mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-10-02 08:42:32 +00:00
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
|
|
@ -463,88 +463,107 @@ gst_omx_audio_enc_loop (GstOMXAudioEnc * self)
|
|||
return;
|
||||
}
|
||||
|
||||
g_assert (acq_return == GST_OMX_ACQUIRE_BUFFER_OK && buf != NULL);
|
||||
g_assert (acq_return == GST_OMX_ACQUIRE_BUFFER_OK);
|
||||
|
||||
GST_DEBUG_OBJECT (self, "Handling buffer: 0x%08x %lu", buf->omx_buf->nFlags,
|
||||
buf->omx_buf->nTimeStamp);
|
||||
if (buf) {
|
||||
|
||||
GST_AUDIO_ENCODER_STREAM_LOCK (self);
|
||||
is_eos = ! !(buf->omx_buf->nFlags & OMX_BUFFERFLAG_EOS);
|
||||
GST_DEBUG_OBJECT (self, "Handling buffer: 0x%08x %lu", buf->omx_buf->nFlags,
|
||||
buf->omx_buf->nTimeStamp);
|
||||
|
||||
if ((buf->omx_buf->nFlags & OMX_BUFFERFLAG_CODECCONFIG)
|
||||
&& buf->omx_buf->nFilledLen > 0) {
|
||||
GstCaps *caps;
|
||||
GstBuffer *codec_data;
|
||||
|
||||
caps = gst_caps_copy (GST_PAD_CAPS (GST_AUDIO_ENCODER_SRC_PAD (self)));
|
||||
codec_data = gst_buffer_new_and_alloc (buf->omx_buf->nFilledLen);
|
||||
memcpy (GST_BUFFER_DATA (codec_data),
|
||||
buf->omx_buf->pBuffer + buf->omx_buf->nOffset,
|
||||
buf->omx_buf->nFilledLen);
|
||||
|
||||
gst_caps_set_simple (caps, "codec_data", GST_TYPE_BUFFER, codec_data, NULL);
|
||||
if (!gst_pad_set_caps (GST_AUDIO_ENCODER_SRC_PAD (self), caps)) {
|
||||
gst_caps_unref (caps);
|
||||
if (buf)
|
||||
gst_omx_port_release_buffer (self->out_port, buf);
|
||||
GST_AUDIO_ENCODER_STREAM_UNLOCK (self);
|
||||
goto caps_failed;
|
||||
/* This prevents a deadlock between the srcpad stream
|
||||
* lock and the videocodec stream lock, if ::reset()
|
||||
* is called at the wrong time
|
||||
*/
|
||||
if (gst_omx_port_is_flushing (self->out_port)) {
|
||||
GST_DEBUG_OBJECT (self, "Flushing");
|
||||
gst_omx_port_release_buffer (self->out_port, buf);
|
||||
goto flushing;
|
||||
}
|
||||
gst_caps_unref (caps);
|
||||
flow_ret = GST_FLOW_OK;
|
||||
} else if (buf->omx_buf->nFilledLen > 0) {
|
||||
GstBuffer *outbuf;
|
||||
guint n_samples;
|
||||
|
||||
n_samples =
|
||||
klass->get_num_samples (self, self->out_port,
|
||||
gst_audio_encoder_get_audio_info (GST_AUDIO_ENCODER (self)), buf);
|
||||
GST_AUDIO_ENCODER_STREAM_LOCK (self);
|
||||
is_eos = ! !(buf->omx_buf->nFlags & OMX_BUFFERFLAG_EOS);
|
||||
|
||||
if (buf->omx_buf->nFilledLen > 0) {
|
||||
outbuf = gst_buffer_new_and_alloc (buf->omx_buf->nFilledLen);
|
||||
if ((buf->omx_buf->nFlags & OMX_BUFFERFLAG_CODECCONFIG)
|
||||
&& buf->omx_buf->nFilledLen > 0) {
|
||||
GstCaps *caps;
|
||||
GstBuffer *codec_data;
|
||||
|
||||
memcpy (GST_BUFFER_DATA (outbuf),
|
||||
caps = gst_caps_copy (GST_PAD_CAPS (GST_AUDIO_ENCODER_SRC_PAD (self)));
|
||||
codec_data = gst_buffer_new_and_alloc (buf->omx_buf->nFilledLen);
|
||||
memcpy (GST_BUFFER_DATA (codec_data),
|
||||
buf->omx_buf->pBuffer + buf->omx_buf->nOffset,
|
||||
buf->omx_buf->nFilledLen);
|
||||
} else {
|
||||
outbuf = gst_buffer_new ();
|
||||
}
|
||||
|
||||
gst_buffer_set_caps (outbuf,
|
||||
GST_PAD_CAPS (GST_AUDIO_ENCODER_SRC_PAD (self)));
|
||||
gst_caps_set_simple (caps, "codec_data", GST_TYPE_BUFFER, codec_data,
|
||||
NULL);
|
||||
if (!gst_pad_set_caps (GST_AUDIO_ENCODER_SRC_PAD (self), caps)) {
|
||||
gst_caps_unref (caps);
|
||||
if (buf)
|
||||
gst_omx_port_release_buffer (self->out_port, buf);
|
||||
GST_AUDIO_ENCODER_STREAM_UNLOCK (self);
|
||||
goto caps_failed;
|
||||
}
|
||||
gst_caps_unref (caps);
|
||||
flow_ret = GST_FLOW_OK;
|
||||
} else if (buf->omx_buf->nFilledLen > 0) {
|
||||
GstBuffer *outbuf;
|
||||
guint n_samples;
|
||||
|
||||
GST_BUFFER_TIMESTAMP (outbuf) =
|
||||
gst_util_uint64_scale (buf->omx_buf->nTimeStamp, GST_SECOND,
|
||||
OMX_TICKS_PER_SECOND);
|
||||
if (buf->omx_buf->nTickCount != 0)
|
||||
GST_BUFFER_DURATION (outbuf) =
|
||||
gst_util_uint64_scale (buf->omx_buf->nTickCount, GST_SECOND,
|
||||
n_samples =
|
||||
klass->get_num_samples (self, self->out_port,
|
||||
gst_audio_encoder_get_audio_info (GST_AUDIO_ENCODER (self)), buf);
|
||||
|
||||
if (buf->omx_buf->nFilledLen > 0) {
|
||||
outbuf = gst_buffer_new_and_alloc (buf->omx_buf->nFilledLen);
|
||||
|
||||
memcpy (GST_BUFFER_DATA (outbuf),
|
||||
buf->omx_buf->pBuffer + buf->omx_buf->nOffset,
|
||||
buf->omx_buf->nFilledLen);
|
||||
} else {
|
||||
outbuf = gst_buffer_new ();
|
||||
}
|
||||
|
||||
gst_buffer_set_caps (outbuf,
|
||||
GST_PAD_CAPS (GST_AUDIO_ENCODER_SRC_PAD (self)));
|
||||
|
||||
GST_BUFFER_TIMESTAMP (outbuf) =
|
||||
gst_util_uint64_scale (buf->omx_buf->nTimeStamp, GST_SECOND,
|
||||
OMX_TICKS_PER_SECOND);
|
||||
if (buf->omx_buf->nTickCount != 0)
|
||||
GST_BUFFER_DURATION (outbuf) =
|
||||
gst_util_uint64_scale (buf->omx_buf->nTickCount, GST_SECOND,
|
||||
OMX_TICKS_PER_SECOND);
|
||||
|
||||
flow_ret =
|
||||
gst_audio_encoder_finish_frame (GST_AUDIO_ENCODER (self),
|
||||
outbuf, n_samples);
|
||||
}
|
||||
|
||||
if (is_eos || flow_ret == GST_FLOW_UNEXPECTED) {
|
||||
g_mutex_lock (self->drain_lock);
|
||||
if (self->draining) {
|
||||
GST_DEBUG_OBJECT (self, "Drained");
|
||||
self->draining = FALSE;
|
||||
g_cond_broadcast (self->drain_cond);
|
||||
} else if (flow_ret == GST_FLOW_OK) {
|
||||
GST_DEBUG_OBJECT (self, "Component signalled EOS");
|
||||
flow_ret = GST_FLOW_UNEXPECTED;
|
||||
flow_ret =
|
||||
gst_audio_encoder_finish_frame (GST_AUDIO_ENCODER (self),
|
||||
outbuf, n_samples);
|
||||
}
|
||||
g_mutex_unlock (self->drain_lock);
|
||||
|
||||
if (is_eos || flow_ret == GST_FLOW_UNEXPECTED) {
|
||||
g_mutex_lock (self->drain_lock);
|
||||
if (self->draining) {
|
||||
GST_DEBUG_OBJECT (self, "Drained");
|
||||
self->draining = FALSE;
|
||||
g_cond_broadcast (self->drain_cond);
|
||||
} else if (flow_ret == GST_FLOW_OK) {
|
||||
GST_DEBUG_OBJECT (self, "Component signalled EOS");
|
||||
flow_ret = GST_FLOW_UNEXPECTED;
|
||||
}
|
||||
g_mutex_unlock (self->drain_lock);
|
||||
} else {
|
||||
GST_DEBUG_OBJECT (self, "Finished frame: %s",
|
||||
gst_flow_get_name (flow_ret));
|
||||
}
|
||||
|
||||
gst_omx_port_release_buffer (port, buf);
|
||||
|
||||
self->downstream_flow_ret = flow_ret;
|
||||
} else {
|
||||
GST_DEBUG_OBJECT (self, "Finished frame: %s", gst_flow_get_name (flow_ret));
|
||||
g_assert ((klass->hacks & GST_OMX_HACK_NO_EMPTY_EOS_BUFFER));
|
||||
GST_AUDIO_ENCODER_STREAM_LOCK (self);
|
||||
flow_ret = GST_FLOW_UNEXPECTED;
|
||||
}
|
||||
|
||||
gst_omx_port_release_buffer (port, buf);
|
||||
|
||||
self->downstream_flow_ret = flow_ret;
|
||||
|
||||
if (flow_ret != GST_FLOW_OK)
|
||||
goto flow_error;
|
||||
|
||||
|
|
@ -993,8 +1012,10 @@ static gboolean
|
|||
gst_omx_audio_enc_event (GstAudioEncoder * encoder, GstEvent * event)
|
||||
{
|
||||
GstOMXAudioEnc *self;
|
||||
GstOMXAudioEncClass *klass;
|
||||
|
||||
self = GST_OMX_AUDIO_ENC (encoder);
|
||||
klass = GST_OMX_AUDIO_ENC_GET_CLASS (self);
|
||||
|
||||
if (GST_EVENT_TYPE (event) == GST_EVENT_EOS) {
|
||||
GstOMXBuffer *buf;
|
||||
|
|
@ -1009,6 +1030,17 @@ gst_omx_audio_enc_event (GstAudioEncoder * encoder, GstEvent * event)
|
|||
}
|
||||
self->eos = TRUE;
|
||||
|
||||
if ((klass->hacks & GST_OMX_HACK_NO_EMPTY_EOS_BUFFER)) {
|
||||
GST_WARNING_OBJECT (self, "Component does not support empty EOS buffers");
|
||||
|
||||
/* Insert a NULL into the queue to signal EOS */
|
||||
g_mutex_lock (self->out_port->port_lock);
|
||||
g_queue_push_tail (self->out_port->pending_buffers, NULL);
|
||||
g_cond_broadcast (self->out_port->port_cond);
|
||||
g_mutex_unlock (self->out_port->port_lock);
|
||||
return TRUE;
|
||||
}
|
||||
|
||||
/* Make sure to release the base class stream lock, otherwise
|
||||
* _loop() can't call _finish_frame() and we might block forever
|
||||
* because no input buffers are released */
|
||||
|
|
@ -1042,11 +1074,14 @@ gst_omx_audio_enc_event (GstAudioEncoder * encoder, GstEvent * event)
|
|||
static GstFlowReturn
|
||||
gst_omx_audio_enc_drain (GstOMXAudioEnc * self)
|
||||
{
|
||||
GstOMXAudioEncClass *klass;
|
||||
GstOMXBuffer *buf;
|
||||
GstOMXAcquireBufferReturn acq_ret;
|
||||
|
||||
GST_DEBUG_OBJECT (self, "Draining component");
|
||||
|
||||
klass = GST_OMX_AUDIO_ENC_GET_CLASS (self);
|
||||
|
||||
if (!self->started) {
|
||||
GST_DEBUG_OBJECT (self, "Component not started yet");
|
||||
return GST_FLOW_OK;
|
||||
|
|
@ -1059,6 +1094,11 @@ gst_omx_audio_enc_drain (GstOMXAudioEnc * self)
|
|||
return GST_FLOW_OK;
|
||||
}
|
||||
|
||||
if ((klass->hacks & GST_OMX_HACK_NO_EMPTY_EOS_BUFFER)) {
|
||||
GST_WARNING_OBJECT (self, "Component does not support empty EOS buffers");
|
||||
return GST_FLOW_OK;
|
||||
}
|
||||
|
||||
/* Make sure to release the base class stream lock, otherwise
|
||||
* _loop() can't call _finish_frame() and we might block forever
|
||||
* because no input buffers are released */
|
||||
|
|
|
|||
Loading…
Reference in a new issue