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>
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.
This is done in order to clarify the non-video-specific nature of the
buffersink code, as the result of the video/audio API unification of
the previous commit, and for improving overall consistency.
The new API is more generic (no distinction between audio/video for
pulling frames), and avoids code duplication.
A backward compatibility layer is kept for avoiding tools ABI breaks
(only for the video binary interface, audio interface was never used
in the tools).
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>
This way the content of "vfilters" can be reused.
For example when the frame size changes, the filterchain is
reconfigured reusing again the vfilters value.
Use av_log(AV_LOG_LEVEL...) rather than av_dlog, the log is useful
even for "normal" debugging, and consistent with what is done in
ffmpeg.
Also change the message to achieve better consistency with the
corresponding ffmpeg message.
Use the value specified in the codec context for setting the
filterchain sample aspect ratio, when it is not specified in the
stream context.
Consistent with the ffmpeg behavior.
Fix trac issue #398.
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>
All option parsing functions now match the function pointer signature through
which they are called (int f(const char *, const char *), thereby working
reliably on all platforms.
Prefix all option processing functions with opt_
Marton Balint
Michael Niedermayer
Stefano Sabatini
Diego Biurrun
Carl Eugen Hoyos
Anton Khirnov
Mans Rullgard
Jeff Downs
Clément Bœsch
Justin Ruggles
Mina Nagy Zaki