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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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omxaudioenc: Add hack for encoder components that don't allow empty EOS buffers
This commit is contained in:
Jonas Larsson
Sebastian Dröge
parent
2cab5b5db5
commit
887d43c290
1 changed files with 105 additions and 65 deletions
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@ -463,88 +463,107 @@ gst_omx_audio_enc_loop (GstOMXAudioEnc * self)
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return;
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}
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g_assert (acq_return == GST_OMX_ACQUIRE_BUFFER_OK && buf != NULL);
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g_assert (acq_return == GST_OMX_ACQUIRE_BUFFER_OK);
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GST_DEBUG_OBJECT (self, "Handling buffer: 0x%08x %lu", buf->omx_buf->nFlags,
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buf->omx_buf->nTimeStamp);
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if (buf) {
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GST_AUDIO_ENCODER_STREAM_LOCK (self);
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is_eos = ! !(buf->omx_buf->nFlags & OMX_BUFFERFLAG_EOS);
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GST_DEBUG_OBJECT (self, "Handling buffer: 0x%08x %lu", buf->omx_buf->nFlags,
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buf->omx_buf->nTimeStamp);
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if ((buf->omx_buf->nFlags & OMX_BUFFERFLAG_CODECCONFIG)
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&& buf->omx_buf->nFilledLen > 0) {
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GstCaps *caps;
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GstBuffer *codec_data;
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caps = gst_caps_copy (GST_PAD_CAPS (GST_AUDIO_ENCODER_SRC_PAD (self)));
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codec_data = gst_buffer_new_and_alloc (buf->omx_buf->nFilledLen);
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memcpy (GST_BUFFER_DATA (codec_data),
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buf->omx_buf->pBuffer + buf->omx_buf->nOffset,
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buf->omx_buf->nFilledLen);
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gst_caps_set_simple (caps, "codec_data", GST_TYPE_BUFFER, codec_data, NULL);
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if (!gst_pad_set_caps (GST_AUDIO_ENCODER_SRC_PAD (self), caps)) {
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gst_caps_unref (caps);
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if (buf)
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gst_omx_port_release_buffer (self->out_port, buf);
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GST_AUDIO_ENCODER_STREAM_UNLOCK (self);
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goto caps_failed;
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/* This prevents a deadlock between the srcpad stream
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* lock and the videocodec stream lock, if ::reset()
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* is called at the wrong time
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*/
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if (gst_omx_port_is_flushing (self->out_port)) {
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GST_DEBUG_OBJECT (self, "Flushing");
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gst_omx_port_release_buffer (self->out_port, buf);
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goto flushing;
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}
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gst_caps_unref (caps);
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flow_ret = GST_FLOW_OK;
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} else if (buf->omx_buf->nFilledLen > 0) {
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GstBuffer *outbuf;
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guint n_samples;
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n_samples =
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klass->get_num_samples (self, self->out_port,
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gst_audio_encoder_get_audio_info (GST_AUDIO_ENCODER (self)), buf);
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GST_AUDIO_ENCODER_STREAM_LOCK (self);
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is_eos = ! !(buf->omx_buf->nFlags & OMX_BUFFERFLAG_EOS);
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if (buf->omx_buf->nFilledLen > 0) {
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outbuf = gst_buffer_new_and_alloc (buf->omx_buf->nFilledLen);
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if ((buf->omx_buf->nFlags & OMX_BUFFERFLAG_CODECCONFIG)
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&& buf->omx_buf->nFilledLen > 0) {
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GstCaps *caps;
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GstBuffer *codec_data;
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memcpy (GST_BUFFER_DATA (outbuf),
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caps = gst_caps_copy (GST_PAD_CAPS (GST_AUDIO_ENCODER_SRC_PAD (self)));
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codec_data = gst_buffer_new_and_alloc (buf->omx_buf->nFilledLen);
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memcpy (GST_BUFFER_DATA (codec_data),
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buf->omx_buf->pBuffer + buf->omx_buf->nOffset,
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buf->omx_buf->nFilledLen);
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} else {
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outbuf = gst_buffer_new ();
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}
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gst_buffer_set_caps (outbuf,
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GST_PAD_CAPS (GST_AUDIO_ENCODER_SRC_PAD (self)));
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gst_caps_set_simple (caps, "codec_data", GST_TYPE_BUFFER, codec_data,
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NULL);
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if (!gst_pad_set_caps (GST_AUDIO_ENCODER_SRC_PAD (self), caps)) {
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gst_caps_unref (caps);
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if (buf)
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gst_omx_port_release_buffer (self->out_port, buf);
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GST_AUDIO_ENCODER_STREAM_UNLOCK (self);
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goto caps_failed;
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}
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gst_caps_unref (caps);
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flow_ret = GST_FLOW_OK;
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} else if (buf->omx_buf->nFilledLen > 0) {
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GstBuffer *outbuf;
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guint n_samples;
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GST_BUFFER_TIMESTAMP (outbuf) =
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gst_util_uint64_scale (buf->omx_buf->nTimeStamp, GST_SECOND,
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OMX_TICKS_PER_SECOND);
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if (buf->omx_buf->nTickCount != 0)
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GST_BUFFER_DURATION (outbuf) =
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gst_util_uint64_scale (buf->omx_buf->nTickCount, GST_SECOND,
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n_samples =
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klass->get_num_samples (self, self->out_port,
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gst_audio_encoder_get_audio_info (GST_AUDIO_ENCODER (self)), buf);
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if (buf->omx_buf->nFilledLen > 0) {
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outbuf = gst_buffer_new_and_alloc (buf->omx_buf->nFilledLen);
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memcpy (GST_BUFFER_DATA (outbuf),
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buf->omx_buf->pBuffer + buf->omx_buf->nOffset,
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buf->omx_buf->nFilledLen);
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} else {
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outbuf = gst_buffer_new ();
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}
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gst_buffer_set_caps (outbuf,
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GST_PAD_CAPS (GST_AUDIO_ENCODER_SRC_PAD (self)));
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GST_BUFFER_TIMESTAMP (outbuf) =
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gst_util_uint64_scale (buf->omx_buf->nTimeStamp, GST_SECOND,
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OMX_TICKS_PER_SECOND);
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if (buf->omx_buf->nTickCount != 0)
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GST_BUFFER_DURATION (outbuf) =
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gst_util_uint64_scale (buf->omx_buf->nTickCount, GST_SECOND,
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OMX_TICKS_PER_SECOND);
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flow_ret =
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gst_audio_encoder_finish_frame (GST_AUDIO_ENCODER (self),
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outbuf, n_samples);
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}
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if (is_eos || flow_ret == GST_FLOW_UNEXPECTED) {
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g_mutex_lock (self->drain_lock);
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if (self->draining) {
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GST_DEBUG_OBJECT (self, "Drained");
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self->draining = FALSE;
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g_cond_broadcast (self->drain_cond);
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} else if (flow_ret == GST_FLOW_OK) {
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GST_DEBUG_OBJECT (self, "Component signalled EOS");
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flow_ret = GST_FLOW_UNEXPECTED;
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flow_ret =
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gst_audio_encoder_finish_frame (GST_AUDIO_ENCODER (self),
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outbuf, n_samples);
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}
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g_mutex_unlock (self->drain_lock);
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if (is_eos || flow_ret == GST_FLOW_UNEXPECTED) {
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g_mutex_lock (self->drain_lock);
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if (self->draining) {
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GST_DEBUG_OBJECT (self, "Drained");
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self->draining = FALSE;
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g_cond_broadcast (self->drain_cond);
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} else if (flow_ret == GST_FLOW_OK) {
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GST_DEBUG_OBJECT (self, "Component signalled EOS");
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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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} else {
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GST_DEBUG_OBJECT (self, "Finished frame: %s",
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gst_flow_get_name (flow_ret));
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}
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gst_omx_port_release_buffer (port, buf);
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self->downstream_flow_ret = flow_ret;
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} else {
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GST_DEBUG_OBJECT (self, "Finished frame: %s", gst_flow_get_name (flow_ret));
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g_assert ((klass->hacks & GST_OMX_HACK_NO_EMPTY_EOS_BUFFER));
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GST_AUDIO_ENCODER_STREAM_LOCK (self);
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flow_ret = GST_FLOW_UNEXPECTED;
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}
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gst_omx_port_release_buffer (port, buf);
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self->downstream_flow_ret = flow_ret;
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if (flow_ret != GST_FLOW_OK)
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goto flow_error;
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@ -993,8 +1012,10 @@ static gboolean
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gst_omx_audio_enc_event (GstAudioEncoder * encoder, GstEvent * event)
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{
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GstOMXAudioEnc *self;
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GstOMXAudioEncClass *klass;
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self = GST_OMX_AUDIO_ENC (encoder);
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klass = GST_OMX_AUDIO_ENC_GET_CLASS (self);
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if (GST_EVENT_TYPE (event) == GST_EVENT_EOS) {
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GstOMXBuffer *buf;
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@ -1009,6 +1030,17 @@ gst_omx_audio_enc_event (GstAudioEncoder * encoder, GstEvent * event)
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}
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self->eos = TRUE;
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if ((klass->hacks & GST_OMX_HACK_NO_EMPTY_EOS_BUFFER)) {
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GST_WARNING_OBJECT (self, "Component does not support empty EOS buffers");
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/* Insert a NULL into the queue to signal EOS */
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g_mutex_lock (self->out_port->port_lock);
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g_queue_push_tail (self->out_port->pending_buffers, NULL);
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g_cond_broadcast (self->out_port->port_cond);
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g_mutex_unlock (self->out_port->port_lock);
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return TRUE;
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}
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/* Make sure to release the base class stream lock, otherwise
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* _loop() can't call _finish_frame() and we might block forever
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* because no input buffers are released */
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@ -1042,11 +1074,14 @@ gst_omx_audio_enc_event (GstAudioEncoder * encoder, GstEvent * event)
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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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GstOMXAudioEncClass *klass;
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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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klass = GST_OMX_AUDIO_ENC_GET_CLASS (self);
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if (!self->started) {
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GST_DEBUG_OBJECT (self, "Component not started yet");
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return GST_FLOW_OK;
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@ -1059,6 +1094,11 @@ gst_omx_audio_enc_drain (GstOMXAudioEnc * self)
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return GST_FLOW_OK;
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}
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if ((klass->hacks & GST_OMX_HACK_NO_EMPTY_EOS_BUFFER)) {
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GST_WARNING_OBJECT (self, "Component does not support empty EOS buffers");
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return GST_FLOW_OK;
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}
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/* Make sure to release the base class stream lock, otherwise
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* _loop() can't call _finish_frame() and we might block forever
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* because no input buffers are released */
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