diff options
| author |  wm4 <wm4@nowhere> | 2013-11-23 21:42:31 +0100 |
|---|---|---|
| committer | wm4 <wm4@nowhere> | 2013-11-23 21:42:31 +0100 |
| commit | e901bb7c99e22e9986a62fc721757cb30ac7ceb9 (patch) | |
| tree | 42d7214df70323cc619991c6b24078fb521592f8 /mpvcore | |
| parent | f90e7ef7eafa0d5b1933f93bb116011cf1cb74b1 (diff) | |
audio: respect --end/--length with spdif passthrough
In theory, we can't really do this, because we don't know when a spdif
frame ends. Spdif transports compressed audio through audio setups that
were originally designed for PCM only (which includes the audio filter
chain, the AO API, most audio output APIs, etc.), and to reach this
goal, spdif pretends to be PCM. Compressed data frames are padded with
zeros, until a certain data rate is reached, which corresponds to a
pseudo-PCM format with 2 bytes per sample and 2 channels at 48000 Hz.
Of course an actual spdif frame is significantly larger than a frame
of the PCM format it pretends to be, so cutting audio data on frame
boundaries (as according to the pseudo-PCM format) merely yields an
incomplete and broken frame, not audio that plays for the desired
duration.
However, sending an incomplete frame might still be much better than the
current behavior, which simply ignores --end/--length (but still lets
the video end at the exact end time).
Should this result in trouble with real spdif receivers, this commit
probably has to be reverted.
Diffstat (limited to 'mpvcore')
| -rw-r--r-- | mpvcore/player/audio.c | 3 |
1 files changed, 2 insertions, 1 deletions
diff --git a/mpvcore/player/audio.c b/mpvcore/player/audio.c index 377b393dfd..042e6d66c3 100644 --- a/mpvcore/player/audio.c +++ b/mpvcore/player/audio.c @@ -367,6 +367,7 @@ int fill_audio_out_buffers(struct MPContext *mpctx, double endpts) bool signal_eof = false; bool partial_fill = false; struct dec_audio *d_audio = mpctx->d_audio; + // Can't adjust the start of audio with spdif pass-through. bool modifiable_audio_format = !(ao->format & AF_FORMAT_SPECIAL_MASK); assert(d_audio); @@ -406,7 +407,7 @@ int fill_audio_out_buffers(struct MPContext *mpctx, double endpts) audio_eof = true; } - if (endpts != MP_NOPTS_VALUE && modifiable_audio_format) { + if (endpts != MP_NOPTS_VALUE) { double samples = (endpts - written_audio_pts(mpctx) + mpctx->audio_delay) * ao->samplerate / opts->playback_speed; if (playsize > samples) { |