels" option was not found.
 2012-04-16 23:44:49 -04:00
Justin Ruggles 5e8280d177 avutil: add better documentation for AVSampleFormat 2012-04-16 15:26:55 -04:00
Justin RugglesandRonald S. Bultje de7f22ab0c ac3dsp: call femms/emms at the end of float_to_fixed24() for 3DNow and SSE 
Fixes ac3-encode and eac3-encode FATE test failures with SSE2 disabled.

Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
 2012-04-12 21:33:04 -07:00
Justin Ruggles d3c59d5003 avconv: use default channel layouts when they are unknown 
If either input or output layout is known and the channel counts match,
use the known layout for both. Otherwise choose the default layout based on
av_get_default_channel_layout().

Changed some FATE references due to some WAVE files now having a non-zero
channel mask.
 2012-04-10 11:30:01 -04:00
Justin Ruggles 95a8dac57b avconv: parse channel layout string 
This allows the user to use channel layout names instead of having to use the
channel mask values.
 2012-04-10 11:30:01 -04:00
Justin Ruggles b073819bc9 avconv: allow '-async -1' to disable timestamp sync for audio encoding 
This will allow a workaround for cases where input timestamps are invalid or
when decoder delay of 1 packet or more confuses avconv into using the wrong
timestamps as a sync reference.
 2012-04-09 13:47:11 -04:00
Justin Ruggles 679a973e97 avconv: use default alignment for audio buffer 2012-04-09 10:20:35 -04:00
Justin Ruggles c58846f3a8 avcodec: use align == 0 for default alignment in avcodec_fill_audio_frame() 
Use default alignment in audio_get_buffer()
 2012-04-09 10:20:35 -04:00
Justin Ruggles 0109a09dc3 avutil: use align == 0 for default alignment in audio sample buffer functions 2012-04-09 10:20:35 -04:00
Justin Ruggles 18ed3788b0 avutil: allow NULL linesize in av_samples_fill_arrays() and av_samples_alloc() 2012-04-09 10:20:35 -04:00
Justin Ruggles 1337de0c4b avcodec: validate the channel layout vs. channel count for decoders 
Set avctx->channel_layout to 0 if the channel count does not match
avctx->channels.
 2012-04-08 18:45:58 -04:00
Justin Ruggles 4d693b023c avutil: add av_get_packed_sample_fmt() and av_get_planar_sample_fmt() 
Based on a patch by Clément Bœsch <ubitux@gmail.com>
 2012-04-08 15:13:28 -04:00
Justin Ruggles bb0618e68b avcodec: remove avcodec_guess_channel_layout() 
It is not public because the header is not installed, and its functionality
has been replaced by av_get_default_channel_layout().
 2012-04-05 16:40:51 -04:00
Justin Ruggles 5cc51a5847 avutil: Add av_get_default_channel_layout() 
Also, use the new function in the AC-3 encoder.
 2012-04-05 16:35:14 -04:00
Justin Ruggles 67aec401d8 libspeexdec: set frame size in libspeex_decode_init() 
This fixes speex decoding, which was broken in 85469f1c.
 2012-03-30 16:30:37 -04:00
Justin Ruggles eed691f7d1 oggdec: calculate correct timestamps in Ogg/FLAC 
We need to parse the individual packet durations when there is more than one
packet in a page.
 2012-03-27 16:11:06 -04:00
Justin Ruggles 4094fc9971 avconv: fix the resampling safety factors for output audio buffer allocation 
This matches the output size required for audio_resample()
 2012-03-23 01:19:43 -04:00
Justin Ruggles 9869e963a6 avconv: move audio output buffer allocation to a separate function 
Allows for removing a goto and makes the code easier to follow.
 2012-03-23 01:19:43 -04:00
Justin Ruggles f3ab3e1aee avconv: make the async buffer global and free it in exit_program() 2012-03-23 01:19:42 -04:00
Justin Ruggles f036342b4b aiffdec: set block_duration to 1 for PCM codecs that are supported in AIFF-C 2012-03-22 11:45:46 -04:00
Justin Ruggles b38b7cc392 aiffdec: factor out handling of integer PCM for AIFF-C and plain AIFF 2012-03-22 11:45:46 -04:00
Justin Ruggles 2c07c18048 aiffdec: use av_get_audio_frame_duration() to set block_duration for AIFF-C 2012-03-22 11:45:46 -04:00
Justin Ruggles 02f88eec1d aiffdec: do not set bit rate if block duration is unknown 
CC: libav-stable@libav.org
 2012-03-22 11:45:36 -04:00
Justin Ruggles 6aba117f12 adxenc: use AVCodec.encode2() 2012-03-21 15:04:42 -04:00
Justin Ruggles 54e6cf8a94 adxenc: Use the AVFrame in ADXContext for coded_frame 2012-03-21 15:04:42 -04:00
Justin Ruggles 5d4017b8f3 vorbisenc: use AVCodec.encode2() 2012-03-21 12:49:36 -04:00
Justin Ruggles e5aab2d7a4 libvorbis: use AVCodec.encode2() 2012-03-21 12:49:36 -04:00
Justin Ruggles 8ccf545b95 libopencore-amrnbenc: use AVCodec.encode2() 2012-03-21 12:49:36 -04:00
Justin Ruggles 330d9d1b50 ra144enc: use AVCodec.encode2() 2012-03-21 12:49:36 -04:00
Justin Ruggles f090428234 nellymoserenc: use AVCodec.encode2() 2012-03-21 12:49:35 -04:00
Justin Ruggles 32173df3d2 roqaudioenc: use AVCodec.encode2() 
The first frame pts must be saved until we have 8 frames since RoQ audio
requires 8 frames in the first packet.
 2012-03-21 12:49:35 -04:00
Justin Ruggles b03dcf07f6 libspeex: use AVCodec.encode2() 2012-03-21 12:49:35 -04:00
Justin Ruggles 57a52f258e libvo_amrwbenc: use AVCodec.encode2() 2012-03-21 12:49:35 -04:00
Justin Ruggles db440fa12d libvo_aacenc: use AVCodec.encode2() 2012-03-21 12:49:35 -04:00
Justin Ruggles 27bacfeb57 wmaenc: use AVCodec.encode2() 2012-03-21 12:49:32 -04:00
Justin Ruggles b0f75ba272 mpegaudioenc: use AVCodec.encode2() 
Update FATE references due to encoder delay.
 2012-03-20 18:56:22 -04:00
Justin Ruggles 3d853d7ab3 libmp3lame: use AVCodec.encode2() 2012-03-20 18:56:18 -04:00
Justin Ruggles 1987a940b7 libgsmenc: use AVCodec.encode2() 2012-03-20 18:55:39 -04:00
Justin Ruggles d1afb2f94e libfaac: use AVCodec.encode2() 
Encoder output is delayed by several frames, so we keep a queue of input
frame timing info to match up with corresponding output packets.
 2012-03-20 18:55:36 -04:00
Justin Ruggles 59041fd053 g726enc: use AVCodec.encode2() 2012-03-20 18:47:23 -04:00
Justin Ruggles bb03b6f7b1 g722enc: use AVCodec.encode2() 
FATE reference updated due timestamp rounding because of resampling from
44100 Hz to 16000 Hz in avconv.
 2012-03-20 18:47:23 -04:00