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14657 Commits

Author SHA1 Message Date
Justin Ruggles
2467d8d9ea ra288: return error if input buffer is too small 2011-11-08 12:36:56 -05:00
Justin Ruggles
0131e70af5 ra288: utilize DSPContext.vector_fmul() 2011-11-08 12:36:48 -05:00
Justin Ruggles
03e5d6118c ra288: use memcpy() to copy decoded samples to output 2011-11-08 12:36:41 -05:00
Justin Ruggles
f50b6be57d mace: only calculate output buffer size once 2011-11-08 12:36:14 -05:00
Diego Biurrun
ce33320b30 Remove redundant filename self-references inside files. 
Filenames are brittle across renames and add no useful information.
 2011-11-08 17:52:56 +01:00
Diego Biurrun
9412f81138 indeo3data: add missing config.h #include for HAVE_BIGENDIAN 2011-11-08 17:52:56 +01:00
Diego Biurrun
276b995d85 x86: drop pointless ARCH_X86 #ifdef from files in x86 subdirectory 2011-11-08 17:52:55 +01:00
Kostya Shishkov
f545e00677 BMV demuxer and decoder 
Signed-off-by: Martin Storsjö <martin@martin.st>
 2011-11-08 00:36:45 +02:00
Alex Converse
090aaaf752 mpeg12enc: Remove write-only variables. 2011-11-07 10:53:55 -08:00
Alex Converse
7c5dfc174b mpeg12enc: Don't set up run-level info for level 0. 
run: The number of zero coefficients preceding a non-zero coefficient,
in the scan order. The absolute value of the non-zero coefficient is
called "level".

The run-level code makes illegal reads when trying to set up tables for
nonsense level 0.
 2011-11-07 10:50:48 -08:00
Alex Converse
a1684cf82d msmpeg4: Don't set up run-level info for level 0. 
run: The number of zero coefficients preceding a non-zero coefficient,
in the scan order. The absolute value of the non-zero coefficient is
called "level".

The run-level code makes illegal reads when trying to set up tables for
nonsense level 0.
 2011-11-07 10:48:53 -08:00
Justin Ruggles
b8f02f5b4e dsputil: use cpuflags in x86 versions of vector_clip_int32() 2011-11-06 20:50:06 -05:00
Kostya Shishkov
66760b30a3 cosmetics: insert some spaces in explicit enum value assignments 
Signed-off-by: Anton Khirnov <anton@khirnov.net>
 2011-11-06 08:13:11 +01:00
Kostya Shishkov
19900d60fd move 8SVX audio codecs to the audio codec list part on the next bump 
Signed-off-by: Anton Khirnov <anton@khirnov.net>
 2011-11-06 08:12:42 +01:00
Kostya Shishkov
0e288b8c52 deprecate codec IDs that won't ever be used 
Signed-off-by: Anton Khirnov <anton@khirnov.net>
 2011-11-06 07:41:58 +01:00
Ronald S. Bultje
717401aff2 h264_weight: remove duplication functions. 2011-11-05 07:16:30 -07:00
Ronald S. Bultje
ea2bb12e3e h264: improve calculation of codec delay. 
Fixes the following conformance suite samples:
HCBP1_HHI_A.264, HCBP2_HHI_A.264, HCMP1_HHI_A.264 (main)
HCHP1_HHI_B.264, HCHP2_HHI_A.264, HCHP3_HHI_A.264 (frext)
 2011-11-05 06:58:52 -07:00
Martin Storsjö
8148631269 w32threads: Wrap the mutex functions in inline functions returning int 
This allows using these wrappers in the gcrypt mutex callbacks.

Signed-off-by: Martin Storsjö <martin@martin.st>
 2011-11-05 12:09:24 +02:00
Martin Storsjö
2d1b6fb72b avcodec: Allow locking and unlocking an avformat specific mutex 
This extends the lock manager in avcodec to manage two separate
mutexes via the user-specified lock functions.

Signed-off-by: Martin Storsjö <martin@martin.st>
 2011-11-05 12:08:53 +02:00
Justin Ruggles
add7b1140f binkaudio: expand quant_table to accommodate all possible values 2011-11-04 10:23:53 -04:00
Martin Storsjö
c38404ee1a libx264: Set the default of the rc_lookahead option to -1 
This allows it to use the defaults specified by preset/tune,
without overwriting it with the default value from the
AVCodecContext field.

Signed-off-by: Martin Storsjö <martin@martin.st>
 2011-11-04 12:46:35 +02:00
Martin Storsjö
adc85ce20b avcodec: Set flags2 default value depending on availability 
This makes the code compile when FF_API_X264_GLOBAL_OPTS or
FF_API_LAME_GLOBAL_OPTS is 0.

Signed-off-by: Martin Storsjö <martin@martin.st>
 2011-11-04 12:46:33 +02:00
Alex Converse
2a6eb06254 vp6: Fix illegal read. 2011-11-03 16:40:34 -07:00
Martin Storsjö
cae4f4b77e avcodec: Make sure codec_type is set by avcodec_get_context_defaults2 
This function used to set codec_type. With the current fallback
implementation based on avcodec_get_context_defaults3, codec_type
won't be set to the value passed in, but will be set to
AVMEDIA_TYPE_UNKNOWN. Legacy callers of this function might expect
this field to be set to the value passed in.

Signed-off-by: Martin Storsjö <martin@martin.st>
 2011-11-03 13:29:48 +02:00
Martin Storsjö
1b6da627d4 avcodec: Remove a misplaced and useless attribute_deprecated 
If attribute_deprecated is used in an enum declaration, it
should follow the 'enum' keyword, otherwise it's ignored
silently. This is the only case of attribute_deprecated for
enum declarations currently.

Currently, this attribute_deprecated doesn't have any effect.
If moved to the right place, it emits a warning every single
time avcodec.h is included, like this:

avcodec.h:2827: warning: ‘AVLPCType’ is deprecated (declared at avcodec.h:543)

There is already a working attribute_deprecated for the
corresponding field in AVCodecContext, so therefore this
one shouldn't be needed.

Signed-off-by: Martin Storsjö <martin@martin.st>
 2011-11-03 09:47:57 +02:00
Justin Ruggles
5463e83dbc fmtconvert: fix int32_to_float_fmul_scalar() for windows x86_64 
The calling convention only allows 4 non-stack parameter, with each
float or int register being skipped if not used.

fixes Bug 64
 2011-11-02 21:44:58 -04:00
Maxim Poliakovski
594b54b51e replacement Indeo 3 decoder 
The new decoder is much smaller and has better code quality.
Cleanup and fixes courtesy of Kostya Shishkov.

Signed-off-by: Diego Biurrun <diego@biurrun.de>
 2011-11-03 00:59:12 +01:00
Justin Ruggles
c2d9a65bc0 ac3enc: doxygen update. 
Add some parameters to existing function documentation.
Remove some unneeded documentation.
Convert some static function documentation to non-doxygen style.
 2011-11-02 17:21:45 -04:00
Justin Ruggles
a4998e448f imc: return error codes instead of 0 for error conditions. 
This fixes a bug where the whole buffer was returned as decoded audio due to
*data_size not being set to zero and the return value being >= 0.
 2011-11-02 17:02:22 -04:00
Justin Ruggles
08e5cd3810 imc: return meaningful error codes instead of -1 2011-11-02 17:02:22 -04:00
Justin Ruggles
0473f29b60 imc: do not set channel layout for stereo 
we only support decoding of mono imc
 2011-11-02 17:02:22 -04:00
Justin Ruggles
7b7f47e733 imc: validate channel count 
ask for a sample if not mono
 2011-11-02 17:02:22 -04:00
Justin Ruggles
95fee70d67 imc: check for ff_fft_init() failure 2011-11-02 17:02:22 -04:00
Justin Ruggles
86962b13f6 imc: check output buffer size before decoding 2011-11-02 17:02:22 -04:00
Justin Ruggles
e9362aaedf imc: use DSPContext.bswap16_buf() to byte-swap packet data 2011-11-02 17:02:22 -04:00
Justin Ruggles
20e081dddc libgsm: add flush function to reset the decoder state when seeking 2011-11-02 14:41:17 -04:00
Justin Ruggles
480324e7ca libgsm: simplify decoding by using a loop 2011-11-02 14:41:17 -04:00
Justin Ruggles
9d52f0a711 gsm: log error message when packet is too small 2011-11-02 14:41:17 -04:00
Justin Ruggles
9671db8245 libgsmdec: do not needlessly set *data_size to 0 2011-11-02 14:41:16 -04:00
Justin Ruggles
a2e255783e gsmdec: do not needlessly set *data_size to 0 2011-11-02 14:41:16 -04:00
Justin Ruggles
fc43fc9faa gsmdec: add flush function to reset the decoder state when seeking 2011-11-02 14:41:16 -04:00
Justin Ruggles
b03761b130 libgsmdec: check output buffer size before decoding 2011-11-02 14:41:16 -04:00
Justin Ruggles
bac2597a32 gsmdec: log error message when output buffer is too small. 
also return AVERROR(EINVAL) instead of -1
 2011-11-02 14:41:16 -04:00
Justin Ruggles
d9c6eece21 gsm: use av_get_bytes_per_sample() in frame_bytes calculation 2011-11-02 14:41:16 -04:00
Diego Biurrun
f36b390275 Replace some forgotten FFmpeg references by Libav. 2011-11-02 10:42:55 +01:00
Diego Biurrun
2f5df0b12c Replace ffmpeg references with more accurate libav* references. 2011-11-02 10:42:55 +01:00
Diego Biurrun
20566eb0f0 Replace outdated references to ffmpeg tool with avconv. 2011-11-02 10:42:54 +01:00
Diego Biurrun
124e28847b Remove some stray unnecessary ffmpeg references. 2011-11-02 10:42:54 +01:00
Diego Biurrun
d1dfcb0829 vp3: remove some pointless comments 2011-11-02 10:42:54 +01:00
Anton Khirnov
5511ad14fe lavc: use designated initialisers for parsers. 2011-11-02 10:03:43 +01:00
Justin Ruggles
da24963725 g726dec: add flush() function to reset state when seeking 2011-11-01 21:23:04 -04:00
Justin Ruggles
97f5dd1d84 g726: don't pass index to g726_reset() 
calculate it from c->code_size instead.
 2011-11-01 21:23:04 -04:00
Justin Ruggles
615b2a2cf5 g726enc: add private option for setting code size directly. 
This is an easy alternative to setting bit_rate. This patch also selects the
closest bit_rate to the requested one rather than requiring an exact value.
 2011-11-01 21:23:04 -04:00
Justin Ruggles
7abb73d4ba g726: wrap the decoder functions with a CONFIG_ADPCM_G726_DECODER check 2011-11-01 21:23:04 -04:00
Justin Ruggles
437c11ca16 g726: group the g726_encoder AVCodec with the other encoding functions 2011-11-01 21:23:04 -04:00
Justin Ruggles
50969c0f46 g726: return AVERROR(EINVAL) instead of -1 for invalid channel count 2011-11-01 21:23:03 -04:00
Justin Ruggles
50c466d609 g726enc: use av_assert0() for sample_rate validation 
This should never happen, but the check avoids a divide-by-zero.
 2011-11-01 21:23:03 -04:00
Justin Ruggles
9e78d8cfdf g726: treat sample rates other than 8kHz as unofficial. 2011-11-01 21:23:03 -04:00
Justin Ruggles
6e8d4a7afb g726dec: remove the sample_rate validation 2011-11-01 21:23:03 -04:00
Justin Ruggles
6ac34eed54 g726: use bits_per_coded_sample instead of bitrate to determine mode 
This requires some workarounds in the WAV muxer and demuxer. We need to write
the correct bits_per_coded_sample and block_align in the muxer. In the
demuxer, we cannot rely on the bits_per_coded_sample value, so we use the bit
rate and sample rate to determine the value.

This avoids having the decoder rely on AVCodecContext.bit_rate, which is not
required to be set by the user for decoding according to our API.
 2011-11-01 21:23:03 -04:00
Justin Ruggles
d405237bae g726: split the init function for the encoder and decoder 
This also allows for not having a decoder close function.
 2011-11-01 21:23:03 -04:00
Justin Ruggles
c8d36d254e g726: pre-calculate the number of output samples. 
Allows for checking output buffer size and simplification of decoding loop.
 2011-11-01 21:23:03 -04:00
Justin Ruggles
e61a670b53 g726: use int16_t instead of short 2011-11-01 21:23:02 -04:00
Diego Biurrun
45235d69c2 libdirac/libschroedinger: Drop unnecessary symbol prefixes. 
The names used in the libdirac/libschroedinger wrappers are long enough as-is.
Bloating them with unnecessary prefixes makes them even more unwieldy.
 2011-10-30 21:40:52 +01:00
Justin Ruggles
7d1b17b833 cin audio: use sign_extend() instead of casting to int16_t 2011-10-29 16:43:40 -04:00
Justin Ruggles
405af43104 cin audio: restructure decoding loop to avoid a separate counter variable 
Also check output buffer size instead of truncating output.
 2011-10-29 16:43:40 -04:00
Justin Ruggles
859bdc33e4 cin audio: use local variable for delta value 2011-10-29 16:43:40 -04:00
Justin Ruggles
64e19ba48b cin audio: remove unneeded cast from void* 2011-10-29 16:43:40 -04:00
Justin Ruggles
03381c12b3 cin audio: validate the channel count 2011-10-29 16:43:40 -04:00
Justin Ruggles
664eb77dc3 cin audio: remove unneeded AVCodecContext pointer from CinAudioContext 2011-10-29 16:43:40 -04:00
Justin Ruggles
5bd0343bee flacdec: use av_get_bytes_per_sample() to get sample size 2011-10-29 16:05:25 -04:00
Justin Ruggles
272fcc32bb dca: handle errors from dca_decode_block() 
Return error if core block decoding fails.
Do not enable XCh if XCh extension block decoding fails.
 2011-10-29 16:04:07 -04:00
Justin Ruggles
aae6eead6a dca: return error if the frame header is invalid 2011-10-29 16:04:07 -04:00
Justin Ruggles
f44059d260 dca: return proper error codes instead of -1 2011-10-29 16:04:06 -04:00
Kostya Shishkov
46e1af3b0f utvideo: handle empty Huffman trees 
If the frame is filled with the same colour, encoder may produce no data
and the fill value is indicated by zero code length (the rest of symbols
will have 0xFF for code length, meaning invalid).  So such Huffman trees
should be treated specially.

Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
 2011-10-29 12:54:08 -07:00
Justin Ruggles
425a843505 binkaudio: change short to int16_t 2011-10-29 15:16:54 -04:00
Justin Ruggles
4f4e19480a binkaudio: only decode one block at a time. 
This prevents truncating output due to an output buffer that is too small for
all blocks. There is no limit on the number of blocks in a packet.
 2011-10-29 15:16:54 -04:00
Justin Ruggles
eaddd29e00 binkaudio: store interleaved overlap samples in BinkAudioContext. 
This fixes the requirement for the buffer size to be larger than the number of
samples actually decoded.
 2011-10-29 15:16:54 -04:00
Justin Ruggles
9f48039a37 binkaudio: pre-calculate quantization factors 2011-10-29 15:16:53 -04:00
Justin Ruggles
101ef19ef4 binkaudio: add some buffer overread checks. 
This stops decoding before overreads instead of after.
 2011-10-29 15:16:53 -04:00
Justin Ruggles
2073224697 atrac3: support float or int16 output using request_sample_fmt 2011-10-29 15:06:32 -04:00
Justin Ruggles
a047851329 atrac3: add CODEC_CAP_SUBFRAMES capability 
the decoder can handle multiple frames in a packet
 2011-10-29 15:06:32 -04:00
Justin Ruggles
8f98577d4d atrac3: return appropriate error codes instead of -1 2011-10-29 15:06:32 -04:00
Justin Ruggles
47b617021d atrac3: make sure all memory is freed on init failure 2011-10-29 15:06:32 -04:00
Justin Ruggles
c91613857d atrac3: add a couple macro constants 2011-10-29 15:06:31 -04:00
Justin Ruggles
1fead73d7b atrac3: return error if packet is too small 2011-10-29 15:06:31 -04:00
Justin Ruggles
7e4881a2d0 atrac3: check output buffer size before decoding 2011-10-29 15:06:31 -04:00
Justin Ruggles
6ba7f78bbb atrac3: use separate pointers for each channel in decodeFrame() 2011-10-29 15:06:31 -04:00
Justin Ruggles
5e76b8bb76 atrac3: use optimized float_interleave() function for stereo interleaving 2011-10-29 15:06:31 -04:00
Justin Ruggles
8af33cb38a atrac3: decode mono directly to the output buffer 2011-10-29 15:06:31 -04:00
Justin Ruggles
c4a6fde33f atrac3: decode output to float samples instead of converting to s16 2011-10-29 15:06:31 -04:00
Justin Ruggles
f20dd574f1 atrac1: return appropriate error codes instead of -1 2011-10-29 15:06:31 -04:00
Justin Ruggles
6dc7dd7af4 atrac1: check for ff_mdct_init() failure 2011-10-29 15:06:31 -04:00
Justin Ruggles
21dcecc310 atrac1: use optimized float_interleave() function for stereo interleaving 2011-10-29 15:06:31 -04:00
Justin Ruggles
96b5702efe atrac1: fix a typo 2011-10-29 15:06:31 -04:00
Justin Ruggles
bff5b2c1ca atrac1: validate number of channels 2011-10-29 15:06:31 -04:00
Justin Ruggles
9a35ff3841 atrac1: decode mono audio directly to output buffer 
For stereo we need to use intermediate planar buffers, but mono does not need
to be deinterleaved so the output buffer can be used directly.
 2011-10-29 15:06:31 -04:00
Justin Ruggles
33684b9c12 atrac1: check output buffer size before decoding 2011-10-29 15:06:30 -04:00
Justin Ruggles
5c353eb8e3 cook: return AVERROR_PATCHWELCOME instead of ENOTSUP 
ENOTSUP is not defined on some systems
 2011-10-29 14:32:55 -04:00
Justin Ruggles
e34c6c9708 cook: check output buffer size before decoding 2011-10-29 13:43:28 -04:00