Since c977039e58 plane count for
PIX_FMT_HWACCEL pixel formats is 0 instead of 1. The created dummy
AVBuffers are still bogus since AVFrame does not hold frame data when
AVHWAccels are used.
Also move the declaration to internal.h, and add restrict qualifiers
to the declaration (as in the implementation).
Signed-off-by: Martin Storsjö <martin@martin.st>
This allows dropping the mpegvideo dependency from a number of
components.
This also fixes standalone building of the h264 parser, which
was broken in 64e438697.
Signed-off-by: Martin Storsjö <martin@martin.st>
Number of planes is not always equal to the number of components even
for formats marked with PIX_FMT_PLANAR -- e.g. NV12 has three components
in two planes.
AVCodecContext release_buffer() shall be NULL for audio codecs using
get_buffer. The backward compatibility code hence have to check before
calling it.
Make av_get_codec_tag_string() show codec tag string characters in a more
intelligible ways. For example the ascii char "@" is used as a number, so
should be displayed like "[64]" rather than as a printable character.
Apart alphanumeric chars, only the characters ' ' and '.' are used
literally in codec tags, all the other characters represent numbers.
This also avoids relying on locale-dependent character class functions.
Signed-off-by: Martin Storsjö <martin@martin.st>
rc_buffer_size is not set before.
Solve the initial the rate control underflow issue reported in
bug 222.
CC: libav-stable@libav.org
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
Direct rendering capable decoders call get_buffer() which will set the
frame parameters.
Prevents frames with wrong parameters when a decoder outputs delayed
frames after a resolution or pixel format change.
Ensures the decoder did not set channel count to an insanely high value
during initialization, which could cause large memory usage when it tries to
get a buffer during decoding.
Avoids memleaks with audio when extended_data is nontrivial and the user
doesn't reset the frame.
Shouldn't have any effect for video for now, but might be useful in the
future.
Error out on init if a codec with CODEC_CAP_EXPERIMENTAL is requested
and strict_std_compliance is not FF_COMPLIANCE_EXPERIMENTAL.
Move the check from avconv to avcodec_open2() and return
AVERROR_EXPERIMENTAL accordingly.
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
Some invocations include a verb in the log message, others do not. Yet
av_log_missing_feature expects callers to provide a verb. Change the
function to include a verb instead and update the callers accordingly.
The result is a more natural function API and correct English in the
function invocations.
This only returns bits per sample when it is exactly correct. That is, the
codec contains only raw samples with no frame headers or padding. This applies
to basically all PCM codecs and a small subset of ADPCM codecs.
This fixes crashes in e.g. PNG decoding with SSE2 enabled. In fact, many
x86 optimizations for codecs assume that our buffer strides are 16-byte
aligned.
Wrapper around av_fast_malloc() that keeps FF_INPUT_BUFFER_PADDING_SIZE
zero-padded bytes at the end of the used buffer.
Based on a patch by Reimar Döffinger <Reimar.Doeffinger@gmx.de>.
This way, if the AVCodecContext is allocated for a specific codec, the
caller doesn't need to store this codec separately and then pass it
again to avcodec_open2().
It also allows to set codec private options using av_opt_set_* before
opening the codec.
It allows to check whether an AVCodecContext is open in a documented
way. Right now the undocumented way this check is done in lavf/lavc is
by checking whether AVCodecContext.codec is NULL. However it's desirable
to be able to set AVCodecContext.codec before avcodec_open2().
Earlier, calling avcodec_encode_audio worked fine even if time_base
wasn't set. Now it crashes due to trying to scale the output pts to
the codec context time base. This affects e.g. VLC.
If no time_base is set for audio codecs, set it to the sample
rate.
CC: libav-stable@libav.org
Signed-off-by: Martin Storsjö <martin@martin.st>
This allows audio encoders to optionally take an AVFrame as input and write
encoded output to an AVPacket.
This also adds AVCodec.encode2() which will also be usable by video and
subtitle encoders once support is implemented in the public functions.
Do not fail audio decoding with avcodec_decode_audio3 if user has set a
custom get_buffer. Strictly speaking, this was never allowed by the API,
but it seems that some software packages did so anyways. In order to
unbreak applications (cf. http://bugs.debian.org/655890), this change
clarifies the API and overrides the custom get_buffer() with the defaults.
This change is inspired by a similar
commit (c3846e3eba) in FFmpeg.
Signed-off-by: Reinhard Tartler <siretart@tauware.de>
The format is a per-frame property, having it in AVFrame simplify the
operation of extraction of that information, since avoids the need to
access the codec/stream context.
width and height are per-frame properties, setting these values in
AVFrame simplify the operation of extraction of that information,
since avoids the need to check the codec/stream context.
The sample aspect ratio is a per-frame property, so it makes sense to
define it in AVFrame rather than in the codec/stream context.
Simplify application-level sample aspect ratio information extraction,
and allow further simplifications.
Interlaced content for most codec requires it.
This patch is a stop-gap pending a serious rework to support
codecs with non 16 pixel macroblocks.
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
This was intended as an optimisation for skipped blocks in MPEG2
P-frames and never used elsewhere. Removing this "optimisation"
speeds up MPEG2 decoding by 1-2% (ARM Cortex-A9).
Signed-off-by: Mans Rullgard <mans@mansr.com>
When the buf and last pointers are equal, the FFSWAP() results
in an invalid call to memcpy() with same source and destination
on some targets. Although assigning a struct to itself is valid
C99, gcc does not check for this before calling memcpy().
See http://gcc.gnu.org/bugzilla/show_bug.cgi?id=32667
Signed-off-by: Mans Rullgard <mans@mansr.com>
Earlier, bits per sample was defined as 8, since
bits_per_coded_sample was used to indicate whether to ignore
the lower bits of the codeword, having values 6, 7 or 8.
g722 encodes 2 samples into one byte codeword, therefore the
bits per sample is 4. By changing this, the generated timestamps
for streams encoded with g722 become correct.
This makes timestamp generation for g722 data correct (both when
encoding and when demuxing from raw g722 files).
Signed-off-by: Martin Storsjö <martin@martin.st>
Add AV_NUM_DATA_POINTERS to simplify the bump transition.
This will allow for supporting more planar audio channels without having to
allocate separate pointer arrays.
A new field, AVCodecContext.internal is used to hold a new struct
AVCodecInternal, which has private fields that are not codec-specific and are
used by general libavcodec functions.
Moved internal_buffer, internal_buffer_count, and is_copy.
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>
Replace our incomplete w32threads implementation with x264's pthreads
w32threads wrapper.
Relicensed to LGPL with kind permission by Pegasys Inc.
Signed-off-by: Janne Grunau <janne-libav@jannau.net>
There is no valid reason the user should ever send such packets in the
first place, but the documentation for CODEC_CAP_DELAY states that the
codec is guaranteed not to get a NULL packet unless that capability is
set. That isn't true without preventing this case.
