| Commit message (Collapse) | Author | Age |
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Remove unused function argument.
Return an AVERROR.
Don't zero out ICS on failure.
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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.
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Deprecate avcodec_decode_audio3().
Implement audio support in avcodec_default_get_buffer().
Implement the new audio decoder API in all audio decoders.
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Fixes File1.aac and Bug 80.
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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>
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This is already checked by avcodec_decode_audio3()
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use AVCodecContext.err_recognition
Signed-off-by: Anton Khirnov <anton@khirnov.net>
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Specifically, ff_mpeg4audio_sample_rates, ff_mpeg4audio_get_config and
ff_copy_pce_data
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It's used in lavf.
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Since nnz can be zero, this is needed to avoid a shift by 32.
Signed-off-by: Mans Rullgard <mans@mansr.com>
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configured.
Long term it would be nice to support error resilient reconfiguration
but right now setting this every frame does more harm than help.
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
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Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
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Audio found is not triggered on a CCE because a CCE alone has no output.
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
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It's more readable and less prone to breakage.
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This fixes AAC playback in ffplay.
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av_get_bits_per_sample_fmt() is deprecated.
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Fixes memory leaks which are the result of overwriting already-initialized
MDCT contexts during context reinitialization, e.g. in valgrind
fate-aac-latm_000000001180bc60.
Signed-off-by: Diego Biurrun <diego@biurrun.de>
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decoders.
Based on patches by clsid2 in ffdshow-tryout.
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Instead, scalefactors are adjusted by the offset amount, removing the need
for sf_scale, and the MDCT scales are adjusted to compensate for the higher
scalefactors. Floating-point output will be handled by modifying the MDCT
scales.
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table values from the spec.
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of hardcoding 200 everywhere.
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erroring out. A magnitude of 100 corresponds to 2^25 so the will most
likely result in clipped output anyway.
None of the conformance streams fall in the range that need to be clipped.
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On Wed, Apr 20, 2011 at 11:39 AM, Justin Ruggles
<justin.ruggles@gmail.com> wrote:
> On 04/20/2011 02:26 PM, Alex Converse wrote:
>
>> ---
>> libavcodec/aacdec.c | 10 +++++++++-
>> 1 files changed, 9 insertions(+), 1 deletions(-)
>>
>>
>>
>> 0002-Add-some-Debug-log-messages-to-AAC-extradata.patch
>>
>>
>> diff --git a/libavcodec/aacdec.c b/libavcodec/aacdec.c
>> index c9761a1..3ec274f 100644
>> --- a/libavcodec/aacdec.c
>> +++ b/libavcodec/aacdec.c
>> @@ -79,7 +79,6 @@
>> Parametric Stereo.
>> */
>>
>> -
>> #include "avcodec.h"
>> #include "internal.h"
>> #include "get_bits.h"
>
>
> stray whitespace change
>
oops, fixed
>From 94e8d0eea77480630f84368c97646cabc0f50628 Mon Sep 17 00:00:00 2001
From: Alex Converse <aconverse@google.com>
Date: Wed, 20 Apr 2011 11:23:34 -0700
Subject: [PATCH] Add some debug log messages to AAC extradata
MIME-Version: 1.0
Content-Type: multipart/mixed; boundary="------------1"
This is a multi-part message in MIME format.
--------------1
Content-Type: text/plain; charset=UTF-8; format=fixed
Content-Transfer-Encoding: 8bit
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Signed-off-by: Diego Biurrun <diego@biurrun.de>
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According to ISO 9899:1999 S 6.5.7/4:
The result of E1 << E2 is E1 left-shifted E2 bit positions; vacated bits
are filled with zeros. If E1 has an unsigned type, the value of the
result is E1× 2^E2, reduced modulo one more than the maximum value
representable in the result type. If E1 has a signed type and
nonnegative value, and E1× 2^E2 is representable in the result type, then
that is the resulting value; otherwise, the behavior is undefined.
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This reverts commit 36864ac3540445c513484017aa9927e942fac24a since it
breaks LATM decoding in ffplay.
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These windows do not really belong in fft/mdct files and were
easily confused with the similarly named tables used by rdft.
Signed-off-by: Mans Rullgard <mans@mansr.com>
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This function is not tightly coupled to mdct, and it's in the way
of making a fixed-point mdct implementation.
Signed-off-by: Mans Rullgard <mans@mansr.com>
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This removes the rather pointless wrappers (one not even inline)
for calling the fft_calc and related function pointers.
Signed-off-by: Mans Rullgard <mans@mansr.com>
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Signed-off-by: Mans Rullgard <mans@mansr.com>
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When decoding latm config, use the corresponding aac context and its
m4ac instead of using NULL and a local variable. This fixes decoding of
audio in MPEG TS from SBTVD (the Brazillian Digital TV Sytem), when
there is no extradata. This is the case when using the decoder with
gst-ffmpeg and a GStreamer mpegts demuxer.
Signed-off-by: Thadeu Lima de Souza Cascardo <cascardo@holoscopio.com>
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
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It was doubled in size for the LTP implementation. This brings it back
down to its original size.
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Ported from gsoc svn.
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Tested to compile with lavc major bump.
Signed-off-by: Mans Rullgard <mans@mansr.com>
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This will be beneficial for use with the audio conversion API without
requiring it to depend on all of dsputil.
Signed-off-by: Mans Rullgard <mans@mansr.com>
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DSPContext.vector_fmul_window()
DCADSPContext.lfe_fir()
SynthFilterContext.synth_filter_float()
Signed-off-by: Mans Rullgard <mans@mansr.com>
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Signed-off-by: Mans Rullgard <mans@mansr.com>
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