We simply allocate the memory using ffmpeg's av_malloc which provides us
with properly memalign'ed data.
This avoids write-outside-of-bounds when sse/altivec code is being used.
The internal resampling functions seem to require a slightly bigger buffer
for output than what we require. Therefore we give it an extra 64bytes (although
16 should have been enough).
We should post a STREAM DECODE error message on the bus when we return
GST_FLOW_ERROR, otherwise the user ends up seeing an ugly internal flow
error message, which isn't very nice.
The problem is that the ffmpeg aac decoder fails... but still accepts
the following buffers as if nothing happened. But because some things
were not properly set in the internal code, all hell breaks loose.
AVOutputFormat does *NOT* contain the full list of codecs a muxer can handle,
but does contain the recommended audio and video codecs. Therefore we use that
information to expose more muxers, until AVOutputFormat contains a list of
*ALL* compatible codecs.
For a given AVCodec, when the sample_fmts field is non-NULL, that means that
that codec can only handle a specific set of SampleFormat.
With this patch, we now look for its presence and create the proper pad template
caps.
Fixes#569441
Use a separate helper directory to build ffmpeg distributables
rather than replacing and corrupting (no more .svn dirs)
the existing checkout used for standard make/building.
Bring make dist in sync with autogen.sh's retrieval of ffmpeg
checkout, which also includes an update to selected revision
of libswscale external. Also include *.S files (needed for
e.g. ARM build).
Where no more data is available, av_read_frame just returns an error code
instead of making the difference between "I am not returning anything because
we finished reading" and "I am not returning anything because the underlying
read failed".
We differentiate between the two by looking at whether we outputted any
data previously or not.
Original commit message from CVS:
Patch by: Dejan Sakelšak <sakdean at gmail dot com>
* ext/ffmpeg/gstffmpegcodecmap.c: (gst_ff_aud_caps_new):
Narrow down the allowed channels and sample rates for AMR.
Fixes#566647.
