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omxaudiodec: Fix flushing logic and make it more similar to the video decoder

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This commit is contained in:
Sebastian Dröge 2014-08-14 17:33:07 +03:00
parent b4c7c726ef
commit 19d77004d2
1 changed files with 42 additions and 15 deletions
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57
omx/gstomxaudiodec.c
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@ -920,11 +920,7 @@ gst_omx_audio_dec_set_format (GstAudioDecoder * decoder, GstCaps * caps)
return FALSE;
}
/* Start the srcpad loop again */
GST_DEBUG_OBJECT (self, "Starting task again");
self->downstream_flow_ret = GST_FLOW_OK;
gst_pad_start_task (GST_AUDIO_DECODER_SRC_PAD (self),
(GstTaskFunction) gst_omx_audio_dec_loop, decoder, NULL);
return TRUE;
}
@ -933,28 +929,53 @@ static void
gst_omx_audio_dec_flush (GstAudioDecoder * decoder, gboolean hard)
{
GstOMXAudioDec *self = GST_OMX_AUDIO_DEC (decoder);
OMX_ERRORTYPE err = OMX_ErrorNone;
GST_DEBUG_OBJECT (self, "Resetting decoder");
GST_DEBUG_OBJECT (self, "Flushing decoder");
if (gst_omx_component_get_state (self->dec, 0) == OMX_StateLoaded)
return;
/* 0) Pause the components */
if (gst_omx_component_get_state (self->dec, 0) == OMX_StateExecuting) {
gst_omx_component_set_state (self->dec, OMX_StatePause);
gst_omx_component_get_state (self->dec, GST_CLOCK_TIME_NONE);
}
/* 1) Wait until the srcpad loop is stopped,
* unlock GST_AUDIO_DECODER_STREAM_LOCK to prevent deadlocks
* caused by using this lock from inside the loop function */
GST_AUDIO_DECODER_STREAM_UNLOCK (self);
gst_pad_stop_task (GST_AUDIO_DECODER_SRC_PAD (decoder));
GST_DEBUG_OBJECT (self, "Flushing -- task stopped");
GST_AUDIO_DECODER_STREAM_LOCK (self);
/* 2) Flush the ports */
GST_DEBUG_OBJECT (self, "flushing ports");
gst_omx_port_set_flushing (self->dec_in_port, 5 * GST_SECOND, TRUE);
gst_omx_port_set_flushing (self->dec_out_port, 5 * GST_SECOND, TRUE);
/* Wait until the srcpad loop is finished */
GST_AUDIO_ENCODER_STREAM_UNLOCK (self);
GST_PAD_STREAM_LOCK (GST_AUDIO_ENCODER_SRC_PAD (self));
GST_PAD_STREAM_UNLOCK (GST_AUDIO_ENCODER_SRC_PAD (self));
GST_AUDIO_ENCODER_STREAM_LOCK (self);
/* 3) Resume components */
gst_omx_component_set_state (self->dec, OMX_StateExecuting);
gst_omx_component_get_state (self->dec, GST_CLOCK_TIME_NONE);
/* 4) Unset flushing to allow ports to accept data again */
gst_omx_port_set_flushing (self->dec_in_port, 5 * GST_SECOND, FALSE);
gst_omx_port_set_flushing (self->dec_out_port, 5 * GST_SECOND, FALSE);
gst_omx_port_populate (self->dec_out_port);
/* Start the srcpad loop again */
err = gst_omx_port_populate (self->dec_out_port);
if (err != OMX_ErrorNone) {
GST_WARNING_OBJECT (self, "Failed to populate output port: %s (0x%08x)",
gst_omx_error_to_string (err), err);
}
/* Reset our state */
self->last_upstream_ts = 0;
self->downstream_flow_ret = GST_FLOW_OK;
self->eos = FALSE;
gst_pad_start_task (GST_AUDIO_DECODER_SRC_PAD (self),
(GstTaskFunction) gst_omx_audio_dec_loop, decoder, NULL);
self->downstream_flow_ret = GST_FLOW_OK;
self->started = FALSE;
GST_DEBUG_OBJECT (self, "Flush finished");
}
static GstFlowReturn
@ -993,6 +1014,12 @@ gst_omx_audio_dec_handle_frame (GstAudioDecoder * decoder, GstBuffer * inbuf)
return self->downstream_flow_ret;
}
if (!self->started) {
GST_DEBUG_OBJECT (self, "Starting task");
gst_pad_start_task (GST_AUDIO_DECODER_SRC_PAD (self),
(GstTaskFunction) gst_omx_audio_dec_loop, decoder, NULL);
}
if (inbuf == NULL)
return gst_omx_audio_dec_drain (self, TRUE);