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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-12-19 14:56:36 +00:00
omxaudioenc: Implement draining of the component and use it
This makes sure that all buffers are encoded and pushed downstream before flushing the ports and losing some buffers.
This commit is contained in:
parent
8eaf25c10e
commit
1ef38751c7
2 changed files with 72 additions and 3 deletions
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@ -47,6 +47,8 @@ static GstFlowReturn gst_omx_audio_enc_handle_frame (GstBaseAudioEncoder *
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encoder, GstBuffer * buffer);
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static void gst_omx_audio_enc_flush (GstBaseAudioEncoder * encoder);
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static GstFlowReturn gst_omx_audio_enc_drain (GstOMXAudioEnc * self);
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enum
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{
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PROP_0
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@ -251,6 +253,8 @@ gst_omx_audio_enc_class_init (GstOMXAudioEncClass * klass)
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static void
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gst_omx_audio_enc_init (GstOMXAudioEnc * self, GstOMXAudioEncClass * klass)
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{
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self->drain_lock = g_mutex_new ();
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self->drain_cond = g_cond_new ();
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}
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static gboolean
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@ -311,7 +315,10 @@ gst_omx_audio_enc_close (GstOMXAudioEnc * self)
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static void
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gst_omx_audio_enc_finalize (GObject * object)
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{
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/* GstOMXAudioEnc *self = GST_OMX_AUDIO_ENC (object); */
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GstOMXAudioEnc *self = GST_OMX_AUDIO_ENC (object);
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g_mutex_free (self->drain_lock);
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g_cond_free (self->drain_cond);
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G_OBJECT_CLASS (parent_class)->finalize (object);
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}
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@ -337,6 +344,11 @@ gst_omx_audio_enc_change_state (GstElement * element, GstStateChange transition)
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if (self->out_port)
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gst_omx_port_set_flushing (self->out_port, FALSE);
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self->downstream_flow_ret = GST_FLOW_OK;
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g_mutex_lock (self->drain_lock);
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self->draining = FALSE;
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g_cond_broadcast (self->drain_cond);
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g_mutex_unlock (self->drain_lock);
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break;
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case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
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break;
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@ -345,6 +357,11 @@ gst_omx_audio_enc_change_state (GstElement * element, GstStateChange transition)
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gst_omx_port_set_flushing (self->in_port, TRUE);
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if (self->out_port)
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gst_omx_port_set_flushing (self->out_port, TRUE);
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g_mutex_lock (self->drain_lock);
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self->draining = FALSE;
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g_cond_broadcast (self->drain_cond);
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g_mutex_unlock (self->drain_lock);
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break;
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default:
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break;
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@ -484,8 +501,16 @@ gst_omx_audio_enc_loop (GstOMXAudioEnc * self)
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outbuf, n_samples);
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}
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if (flow_ret == GST_FLOW_OK && (buf->omx_buf->nFlags & OMX_BUFFERFLAG_EOS))
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flow_ret = GST_FLOW_UNEXPECTED;
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if (flow_ret == GST_FLOW_OK && (buf->omx_buf->nFlags & OMX_BUFFERFLAG_EOS)) {
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g_mutex_lock (self->drain_lock);
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if (self->draining) {
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self->draining = FALSE;
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g_cond_broadcast (self->drain_cond);
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} else {
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flow_ret = GST_FLOW_UNEXPECTED;
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}
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g_mutex_unlock (self->drain_lock);
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}
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gst_omx_port_release_buffer (port, buf);
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@ -584,6 +609,12 @@ gst_omx_audio_enc_stop (GstBaseAudioEncoder * encoder)
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gst_omx_component_set_state (self->component, OMX_StateIdle);
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self->downstream_flow_ret = GST_FLOW_WRONG_STATE;
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g_mutex_lock (self->drain_lock);
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self->draining = FALSE;
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g_cond_broadcast (self->drain_cond);
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g_mutex_unlock (self->drain_lock);
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gst_omx_port_set_flushing (self->in_port, TRUE);
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gst_omx_port_set_flushing (self->out_port, TRUE);
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@ -625,6 +656,7 @@ gst_omx_audio_enc_set_format (GstBaseAudioEncoder * encoder,
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* format change happened we can just exit here.
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*/
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if (needs_disable) {
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gst_omx_audio_enc_drain (self);
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if (gst_omx_port_manual_reconfigure (self->in_port, TRUE) != OMX_ErrorNone)
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return FALSE;
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if (gst_omx_port_set_enabled (self->in_port, FALSE) != OMX_ErrorNone)
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@ -762,6 +794,8 @@ gst_omx_audio_enc_flush (GstBaseAudioEncoder * encoder)
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GST_DEBUG_OBJECT (self, "Resetting encoder");
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if (self->started) {
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gst_omx_audio_enc_drain (self);
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gst_omx_port_set_flushing (self->in_port, TRUE);
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gst_omx_port_set_flushing (self->out_port, TRUE);
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@ -931,3 +965,32 @@ gst_omx_audio_enc_event (GstBaseAudioEncoder * encoder, GstEvent * event)
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return TRUE;
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}
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static GstFlowReturn
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gst_omx_audio_enc_drain (GstOMXAudioEnc * self)
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{
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GstOMXBuffer *buf;
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GstOMXAcquireBufferReturn acq_ret;
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GST_DEBUG_OBJECT (self, "Draining component");
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/* Send an EOS buffer to the component and let the base
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* class drop the EOS event. We will send it later when
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* the EOS buffer arrives on the output port. */
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acq_ret = gst_omx_port_acquire_buffer (self->in_port, &buf);
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if (acq_ret != GST_OMX_ACQUIRE_BUFFER_OK)
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return GST_FLOW_ERROR;
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GST_BASE_AUDIO_ENCODER_STREAM_UNLOCK (self);
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g_mutex_lock (self->drain_lock);
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self->draining = TRUE;
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buf->omx_buf->nFlags |= OMX_BUFFERFLAG_EOS;
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gst_omx_port_release_buffer (self->in_port, buf);
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GST_DEBUG_OBJECT (self, "Waiting until component is drained");
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g_cond_wait (self->drain_cond, self->drain_lock);
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GST_DEBUG_OBJECT (self, "Drained component");
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g_mutex_unlock (self->drain_lock);
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GST_BASE_AUDIO_ENCODER_STREAM_LOCK (self);
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return GST_FLOW_OK;
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}
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@ -58,6 +58,12 @@ struct _GstOMXAudioEnc
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* the first buffer */
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gboolean started;
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/* Draining state */
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GMutex *drain_lock;
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GCond *drain_cond;
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/* TRUE if EOS buffers shouldn't be forwarded */
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gboolean draining;
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GstFlowReturn downstream_flow_ret;
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};
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