| Commit message (Collapse) | Author | Age | Files | Lines |
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These variables are coming from mpegvideoenc where are supposedly used
as bit counters on various frame properties. However their use is
unclear as they lack documentation, are available only from a very small
subset of encoders, and they are hardly used in the wild. Also frame_bits
in aacenc is employed in a similar way.
Remove this functionality from AVCodecContex, these variable are mostly
frame properties, and too few encoders support setting them with anything
useful.
Signed-off-by: Vittorio Giovara <vittorio.giovara@gmail.com>
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Bug-Id: 761
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Signed-off-by: Vittorio Giovara <vittorio.giovara@gmail.com>
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Express bitfields more simply.
Signed-off-by: Vittorio Giovara <vittorio.giovara@gmail.com>
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Convert doxygen to multiline and express bitfields more simply.
Signed-off-by: Vittorio Giovara <vittorio.giovara@gmail.com>
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CC: libav-stable@libav.org
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Currently, the amount of padding inserted at the beginning by some audio
encoders, is exported through AVCodecContext.delay. However
- the term 'delay' is heavily overloaded and can have multiple different
meanings even in the case of audio encoding.
- this field has entirely different meanings, depending on whether the
codec context is used for encoding or decoding (and has yet another
different meaning for video), preventing generic handling of the codec
context.
Therefore, add a new field -- AVCodecContext.initial_padding. It could
conceivably be used for decoding as well at a later point.
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This was due to a miscomputation of s->cur_channel, which led to
psy-based encoders using the psy coefficients for the wrong channel.
Signed-off-by: Martin Storsjö <martin@martin.st>
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Now, nellymoserenc and aacenc no longer depends on dsputil. Independent
of this patch, wmaprodec also does not depend on dsputil, so I removed
it from there also.
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Signed-off-by: Martin Storsjö <martin@martin.st>
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This fixes segfault caused by 3d3cf6745e2a5dc9c377244454c3186d75b177fa
when SingleChannelElement.ret was renamed to SingleChannelElement.ret_buf.
Signed-off-by: Justin Ruggles <justin.ruggles@gmail.com>
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Signed-off-by: Martin Storsjö <martin@martin.st>
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Signed-off-by: Martin Storsjö <martin@martin.st>
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Move vector_fmul() from DSPContext to AVFloatDSPContext.
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Do not pointlessly call ff_alloc_packet multiple times,
and fix an infinite loop by clamping the maximum
number of bits to target in the algorithm that does
not use lambda.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
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Also break some long lines, remove codec function placeholder comments
and add spaces in sample/pixel format lists.
Signed-off-by: Martin Storsjö <martin@martin.st>
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This fixes the warning:
libavcodec/aacenc.c:524: warning: passing argument 2 of ‘deinterleave_input_samples’ discards qualifiers from pointer target type
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Signed-off-by: Martin Storsjö <martin@martin.st>
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Currently, any samples in the final frame are not decoded because they are
only represented by one frame instead of two. So we encode two final frames to
cover both the analysis delay and the MDCT delay.
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Fixes handling of CODEC_CAP_SMALL_LAST_FRAME.
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Some compilers still do not support this syntax.
Signed-off-by: Mans Rullgard <mans@mansr.com>
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Forgot to add the equivalent amount to the incoming sample pointer as the output pointer.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
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10l: Forgot to adjust deinterleave for new location of incoming samples in 7946a5a.
This produced incorrect, but surprisingly listenable results.
Thanks to Justin Ruggles for the report.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
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When the length is the escape value (15), the new length is calculated by
15 + get_bits(8) - 1.
Signed-off-by: Alex Converse <alex.converse@gmail.com>
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Signed-off-by: Alex Converse <alex.converse@gmail.com>
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incoming samples.
Signed-off-by: Alex Converse <alex.converse@gmail.com>
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Signed-off-by: Alex Converse <alex.converse@gmail.com>
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Signed-off-by: Alex Converse <alex.converse@gmail.com>
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This should be moved to tablegen at some point.
Signed-off-by: Alex Converse <alex.converse@gmail.com>
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to float.
Signed-off-by: Alex Converse <alex.converse@gmail.com>
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Signed-off-by: Alex Converse <alex.converse@gmail.com>
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Macroify sanity checks and check return values of allocs and other functions.
Signed-off-by: Alex Converse <alex.converse@gmail.com>
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They are used in lavf.
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Specifically, ff_mpeg4audio_sample_rates, ff_mpeg4audio_get_config and
ff_copy_pce_data
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The spec states:
* Only the lowest 12 spectral coefficients of any LFE may be non-zero
We were using the 12 lowest *bands*.
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10l: Forgot about TYPE_LFE being after TYPE_CPE. Which causes a
segfault when encoding 5.1
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It's more readable and less prone to breakage.
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In adjust_frame_information(), msc is incremented for each sfb in each
sub-window then compared against max_sfb which is for a single sub-window.
This resulted in frames using EIGHT_SHORT_SEQUENCE where the first few
sub-windows increment msc to a value that results in ms_mode == 2. Even
though only some of the bands are actually using Mid/Side.
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