This fixes: ffplay -f lavfi -i cellauto
This was a regression since factorizing the filter code with ffmpeg.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Signed-off-by: Marton Balint <cus@passwd.hu>
VideoPicture pixel format is set at allocation time, therefore it is not
reflecting the proper value.
Fixes files with changing pixel format in the avfilter disabled case.
Signed-off-by: Marton Balint <cus@passwd.hu>
Switching to visualization instead of exiting ffplay is a bit more tricky, so
just exit for now.
Fixes ticket 38.
Signed-off-by: Marton Balint <cus@passwd.hu>
We do this to ensure that input_get_buffer is not called from a
frame_worker_thread of a multithreaded decoder when we already freed the
filters.
Fixes occasional segfaults on video stream change.
Signed-off-by: Marton Balint <cus@passwd.hu>
This fixes occasional segfaults caused by lock request of the packet queue from
the reader thread.
Also don't allow to put frames into the queue when it's aborted, and don't try
to fill the queue with frames when it is aborted.
Signed-off-by: Marton Balint <cus@passwd.hu>
Also use av_guess_sample_aspect_ratio for determining aspect ratio of the video
frame if not using avfilter.
Signed-off-by: Marton Balint <cus@passwd.hu>
Codec values may not reflect the actual frame size, and it also enables us to
simplify code in the avfilter enabled and the avfilter disabled case.
Signed-off-by: Marton Balint <cus@passwd.hu>
When we are using filter chains we have to set the aspect ratio of the source
to the best known value, we use the av_guess_sample_aspect_ratio function to
determine that.
Fixes ticket 1228.
Signed-off-by: Marton Balint <cus@passwd.hu>
The new lowres support is limited to decoders where lowres decoding
is possible in high quality.
I was not able to measure any speed difference, but if one is found
the 2-3 lines that might affect speed can be made compile time conditional
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This reduces the number of queued frames for audio data but also reduces the
amount of A-V difference after changing the audio stream (because less frames
are queued). Fixes bug #1035.
Signed-off-by: Marton Balint <cus@passwd.hu>
Otherwise we may use the remaining data of the last packet from the previous
audio stream. Fixes bug #951.
Signed-off-by: Marton Balint <cus@passwd.hu>
This is somewhat redundant as no decoder should call get_buffer() with such argument.
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Also change synchronize_audio to only calculate the wanted number of samples
instead of doing the actual synchronization, and make swr_convert handle the
sample addition or deletion.
This new method replaces the old buggy way which simply deleted or
multiplied samples.
Signed-off-by: Marton Balint <cus@passwd.hu>
This allows to get out of ffplay if it or SDL got stuck.
This for example happens when the audio driver is playing something
else and doesnt support mixing multiple sources.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Signed-off-by: Marton Balint <cus@passwd.hu>
Marton Balint
Michael Niedermayer
Nicolas George
Stefano Sabatini