void
gst_decklink_audio_src_get_property (GObject * object, guint property_id,
GValue * value, GParamSpec * pspec)
{
GstDecklinkAudioSrc *self = GST_DECKLINK_AUDIO_SRC_CAST (object);
switch (property_id) {
case PROP_CONNECTION:
g_value_set_enum (value, self->connection);
break;
case PROP_DEVICE_NUMBER:
g_value_set_int (value, self->device_number);
break;
case PROP_ALIGNMENT_THRESHOLD:
g_value_set_uint64 (value, self->alignment_threshold);
break;
case PROP_DISCONT_WAIT:
g_value_set_uint64 (value, self->discont_wait);
break;
case PROP_BUFFER_SIZE:
g_value_set_uint (value, self->buffer_size);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, property_id, pspec);
break;
}
}
static GstCaps *
gst_decklink_audio_src_get_caps (GstBaseSrc * bsrc, GstCaps * filter)
{
GstDecklinkAudioSrc *self = GST_DECKLINK_AUDIO_SRC_CAST (bsrc);
GstCaps *caps;
// We don't support renegotiation
caps = gst_pad_get_current_caps (GST_BASE_SRC_PAD (bsrc));
if (!caps) {
GstCaps *channel_filter, *templ;
templ = gst_pad_get_pad_template_caps (GST_BASE_SRC_PAD (bsrc));
channel_filter =
gst_caps_new_simple ("audio/x-raw", "channels", G_TYPE_INT,
self->channels, NULL);
caps = gst_caps_intersect (channel_filter, templ);
gst_caps_unref (channel_filter);
gst_caps_unref (templ);
}
if (filter) {
GstCaps *tmp =
gst_caps_intersect_full (filter, caps, GST_CAPS_INTERSECT_FIRST);
gst_caps_unref (caps);
caps = tmp;
}
return caps;
}
void
gst_decklink_audio_src_finalize (GObject * object)
{
GstDecklinkAudioSrc *self = GST_DECKLINK_AUDIO_SRC_CAST (object);
g_mutex_clear (&self->lock);
g_cond_clear (&self->cond);
G_OBJECT_CLASS (parent_class)->finalize (object);
}
static gboolean
gst_decklink_audio_src_unlock (GstBaseSrc * bsrc)
{
GstDecklinkAudioSrc *self = GST_DECKLINK_AUDIO_SRC_CAST (bsrc);
g_mutex_lock (&self->lock);
self->flushing = TRUE;
g_cond_signal (&self->cond);
g_mutex_unlock (&self->lock);
return TRUE;
}
static gboolean
gst_decklink_audio_src_unlock_stop (GstBaseSrc * bsrc)
{
GstDecklinkAudioSrc *self = GST_DECKLINK_AUDIO_SRC_CAST (bsrc);
g_mutex_lock (&self->lock);
self->flushing = FALSE;
g_queue_foreach (&self->current_packets, (GFunc) capture_packet_free, NULL);
g_queue_clear (&self->current_packets);
g_mutex_unlock (&self->lock);
return TRUE;
}
static void
gst_decklink_audio_src_got_packet (GstElement * element,
IDeckLinkAudioInputPacket * packet, GstClockTime capture_time,
gboolean discont)
{
GstDecklinkAudioSrc *self = GST_DECKLINK_AUDIO_SRC_CAST (element);
GstDecklinkVideoSrc *videosrc = NULL;
GST_LOG_OBJECT (self, "Got audio packet at %" GST_TIME_FORMAT,
GST_TIME_ARGS (capture_time));
g_mutex_lock (&self->input->lock);
if (self->input->videosrc)
videosrc =
GST_DECKLINK_VIDEO_SRC_CAST (gst_object_ref (self->input->videosrc));
g_mutex_unlock (&self->input->lock);
if (videosrc) {
gst_decklink_video_src_convert_to_external_clock (videosrc, &capture_time,
NULL);
gst_object_unref (videosrc);
GST_LOG_OBJECT (self, "Actual timestamp %" GST_TIME_FORMAT,
GST_TIME_ARGS (capture_time));
}
g_mutex_lock (&self->lock);
if (!self->flushing) {
CapturePacket *p;
while (g_queue_get_length (&self->current_packets) >= self->buffer_size) {
p = (CapturePacket *) g_queue_pop_head (&self->current_packets);
GST_WARNING_OBJECT (self, "Dropping old packet at %" GST_TIME_FORMAT,
GST_TIME_ARGS (p->capture_time));
capture_packet_free (p);
}
p = (CapturePacket *) g_malloc0 (sizeof (CapturePacket));
p->packet = packet;
p->capture_time = capture_time;
p->discont = discont;
packet->AddRef ();
g_queue_push_tail (&self->current_packets, p);
g_cond_signal (&self->cond);
}
g_mutex_unlock (&self->lock);
}
static gboolean
gst_decklink_audio_src_query (GstBaseSrc * bsrc, GstQuery * query)
{
GstDecklinkAudioSrc *self = GST_DECKLINK_AUDIO_SRC_CAST (bsrc);
gboolean ret = TRUE;
switch (GST_QUERY_TYPE (query)) {
case GST_QUERY_LATENCY:{
if (self->input) {
g_mutex_lock (&self->input->lock);
if (self->input->mode) {
GstClockTime min, max;
min =
gst_util_uint64_scale_ceil (GST_SECOND, self->input->mode->fps_d,
self->input->mode->fps_n);
max = self->buffer_size * min;
gst_query_set_latency (query, TRUE, min, max);
ret = TRUE;
} else {
ret = FALSE;
}
g_mutex_unlock (&self->input->lock);
} else {
ret = FALSE;
}
break;
}
default:
ret = GST_BASE_SRC_CLASS (parent_class)->query (bsrc, query);
break;
}
return ret;
}
static GstStateChangeReturn
gst_decklink_audio_src_change_state (GstElement * element,
GstStateChange transition)
{
GstDecklinkAudioSrc *self = GST_DECKLINK_AUDIO_SRC_CAST (element);
GstStateChangeReturn ret;
switch (transition) {
case GST_STATE_CHANGE_NULL_TO_READY:
if (!gst_decklink_audio_src_open (self)) {
ret = GST_STATE_CHANGE_FAILURE;
goto out;
}
break;
case GST_STATE_CHANGE_READY_TO_PAUSED:{
GstElement *videosrc = NULL;
// Check if there is a video src for this input too and if it
// is actually in the same pipeline
g_mutex_lock (&self->input->lock);
if (self->input->videosrc)
videosrc = GST_ELEMENT_CAST (gst_object_ref (self->input->videosrc));
g_mutex_unlock (&self->input->lock);
if (!videosrc) {
GST_ELEMENT_ERROR (self, STREAM, FAILED,
(NULL), ("Audio src needs a video src for its operation"));
ret = GST_STATE_CHANGE_FAILURE;
goto out;
}
// FIXME: This causes deadlocks sometimes
#if 0
else if (!in_same_pipeline (GST_ELEMENT_CAST (self), videosrc)) {
GST_ELEMENT_ERROR (self, STREAM, FAILED,
(NULL),
("Audio src and video src need to be in the same pipeline"));
ret = GST_STATE_CHANGE_FAILURE;
gst_object_unref (videosrc);
goto out;
}
#endif
if (videosrc)
gst_object_unref (videosrc);
self->flushing = FALSE;
self->next_offset = -1;
break;
}
default:
break;
}
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
if (ret == GST_STATE_CHANGE_FAILURE)
return ret;
switch (transition) {
case GST_STATE_CHANGE_PAUSED_TO_READY:
gst_decklink_audio_src_stop (self);
break;
case GST_STATE_CHANGE_READY_TO_NULL:
gst_decklink_audio_src_close (self);
break;
default:
break;
}
out:
return ret;
}
static GstFlowReturn
gst_decklink_audio_src_create (GstPushSrc * bsrc, GstBuffer ** buffer)
{
GstDecklinkAudioSrc *self = GST_DECKLINK_AUDIO_SRC_CAST (bsrc);
GstFlowReturn flow_ret = GST_FLOW_OK;
const guint8 *data;
glong sample_count;
gsize data_size;
CapturePacket *p;
AudioPacket *ap;
GstClockTime timestamp, duration;
GstClockTime start_time, end_time;
guint64 start_offset, end_offset;
gboolean discont = FALSE;
g_mutex_lock (&self->lock);
while (g_queue_is_empty (&self->current_packets) && !self->flushing) {
g_cond_wait (&self->cond, &self->lock);
}
p = (CapturePacket *) g_queue_pop_head (&self->current_packets);
g_mutex_unlock (&self->lock);
if (self->flushing) {
if (p)
capture_packet_free (p);
GST_DEBUG_OBJECT (self, "Flushing");
return GST_FLOW_FLUSHING;
}
p->packet->GetBytes ((gpointer *) & data);
sample_count = p->packet->GetSampleFrameCount ();
data_size = self->info.bpf * sample_count;
ap = (AudioPacket *) g_malloc0 (sizeof (AudioPacket));
*buffer =
gst_buffer_new_wrapped_full ((GstMemoryFlags) GST_MEMORY_FLAG_READONLY,
(gpointer) data, data_size, 0, data_size, ap,
(GDestroyNotify) audio_packet_free);
ap->packet = p->packet;
p->packet->AddRef ();
ap->input = self->input->input;
ap->input->AddRef ();
timestamp = p->capture_time;
// Jitter and discontinuity handling, based on audiobasesrc
start_time = timestamp;
// Convert to the sample numbers
start_offset =
gst_util_uint64_scale (start_time, self->info.rate, GST_SECOND);
end_offset = start_offset + sample_count;
end_time = gst_util_uint64_scale_int (end_offset, GST_SECOND,
self->info.rate);
duration = end_time - start_time;
if (self->next_offset == (guint64) - 1) {
discont = TRUE;
} else {
guint64 diff, max_sample_diff;
// Check discont
if (start_offset <= self->next_offset)
diff = self->next_offset - start_offset;
else
diff = start_offset - self->next_offset;
max_sample_diff =
gst_util_uint64_scale_int (self->alignment_threshold, self->info.rate,
GST_SECOND);
// Discont!
if (G_UNLIKELY (diff >= max_sample_diff)) {
if (self->discont_wait > 0) {
if (self->discont_time == GST_CLOCK_TIME_NONE) {
self->discont_time = start_time;
} else if (start_time - self->discont_time >= self->discont_wait) {
discont = TRUE;
self->discont_time = GST_CLOCK_TIME_NONE;
}
} else {
discont = TRUE;
}
} else if (G_UNLIKELY (self->discont_time != GST_CLOCK_TIME_NONE)) {
// we have had a discont, but are now back on track!
self->discont_time = GST_CLOCK_TIME_NONE;
}
}
if (discont) {
// Have discont, need resync and use the capture timestamps
if (self->next_offset != (guint64) - 1)
GST_INFO_OBJECT (self, "Have discont. Expected %"
G_GUINT64_FORMAT ", got %" G_GUINT64_FORMAT,
self->next_offset, start_offset);
GST_BUFFER_FLAG_SET (*buffer, GST_BUFFER_FLAG_DISCONT);
self->next_offset = end_offset;
} else {
// No discont, just keep counting
self->discont_time = GST_CLOCK_TIME_NONE;
timestamp =
gst_util_uint64_scale (self->next_offset, GST_SECOND, self->info.rate);
self->next_offset += sample_count;
duration =
gst_util_uint64_scale (self->next_offset, GST_SECOND,
self->info.rate) - timestamp;
}
GST_BUFFER_TIMESTAMP (*buffer) = timestamp;
GST_BUFFER_DURATION (*buffer) = duration;
GST_DEBUG_OBJECT (self,
"Outputting buffer %p with timestamp %" GST_TIME_FORMAT " and duration %"
GST_TIME_FORMAT, *buffer, GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (*buffer)),
GST_TIME_ARGS (GST_BUFFER_DURATION (*buffer)));
capture_packet_free (p);
return flow_ret;
}
static gboolean
gst_decklink_audio_src_set_caps (GstBaseSrc * bsrc, GstCaps * caps)
{
GstDecklinkAudioSrc *self = GST_DECKLINK_AUDIO_SRC_CAST (bsrc);
BMDAudioSampleType sample_depth;
GstCaps *current_caps;
HRESULT ret;
BMDAudioConnection conn = (BMDAudioConnection) - 1;
GST_DEBUG_OBJECT (self, "Setting caps %" GST_PTR_FORMAT, caps);
if ((current_caps = gst_pad_get_current_caps (GST_BASE_SRC_PAD (bsrc)))) {
GST_DEBUG_OBJECT (self, "Pad already has caps %" GST_PTR_FORMAT, caps);
if (!gst_caps_is_equal (caps, current_caps)) {
GST_ERROR_OBJECT (self, "New caps are not equal to old caps");
gst_caps_unref (current_caps);
return FALSE;
} else {
gst_caps_unref (current_caps);
return TRUE;
}
}
if (!gst_audio_info_from_caps (&self->info, caps))
return FALSE;
if (self->info.finfo->format == GST_AUDIO_FORMAT_S16LE) {
sample_depth = bmdAudioSampleType16bitInteger;
} else {
sample_depth = bmdAudioSampleType32bitInteger;
}
switch (self->connection) {
case GST_DECKLINK_AUDIO_CONNECTION_AUTO:{
GstElement *videosrc = NULL;
GstDecklinkConnectionEnum vconn;
// Try to get the connection from the videosrc and try
// to select a sensible audio connection based on that
g_mutex_lock (&self->input->lock);
if (self->input->videosrc)
videosrc = GST_ELEMENT_CAST (gst_object_ref (self->input->videosrc));
g_mutex_unlock (&self->input->lock);
if (videosrc) {
g_object_get (videosrc, "connection", &vconn, NULL);
gst_object_unref (videosrc);
switch (vconn) {
case GST_DECKLINK_CONNECTION_SDI:
conn = bmdAudioConnectionEmbedded;
break;
case GST_DECKLINK_CONNECTION_HDMI:
conn = bmdAudioConnectionEmbedded;
break;
case GST_DECKLINK_CONNECTION_OPTICAL_SDI:
conn = bmdAudioConnectionEmbedded;
break;
case GST_DECKLINK_CONNECTION_COMPONENT:
conn = bmdAudioConnectionAnalog;
break;
case GST_DECKLINK_CONNECTION_COMPOSITE:
conn = bmdAudioConnectionAnalog;
break;
case GST_DECKLINK_CONNECTION_SVIDEO:
conn = bmdAudioConnectionAnalog;
break;
default:
// Use default
break;
}
}
break;
}
case GST_DECKLINK_AUDIO_CONNECTION_EMBEDDED:
conn = bmdAudioConnectionEmbedded;
break;
case GST_DECKLINK_AUDIO_CONNECTION_AES_EBU:
conn = bmdAudioConnectionAESEBU;
break;
case GST_DECKLINK_AUDIO_CONNECTION_ANALOG:
conn = bmdAudioConnectionAnalog;
break;
case GST_DECKLINK_AUDIO_CONNECTION_ANALOG_XLR:
conn = bmdAudioConnectionAnalogXLR;
break;
case GST_DECKLINK_AUDIO_CONNECTION_ANALOG_RCA:
conn = bmdAudioConnectionAnalogRCA;
break;
default:
g_assert_not_reached ();
break;
}
if (conn != (BMDAudioConnection) - 1) {
ret =
self->input->config->SetInt (bmdDeckLinkConfigAudioInputConnection,
conn);
if (ret != S_OK) {
GST_ERROR ("set configuration (audio input connection)");
return FALSE;
}
}
ret = self->input->input->EnableAudioInput (bmdAudioSampleRate48kHz,
sample_depth, 2);
if (ret != S_OK) {
GST_WARNING_OBJECT (self, "Failed to enable audio input");
return FALSE;
}
g_mutex_lock (&self->input->lock);
self->input->audio_enabled = TRUE;
if (self->input->start_streams && self->input->videosrc)
self->input->start_streams (self->input->videosrc);
g_mutex_unlock (&self->input->lock);
return TRUE;
}
static void
gst_decklink_audio_src_got_packet (GstElement * element,
IDeckLinkAudioInputPacket * packet, GstClockTime capture_time,
GstClockTime packet_time, gboolean no_signal)
{
GstDecklinkAudioSrc *self = GST_DECKLINK_AUDIO_SRC_CAST (element);
GstClockTime timestamp;
GST_LOG_OBJECT (self,
"Got audio packet at %" GST_TIME_FORMAT " / %" GST_TIME_FORMAT
", no signal %d", GST_TIME_ARGS (capture_time),
GST_TIME_ARGS (packet_time), no_signal);
g_mutex_lock (&self->input->lock);
if (self->input->videosrc) {
GstDecklinkVideoSrc *videosrc =
GST_DECKLINK_VIDEO_SRC_CAST (gst_object_ref (self->input->videosrc));
if (videosrc->drop_no_signal_frames && no_signal) {
g_mutex_unlock (&self->input->lock);
return;
}
if (videosrc->first_time == GST_CLOCK_TIME_NONE)
videosrc->first_time = packet_time;
if (videosrc->skip_first_time > 0
&& packet_time - videosrc->first_time < videosrc->skip_first_time) {
GST_DEBUG_OBJECT (self,
"Skipping frame as requested: %" GST_TIME_FORMAT " < %"
GST_TIME_FORMAT, GST_TIME_ARGS (packet_time),
GST_TIME_ARGS (videosrc->skip_first_time + videosrc->first_time));
g_mutex_unlock (&self->input->lock);
return;
}
if (videosrc->output_stream_time)
timestamp = packet_time;
else
timestamp = gst_clock_adjust_with_calibration (NULL, packet_time,
videosrc->current_time_mapping.xbase,
videosrc->current_time_mapping.b, videosrc->current_time_mapping.num,
videosrc->current_time_mapping.den);
} else {
timestamp = capture_time;
}
g_mutex_unlock (&self->input->lock);
GST_LOG_OBJECT (self, "Converted times to %" GST_TIME_FORMAT,
GST_TIME_ARGS (timestamp));
g_mutex_lock (&self->lock);
if (!self->flushing) {
CapturePacket *p;
while (g_queue_get_length (&self->current_packets) >= self->buffer_size) {
p = (CapturePacket *) g_queue_pop_head (&self->current_packets);
GST_WARNING_OBJECT (self, "Dropping old packet at %" GST_TIME_FORMAT,
GST_TIME_ARGS (p->timestamp));
capture_packet_free (p);
}
p = (CapturePacket *) g_malloc0 (sizeof (CapturePacket));
p->packet = packet;
p->timestamp = timestamp;
p->no_signal = no_signal;
packet->AddRef ();
g_queue_push_tail (&self->current_packets, p);
g_cond_signal (&self->cond);
}
g_mutex_unlock (&self->lock);
}
