diff options
| -rw-r--r-- | gst-libs/gst/audio/gstaudiobasesink.c | 15 |
1 files changed, 8 insertions, 7 deletions
diff --git a/gst-libs/gst/audio/gstaudiobasesink.c b/gst-libs/gst/audio/gstaudiobasesink.c index 704801e45..45050224f 100644 --- a/gst-libs/gst/audio/gstaudiobasesink.c +++ b/gst-libs/gst/audio/gstaudiobasesink.c @@ -1086,13 +1086,14 @@ gst_audio_base_sink_force_start (GstAudioBaseSink * sink) } /* This waits for the drain to happen and can be canceled */ -static gboolean +static GstFlowReturn gst_audio_base_sink_drain (GstAudioBaseSink * sink) { + GstFlowReturn ret = GST_FLOW_OK; if (!sink->ringbuffer) - return TRUE; + return ret; if (!sink->ringbuffer->spec.info.rate) - return TRUE; + return ret; /* if PLAYING is interrupted, * arrange to have clock running when going to PLAYING again */ @@ -1111,19 +1112,19 @@ gst_audio_base_sink_drain (GstAudioBaseSink * sink) /* wait for the EOS time to be reached, this is the time when the last * sample is played. */ - gst_base_sink_wait (GST_BASE_SINK (sink), sink->priv->eos_time, NULL); + ret = gst_base_sink_wait (GST_BASE_SINK (sink), sink->priv->eos_time, NULL); GST_DEBUG_OBJECT (sink, "drained audio"); } g_atomic_int_set (&sink->eos_rendering, 0); - return TRUE; + return ret; } static GstFlowReturn gst_audio_base_sink_wait_event (GstBaseSink * bsink, GstEvent * event) { GstAudioBaseSink *sink = GST_AUDIO_BASE_SINK (bsink); - GstFlowReturn ret; + GstFlowReturn ret = GST_FLOW_OK; gboolean clear_force_start_flag = FALSE; /* For both gap and EOS events, make sure the ringbuffer is running @@ -1155,7 +1156,7 @@ gst_audio_base_sink_wait_event (GstBaseSink * bsink, GstEvent * event) switch (GST_EVENT_TYPE (event)) { case GST_EVENT_EOS: /* now wait till we played everything */ - gst_audio_base_sink_drain (sink); + ret = gst_audio_base_sink_drain (sink); break; default: break; |