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.