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>
Use codec aspect ratio for frame aspect ratio if AVFrame is NULL.
Signed-off-by: Marton Balint <cus@passwd.hu>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
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>
Guesses the sample aspect ratio of a frame, based on both the stream and the
frame aspect ratio.
Since the frame aspect ratio is set by the codec but the stream aspect ratio
is set by the demuxer, these two may not be equal. This function tries to
return the value that you should use if you would like to display the frame.
Basic logic is to use the stream aspect ratio if it is set to something sane
otherwise use the frame aspect ratio. This way a container setting, which is
usually easy to modify can override the coded value in the frames.
Signed-off-by: Marton Balint <cus@passwd.hu>
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>
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>
The new version returns AVERROR(EINVAL) is the specified paramters are invalid,
and also creates the resampler if none was used so far.
Signed-off-by: Marton Balint <cus@passwd.hu>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This patch reimplements early frame drop, it is now based on the current
difference between the master clock and the video clock, and the pts of the
current and the last displayed (or skipped) frame. If the frame to be added to
the queue is late after decoding, then we drop it early because later we would
drop it anyway (unless it is the only frame in the picture queue).
The current approach has only one downside that I know of, it does not handle
well when the filters are changing significantly the pts of the frames, because
we compare pts values from filtered and unfiltered frames.
We also start using the pictq_mutex to ensure consistent video_current_pts,
video_current_pts_drift, frame_last_pts, frame_last_dropped_pts and
frame_last_dropped_pos values.
Signed-off-by: Marton Balint <cus@passwd.hu>
Fixes missing blue channel when switching from/to fullscren on OSX and libsdl
1.2.14. Fixes issue 548. Thanks for Jean First for the original patch and
for testing.
Signed-off-by: Marton Balint <cus@passwd.hu>
If the picture queue is empty, or when the calculated delay is 0, frame_timer
is not increased but we are still displaying the old frame. When we eventually
get a frame, where the computed delay is positive, so we don't need to drop any
more frames, then it is best to update frame_timer to be as near as the current
time as it can.
This way we dont't have to wait several frames to add the necesarry delays to
frame_timer to reach current time, therefore there are no extra frame drops
after reaching a positive delay.
Signed-off-by: Marton Balint <cus@passwd.hu>
The current impementation of early frame drops (dropping frames before adding
them to the picture queue) has multiple problems:
Even after gettin A-V sync, the frame droping continues until
VideoState->skip_frames reaches 1, which can take a lot of time causing useless
additional frame drops and bad AV-sync. This issue can be easily triggered with
for example changing the audio stream.
Also video_refresh currenly does not handle early skipped frames in every case,
for example if we skip a frame, then the last frame duration calculation will
compute the duration of the sum of the skipped frame and the duration of the
frame before that, and in compute_target_delay we may multiply this unusually
big delay.
Signed-off-by: Marton Balint <cus@passwd.hu>
Since target clock is based on the current A-V delay, it is better calculate it
when we actually need it rather than when we put a picture in the picture
queue.
The patch also makes a code a bit more readable by renaming some delay
variables to duration, and converting compute_target_time to a delay
calculating function which does not modify the state. Factoring out the
iteration of the pictq to standalone function is also done in this patch.
Signed-off-by: Marton Balint <cus@passwd.hu>
Well, probably an ABI breakage at least for 32bit, but hopefully it will go
unnoticed because the fix happened realtively soon.
Signed-off-by: Marton Balint <cus@passwd.hu>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Previously ffplay expected SDL_AudioOpen to provide the requested sample rate
and channel number. This is no longer a requirement because this patch replaces
the audio convert function with libswresample's swr_convert which is capable of
handling different sample formats, sample rates and different number of
channels and different channel layouts.
The patch also removes the hardcoded 16bit samples assumption and uses
av_get_bytes_per_sample almost everywhere. The only exceptions are
the update_sample_display and video_audio_display functions, it
seemed too much of a headache to make them generic.
We also fix a tiny bug in sdl_audio_callback, we ensure that the number of
bytes when we put silence in the buffer is a multiple of the frame size.
Also extend the functionality to use the last found program to start the search
after that program.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
If the video queue is aborted, we have to pop the pending ALLOC event or wait
for the allocation to complete, because the current code assumes that
VideoState->pictq_windex does not change until the allocation is complete.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Altough ffplay is working pretty well without using a lock manager, it is still
a multithreaded application calling libavcodec functions from multiple threads,
so using a lock manager is probably a good idea.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Getting rid of globals are generally a good thing. The patch also makes
toggle_pause and step_to_next_frame use a function parameter instead of
the global cur_stream variable.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Since SDL has no audio buffer fullness info, one can get a much precise audio
clock based on the last time of the audio callback and the elapsed time since.
To achieve this I introduced the audio_current_pts and audio_current_pts_drift
variables (similar to video_current_pts and video_current_pts_drift) and
calculate them in the end of the audio callback, when VideoState->audio_clock
is already updated. The reference time I use is from the start of the audio
callback, because this way the amount of time used for audio decoding is not
interfereing with calculation.
I also replaced the audio_write_get_buf_size function with a calculated
variable because when the audio frame decoding is in progress audio_buf_size
and audio_buf_index are not stable, so using them from other threads are not a
good idea.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Previously the target clock of the next frame was calculated by using video_clock
which is modified from another thread...
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
In the main loop, stream_number is incremented after checking the stream type,
so the search usually will not find the wanted stream.
This patch eliminates the useless stream_number variable and introduces a new
one, called real_stream_index to store the real stream index of the current
stream, no matter if we are looping through all the streams or only the streams
of a program.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Signed-off-by: Anton Khirnov <anton@khirnov.net>
Yet another fix for the code originally designed for use without related_stream.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Signed-off-by: Anton Khirnov <anton@khirnov.net>
In the main loop, stream_number is incremented after checking the stream type,
so the search usually will not find the wanted stream.
This patch eliminates the useless stream_number variable and introduces a new
one, called real_stream_index to store the real stream index of the current
stream, no matter if we are looping through all the streams or only the streams
of a program.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Allows playback of nonprimary audio streams in multiple bitrate sources,
such as mmsh://wmscr1.dr.dk/e02ch03m
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
(cherry picked from commit 74d6871d62)
Allows playback of nonprimary audio streams in multiple bitrate sources,
such as mmsh://wmscr1.dr.dk/e02ch03m
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
The current implementation has a bug, it is returning the stream index
in the found program, and not the stream index in the list of all
streams. The attached patch fixes this issue.
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit 22ec6b738f)
The current implementation has a bug, it is returning the stream index
in the found program, and not the stream index in the list of all
streams. The attached patch fixes this issue.
Signed-off-by: Mans Rullgard <mans@mansr.com>
Marton Balint