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14657 Commits

Author SHA1 Message Date
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
Justin Ruggles
6631294c26 cook: do not needlessly set *data_size to 0 2011-10-29 13:43:28 -04:00
Justin Ruggles
b277ebd508 cook: remove pointless return statements 2011-10-29 13:43:28 -04:00
Justin Ruggles
c9c841e231 cook: simplify decouple_info() 2011-10-29 13:43:28 -04:00
Justin Ruggles
f193c96f49 cook: return appropriate error codes instead of -1 2011-10-29 13:43:28 -04:00
Justin Ruggles
e694831f9d cook: avoid hardcoded sizes in sizeof() 2011-10-29 13:43:28 -04:00
Justin Ruggles
776e9815a5 cook: remove unneeded #includes 2011-10-29 13:43:28 -04:00
Justin Ruggles
c25df22365 cook: output float samples instead of converting to int16 2011-10-29 13:43:28 -04:00
Kostya Shishkov
9a173575fd utvideo: account for coupled lines in YUV420 format 
Luma slices in YUV420 colourspace should have height in multiple of two since
they have the same line of chrominance data corresponding to pair of them.

Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
 2011-10-28 23:51:58 -07:00
Ronald S. Bultje
8370e426e4 vp3: fix oob read for negative tokens and memleaks on error. 2011-10-28 23:50:04 -07:00
Ronald S. Bultje
bfa0f96586 vp8: fix overflow in segmentation map caching. 2011-10-28 23:48:43 -07:00
Anton Mitrofanov
640d5f1c80 Fix decoding of lossless 4:2:2 H.264 
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
 2011-10-28 23:37:30 -07:00
Anton Mitrofanov
fdb5314ea7 Fix decoding of lossless 10-bit 4:4:4 H.264 
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
 2011-10-28 23:37:30 -07:00
Alex Converse
77b5c82b49 isom: Add MPEG4SYSTEMS dummy object type indication. 2011-10-28 14:54:13 -07:00
Michael Niedermayer
bc2dd36740 aacdec: allow output reconfiguration on channel changes 
Locking the decoder against channel config changes in
parse_adts_frame_header() seems to be unnecessary and
streams with channel config changes are reported.

The sample in http://roundup.libav.org/issue999 still works.

Signed-off-by: Janne Grunau <janne-libav@jannau.net>
 2011-10-28 22:44:59 +02:00
Justin Ruggles
f3db0f7403 nellymoserenc: take float input samples instead of int16 
This avoids having to convert all input data from int16 to float, which is used
internally for encoding.
 2011-10-28 14:40:52 -04:00
Justin Ruggles
77c8ef9a36 nellymoserdec: use dsp functions for overlap and windowing 2011-10-28 14:40:52 -04:00
Justin Ruggles
8c9581f052 nellymoserdec: do not fail if there is extra data in the packet 
instead just print a warning
 2011-10-28 14:40:52 -04:00
Justin Ruggles
0aaa85dbed nellymoserdec: fail if output buffer is too small 
avoids silently truncating the output
 2011-10-28 14:40:52 -04:00
Justin Ruggles
f19305fe93 nellymoserdec: remove pointless buffer size check. 2011-10-28 14:40:51 -04:00
Michael Niedermayer
8fa97302e0 snow: do not draw_edge if emu_edge is set 
Fix segfault on emu edge, to reproduce

make fate-vsynth1-snow
avplay -flags emu_edge tests/data/vsynth1/snow.avi

Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
 2011-10-28 10:14:11 -07:00
Justin Ruggles
813907d424 wmavoice: move output buffer size check to synth_superframe(). 
this allows for checking against the actual output size instead of max size.
 2011-10-28 12:02:24 -04:00
Justin Ruggles
d064076570 wmavoice: only set data_size to 0 when necessary 2011-10-28 12:02:24 -04:00
Justin Ruggles
1db6437f6c wmapro: fix strict-aliasing violations by using av_alias32 
Also fix some undefined unsigned/signed conversions.
 2011-10-28 12:02:24 -04:00
Justin Ruggles
b8b4c9c328 wmapro: use FmtConvertContext.float_interleave() to interleave output samples 2011-10-28 12:02:24 -04:00
Justin Ruggles
d0b1b1c5c7 wmadec: consolidate 2 output buffer size checks into 1 check 2011-10-28 12:02:23 -04:00
Justin Ruggles
9a33264478 apedec: assert that s->samples is not negative before trying to decode 2011-10-28 11:47:29 -04:00
Justin Ruggles
0927154d37 apedec: use FFALIGN macro for internal data buffer size 2011-10-28 11:47:28 -04:00
Justin Ruggles
5b8009f4c8 apedec: do not keep incrementing the input data pointer past the end of the 
buffer during entropy decoding.

The pointer address could overflow, which would likely segfault. Instead set
the context error flag to indicate that the decoder tried to read past the
end of the packet data.
 2011-10-28 11:47:28 -04:00
Justin Ruggles
a4c32c9a63 apedec: check for input buffer overflow while reading frame header 2011-10-28 11:47:28 -04:00
Justin Ruggles
fd244ae3a0 apedec: use unsigned int for offset 
avoids implementation-defined unsigned-to-signed conversion and simplifies
the bounds checking.
 2011-10-28 11:47:28 -04:00
Justin Ruggles
89ec474a43 apedec: remove pointless increment of 'buf' 
The variable is not used anymore at that point.
 2011-10-28 11:47:28 -04:00
Justin Ruggles
52d4fb2a3d apedec: set s->currentframeblocks after validating nblocks 2011-10-28 11:47:28 -04:00
Justin Ruggles
2cab578489 apedec: use unsigned int for 'nblocks' and make sure that it's within int range 2011-10-28 11:47:27 -04:00
Justin Ruggles
b7e5145759 apedec: do not set s->samples until after validation. 
This prevents errors and/or invalid writes in the next decode call due to
s->samples still being negative.
 2011-10-28 11:47:27 -04:00
Justin Ruggles
11ca8b2d74 apedec: check for data buffer realloc failure 2011-10-28 11:47:27 -04:00
Justin Ruggles
91b71460b5 apedec: return meaningful error values in ape_decode_frame() 2011-10-28 11:47:27 -04:00
Justin Ruggles
c6defb41ef apedec: correct an error message 2011-10-28 11:47:27 -04:00
Justin Ruggles
da55e0980e apedec: cosmetics 
break some excessively long lines and remove space after '*'
 2011-10-28 11:46:41 -04:00
Justin Ruggles
f64e0a2f37 apedec: return meaningful error codes from ape_decode_init() 2011-10-28 11:41:39 -04:00
Justin Ruggles
7500781313 apedec: check for filter buffer allocation failure 2011-10-28 11:41:39 -04:00
Justin Ruggles
b9d6b02713 apedec: use memcpy for pseudo-stereo mode 2011-10-28 11:41:39 -04:00
Justin Ruggles
3c25209bd9 apedec: remove unneeded check for zero-size packet. 
This is already checked by avcodec_decode_audio3().
 2011-10-28 11:41:39 -04:00
Justin Ruggles
ec6d743118 mp3on4: do not needlessly set data_size to 0 2011-10-27 22:06:32 -04:00
Justin Ruggles
99975966c3 mp3adu: return error instead of just consuming bad packets 2011-10-27 22:06:32 -04:00
Justin Ruggles
e2e6c8799b mpegaudiodec: check output data size based on avctx->frame_size 2011-10-27 22:06:32 -04:00
Justin Ruggles
512557b291 avcodec: remove avcodec_parse_frame and deprecate associated elements. 
The documentation for CODEC_CAP_PARSE_ONLY and AVCodecContext.parse_only
indicates that they are utilized through avcodec_parse_frame(), which was
never actually implemented.
 2011-10-27 22:06:32 -04:00
Justin Ruggles
cd816d9bbb mpegaudiodec: cosmetics: basic pretty-printing 2011-10-27 22:06:32 -04:00
Justin Ruggles
c17e534f2e mpegaudiodec: remove frame_count field from MPADecodeContext. 
Its functionality was removed several years ago, so it doesn't do anything.
AVCodecContext.frame_number could serve the same purpose if someone
wants to debug the frame count.
 2011-10-27 22:06:32 -04:00
Justin Ruggles
dac15a03af mpegaudiodec: return AVERROR return codes instead of -1 2011-10-27 22:06:32 -04:00
Justin Ruggles
4be1e1dfa7 mpegaudiodec: Skip only bad frames instead of the whole packet. 
On frame decoding failure, return an error if the frame is the same size as
the whole packet, otherwise just log an error message and return the number
of bytes consumed.
 2011-10-27 22:06:31 -04:00
Anton Khirnov
15946eb8a9 lavc: remove "legacy" mpegvideo decoder. 2011-10-27 23:06:26 +02:00
Justin Ruggles
4a6a29a7fb libopencore-amr: check output buffer size before decoding 2011-10-26 16:00:37 -04:00
Justin Ruggles
345d15d2f9 libopencore-amr: remove unneeded buf_size==0 check. 
avcodec_decode_audio3() already checks it before sending the packet to the
decoder.
 2011-10-26 16:00:37 -04:00
Justin Ruggles
402c98783d libopencore-amr: remove unneeded frame_count field. 
Use AVCodecContext.frame_number instead.
 2011-10-26 16:00:36 -04:00
Justin Ruggles
71ccfb3f14 aac_latm: remove unneeded check for zero-size packet. 
This is already checked by avcodec_decode_audio3()
 2011-10-26 12:21:18 -04:00
Justin Ruggles
f1901180e0 pcmdec: fix output buffer size check by calculating the actual output size 
prior to decoding.
 2011-10-26 12:01:07 -04:00
Justin Ruggles
154cd253e5 pcmdec: move codec-specific variable declarations to the corresponding codec 
blocks.
 2011-10-26 12:01:07 -04:00
Justin Ruggles
0093f96d34 pcmdec: return buf_size instead of src-buf. 
The values will always be the same, so this change eliminates an unneeded
variable. It also gets rid of the need to reset src when memcpy() is used.
 2011-10-26 12:01:07 -04:00
Justin Ruggles
85579b6381 avcodec: remove the Zork PCM encoder. 
The Zork PCM decoder does not decode the 1 sample we have correctly, therefore
the encoder based on the decoder is also incorrect. There is no good reason to
keep the encoder.
 2011-10-26 12:01:07 -04:00
Justin Ruggles
67a3b67c71 pcm_zork: use AV_SAMPLE_FMT_U8 instead of shifting all samples by 8. 2011-10-26 12:01:07 -04:00
Justin Ruggles
06af335a33 pcmenc: remove unneeded sample_fmt check. 
It is already checked by avcodec_open2().
 2011-10-26 12:01:07 -04:00
Justin Ruggles
d94e29cac9 pcmdec: move number of channels check to pcm_decode_init() 2011-10-26 12:01:06 -04:00
Justin Ruggles
83efd7652e pcmdec: remove unnecessary check for sample_fmt change 2011-10-26 12:01:06 -04:00
Justin Ruggles
381e195b46 pcmdec: move DVD PCM bits_per_coded_sample check near to the code that sets 
the sample size.
 2011-10-26 12:01:06 -04:00
Justin Ruggles
6b94711f15 pcmdec: do not needlessly set *data_size to 0 2011-10-26 12:01:06 -04:00
Justin Ruggles
30f3e7b524 alacdec: remove unneeded NULL or zero-size packet checks. 
This is already done in avcodec_decode_audio3()
 2011-10-26 11:50:17 -04:00
Justin Ruggles
68f7e9cd8e alacdec: simplify buffer allocation by using FF_ALLOC_OR_GOTO() 2011-10-26 11:50:17 -04:00
Justin Ruggles
b316af7a7c alacdec: ask for a sample for unsupported sample depths. 
Also return AVERROR_PATCHWELCOME.
 2011-10-26 11:50:17 -04:00
Justin Ruggles
63cf54df7a alacdec: cosmetics: use 'ch' instead of 'chan' to iterate channels 2011-10-26 11:50:17 -04:00
Justin Ruggles
01200f1283 alacdec: move some declarations to the top of the function 2011-10-26 11:50:17 -04:00
Justin Ruggles
c3a92412c0 alacdec: always use get_sbits_long() for uncompressed samples 2011-10-26 11:50:17 -04:00
Justin Ruggles
b46e58f741 alacdec: remove unneeded local variable 2011-10-26 11:50:17 -04:00
Justin Ruggles
7080533cda alacdec: remove the numchannels parameter from several functions. 
They only operate on stereo content, so the extra param is not necessary and
also allows for simplifying the code.
 2011-10-26 11:50:17 -04:00
Justin Ruggles
cb50329fc5 alacdec: rename 2 functions. 
Now they only do stereo interleaving.
 2011-10-26 11:50:16 -04:00
Justin Ruggles
c39bddd392 alacdec: move appending of extra_bits to a separate function. 
This should also fix decoding of mono 24-bit.
 2011-10-26 11:50:16 -04:00
Justin Ruggles
e739d35156 alacdec: split stereo decorrelation into a separate function. 
It is identical for 16-bit and 24-bit, so there is no need to have duplicate
code.
 2011-10-26 11:50:16 -04:00
Justin Ruggles
d251c85dce alacdec: cosmetics: rename 'wasted_bits' to 'extra_bits'. 
The bits are not wasted, they are additional low bits that are added to the
16-bit decompressed samples to increase the output sample depth.
 2011-10-26 11:50:16 -04:00
Justin Ruggles
dbbb9262ca alacdec: remove unneeded numsamples checks 2011-10-26 11:50:16 -04:00
Justin Ruggles
53df079a73 alacdec: check for buffer allocation failure. 
Also rearranges some functions for easier cleanup on failure.
 2011-10-26 11:50:16 -04:00
Justin Ruggles
e5e4f92b5c alacdec: allocate per-channel buffers based on channel count. 
reduces memory usage when the stream has fewer than MAX_CHANNELS
 2011-10-26 11:50:16 -04:00
Justin Ruggles
dcaa83a0fc alacdec: read/validate number of channels from the extradata. 
check frame header channel count against header/container channel count.
 2011-10-26 11:50:16 -04:00
Justin Ruggles
47e9c75b36 alacdec: remove unneeded validation of setinfo_sample_size. 
It is already done when using it to set sample_fmt.
 2011-10-26 11:50:16 -04:00
Justin Ruggles
0f26f3d5c4 alacdec: set sample_fmt in alac_decode_init() 2011-10-26 11:50:16 -04:00
Justin Ruggles
aec8383348 alacdec: set bytespersample using av_get_bytes_per_sample() 2011-10-26 11:50:15 -04:00
Janne Grunau
d6174bfe5f threads: restore has_b_frames in frame_thread_free 
Otherwise the delay expressed in has_b_frames increases with every
avcodec_close/avcodec_open.
Fixes fate-ea-dct with more than 1 thread.
 2011-10-26 16:55:54 +02:00
Daniel Kang
ded3e9f054 H.264: Cometics to dsputil_mmx.c 
Add whitespace.

Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
 2011-10-26 06:41:32 -07:00
Justin Ruggles
a3a8572165 g722dec: check output buffer size before decoding 2011-10-25 11:30:50 -04:00
Justin Ruggles
4e41973794 g722dec: cosmetics: reindent/linewrap 2011-10-25 11:30:50 -04:00
Justin Ruggles
d0a196962a g722dec: remove the use of lowres for half-rate decoding. 
It is broken because an AVCodecContext can be opened/closed multiple
times, and sample_rate is getting divided by 2 each time that happens.

This removes the only use of lowres for audio.
 2011-10-25 11:30:50 -04:00
Justin Ruggles
2f1d212fd0 tta: check for allocation failure of decode_buffer 2011-10-25 11:22:02 -04:00
Justin Ruggles
b5050539c9 tta: use correct frame_length calculation. 
using a floating-point calculation is not necessary.
 2011-10-25 11:22:02 -04:00
Justin Ruggles
c6056d4004 tta: add support for decoding 24-bit sample format 
Note that this will not work in most cases with avconv and avplay due to the
AVCODEC_MAX_AUDIO_FRAME_SIZE limit, but it will decode correctly if given a
large enough output buffer.
 2011-10-25 11:22:02 -04:00
Justin Ruggles
8664682d0e cosmetics: indentation 2011-10-25 11:22:02 -04:00
Justin Ruggles
7b7a74a150 tta: remove pointless braces 2011-10-25 11:22:02 -04:00
Justin Ruggles
e6923f683c tta: check output buffer size after adjusting frame length for last frame 2011-10-25 11:22:01 -04:00
Justin Ruggles
b16960a8a5 tta: fix reading of format in TTA header. 
TTA does not support float at all, and format 2 is encrypted TTA.
 2011-10-25 11:22:01 -04:00
Justin Ruggles
4d3e7a7516 tta: remove useless commented-out lines 2011-10-25 11:22:01 -04:00
Justin Ruggles
35f9d8c20a tta: check remaining bitstream size while reading unary value 2011-10-25 11:22:01 -04:00
Janne Grunau
28287045ca cosmetics: simplify latm_decode_init 2011-10-25 12:08:21 +02:00
Janne Grunau
785f876cee latm: avoid unnecessary reinit of the aac decoder 2011-10-25 12:08:21 +02:00