Right now it doesn't use any of the extra fields AVPacket provides.
It might be wise to investigate the pts/dts ones to see if we can finally
get rid of the timing-related cruft we have.
Set the caps right after allocation of the buffer because we know the buffer is
writable then and we are correctly negotiated. Since ffmpeg keeps around
references to frames, making the buffer metadata writable where it was done
before pushing will always end up with a copy and that makes the sink do a slow
memcpy all the time.
Set caps on buffers right after we allocate them to avoid refcounting problems
and having to make the buffer metadata writable for no good reason.
Don't unmap the memory with a 0 size or we would modify the memory size when
it's not needed.
FFMpeg parsing and decoding calls require to additionally allocate bytes
at the end of the input bitstream and this padding must be initialized
to zero.
https://bugzilla.gnome.org/show_bug.cgi?id=595590
Sometimes the parser loses track of timestamps and starts to reuse old
timestamp. Feed it some dummy data and clear some context variables to work
around the problem.
When estimating the frame duration, the diff between two incomming timestamps
should be scaled by the amount of frames in the interval. Improves duration
estimation and DTS interpolation.
Use the diff between input timestamps to estimate the duration when no duration
is set on input buffers. Only do this when there are no reordered input
timestamps. Improves interpolation in DTS mode when no input duration is set.
When we flush the parser cache, we only need to clear the bytes of the cache,
not the complete state of the cache. In the case of H264 this doesn't require
the parser to receive a new SPS/PPS after a DISCONT buffer.
Add function to reset the timestamp tracking.
Check for reordered timestamps on the input buffers and assume PTS input
timestamps when we see reordered timestamps.
Recover from an occasionally wrong input timestamp by also tracking the output
timestamps. When we detect a reordered output timestamp, assume DTS input
timestamps again.
Fixes#611500
The buffer durations were not being reordered along with the timestamp
and offset of the buffers, resulting in buffers using the duration of the
latest incoming frame instead of their original frame.
Fixes#611398
When we have an input width/height that should be used for clipping, only
perform the clipping if the rectangle is smaller than the actual picture size.
Fixes#330681
Make check for vdpau decoders more generic. There might be vdpau
decoders we don't expect when using an external ffmpeg version,
and we want those blacklisted as well (e.g. ffdec_mpeg4_vdpau).
Resetting default values is currently very complex in libavcodec, so
we only call it when needed (i.e. when a context was previously used).
Shaves off 10% of the setup of a decoder.
When we are dropping frames because of QoS disable the DTS interpolation because
we won't be able to update the timestamps and end up setting the wrong
timestamps. Instead, simply use the timestamps from ffmpeg.
This now uses ffmpeg functionality to keep random metadata next to
the buffers and to get the correct offset for a frame, similar to how
timestamps are handled.
Fixes bug #578278.
Takes codec frame delay into account (roughly the same way it does for timestamps for reordered frames) to produce frames with correct offsets.
A special hack to allow trailing frame with timestamp=segment.stop to be displayed.
Fixes bug #578278.
After a DISCONT, mark the next frame with DISCONT but don't wait for a new
keyframe. This greatly improves performance on lossy networks or currupted
frames as the decoder can usually continue and conceil errors up to the next
keyframe.
Avoid an infinite loop consuming buffer timestamp info when
the video frames contain only GST_CLOCK_TIME_NONE timestamps.
Add some debug logging in the timestamp tracking paths.
Fixes: #585845
If the same instance of the plugin is asked to be initialised more that once,
instances after the first one do not register the elements properly and the
elements become not usable.
For example, if you call gst_update_registry (), is not possible to create
elements after the call since the plugin is asked to be initialised again and
does not register the elements.
Fixes#584291
The patch from Bug #580796 hacked around existing infrastructure to handle
timestamps as DTS (as in all AVI files) causing the logic to be disabled.
Properly hook the timestamp handling into the existing infrastructure to handle
these cases too, partially reverting a26b94d92c
and moving some stuff around.
Refixes #580796.
ffmpeg only tells us on a per-decoded-buffer basis if the stream is
interlaced or not. When we see a change, we force negotiation.
We can't detect that in our get_buffer() (when doing downstream allocation),
because at that point the interlaced flags aren't set on the outgoing
buffer.
Add a new function new_aligned_buffer() which creates a GstBuffer of
the requested size/caps, with the memory being allocated/freed by ffmpeg's
av_malloc/av_free which guarantees properly aligned memory.
Added a can_allocate_aligned internal property which we use to figure out
whether downstream can provide us with 128bit aligned buffers.
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.
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.
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
Original commit message from CVS:
* ext/ffmpeg/gstffmpegcodecmap.c: (gst_ff_aud_caps_new),
(gst_ffmpeg_codecid_to_caps), (gst_ffmpeg_smpfmt_to_caps),
(gst_ffmpeg_codectype_to_caps), (gst_ffmpeg_caps_to_smpfmt),
(gst_ffmpeg_caps_to_codecid), (av_smp_format_depth):
* ext/ffmpeg/gstffmpegcodecmap.h:
Add mapping for EAC3 and QCELP audio codecs.
Add conversion functions for all available audo SampleFormat.
* ext/ffmpeg/gstffmpegdec.c: (gst_ffmpegdec_open),
(gst_ffmpegdec_setcaps), (gst_ffmpegdec_negotiate),
(clip_audio_buffer), (gst_ffmpegdec_audio_frame):
Remove assumptions that we can only handle stereo 16bit signed integer
audio, and store the depth locally.
Original commit message from CVS:
reviewed by: Edward Hervey <edward.hervey@collabora.co.uk>
* ext/ffmpeg/gstffmpegdec.c: (gst_ffmpegdec_finalize):
Fix check for memory to free.
Fixes#560644
Original commit message from CVS:
* ext/ffmpeg/gstffmpegcodecmap.c: (gst_ffmpeg_avpicture_fill):
Initialize some more variables.
* ext/ffmpeg/gstffmpegdec.c: (gst_ffmpegdec_setcaps),
(alloc_output_buffer):
Disable direct rendering for h264, some functions just seem to read from
invalid memory.
Original commit message from CVS:
* ext/ffmpeg/gstffmpegdec.c: (gst_ffmpegdec_setcaps),
(gst_ffmpegdec_get_buffer), (get_output_buffer):
Enable direct rendering.
Add some more debug info about image strides.
