From 8242676dc20b0a9b5dcac9a0e8048fc672d3b314 Mon Sep 17 00:00:00 2001 From: Thiago Santos Date: Tue, 23 Sep 2014 14:14:36 -0300 Subject: [PATCH] audiosink: compensate for segment restart with clock's time_offset When playing chained data the audio ringbuffer is released and then acquired again. This makes it reset the segbase/segdone variables, but the next sample will be scheduled to play in the next position (right after the sample from the previous media) and, as the segdone is at 0, the audiosink will wait the duration of this previous media before it can write and play the new data. What happens is this: pointer at 0, write to 698-1564, diff 698, segtotal 20, segsize 1764, base 0 it will have to wait the length of 698 samples before being able to write. In a regular sample playback it looks like: pointer at 677, write to 696-1052, diff 19, segtotal 20, segsize 1764, base 0 In this case it will write to the next available position and it doesn't need to wait or fill with silence. This solution is borrowed from pulsesink that resets the clock to start again from 0, which makes it reset the time_offset to the time of the last played sample. This is used to correct the place of writing in the ringbuffer to the new start (0 again) https://bugzilla.gnome.org/show_bug.cgi?id=737055 --- gst-libs/gst/audio/gstaudiosink.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/gst-libs/gst/audio/gstaudiosink.c b/gst-libs/gst/audio/gstaudiosink.c index a978030805..d329512b90 100644 --- a/gst-libs/gst/audio/gstaudiosink.c +++ b/gst-libs/gst/audio/gstaudiosink.c @@ -399,6 +399,7 @@ gst_audio_sink_ring_buffer_acquire (GstAudioRingBuffer * buf, GstAudioSink *sink; GstAudioSinkClass *csink; gboolean result = FALSE; + GstAudioClock *clock; sink = GST_AUDIO_SINK (GST_OBJECT_PARENT (buf)); csink = GST_AUDIO_SINK_GET_CLASS (sink); @@ -408,6 +409,10 @@ gst_audio_sink_ring_buffer_acquire (GstAudioRingBuffer * buf, if (!result) goto could_not_prepare; + /* our clock will now start from 0 again */ + clock = GST_AUDIO_CLOCK (GST_AUDIO_BASE_SINK (sink)->provided_clock); + gst_audio_clock_reset (clock, 0); + /* set latency to one more segment as we need some headroom */ spec->seglatency = spec->segtotal + 1;