gstreamer/sys/decklink/gstdecklinkaudiosink.cpp

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/* GStreamer
* Copyright (C) 2011 David Schleef <ds@entropywave.com>
* Copyright (C) 2014 Sebastian Dröge <sebastian@centricular.com>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 51 Franklin Street, Suite 500,
* Boston, MA 02110-1335, USA.
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include "gstdecklinkaudiosink.h"
GST_DEBUG_CATEGORY_STATIC (gst_decklink_audio_sink_debug);
#define GST_CAT_DEFAULT gst_decklink_audio_sink_debug
// Ringbuffer implementation
#define GST_TYPE_DECKLINK_AUDIO_SINK_RING_BUFFER \
(gst_decklink_audio_sink_ringbuffer_get_type())
#define GST_DECKLINK_AUDIO_SINK_RING_BUFFER(obj) \
(G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_DECKLINK_AUDIO_SINK_RING_BUFFER,GstDecklinkAudioSinkRingBuffer))
#define GST_DECKLINK_AUDIO_SINK_RING_BUFFER_CAST(obj) \
((GstDecklinkAudioSinkRingBuffer*) obj)
#define GST_DECKLINK_AUDIO_SINK_RING_BUFFER_CLASS(klass) \
(G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_DECKLINK_AUDIO_SINK_RING_BUFFER,GstDecklinkAudioSinkRingBufferClass))
#define GST_DECKLINK_AUDIO_SINK_RING_BUFFER_CAST_GET_CLASS(obj) \
(G_TYPE_INSTANCE_GET_CLASS((obj),GST_TYPE_DECKLINK_AUDIO_SINK_RING_BUFFER,GstDecklinkAudioSinkRingBufferClass))
#define GST_IS_DECKLINK_AUDIO_SINK_RING_BUFFER(obj) \
(G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_DECKLINK_AUDIO_SINK_RING_BUFFER))
#define GST_IS_DECKLINK_AUDIO_SINK_RING_BUFFER_CLASS(klass) \
(G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_DECKLINK_AUDIO_SINK_RING_BUFFER))
typedef struct _GstDecklinkAudioSinkRingBuffer GstDecklinkAudioSinkRingBuffer;
typedef struct _GstDecklinkAudioSinkRingBufferClass
GstDecklinkAudioSinkRingBufferClass;
struct _GstDecklinkAudioSinkRingBuffer
{
GstAudioRingBuffer object;
GstDecklinkOutput *output;
GstDecklinkAudioSink *sink;
GMutex clock_id_lock;
GstClockID clock_id;
};
struct _GstDecklinkAudioSinkRingBufferClass
{
GstAudioRingBufferClass parent_class;
};
GType gst_decklink_audio_sink_ringbuffer_get_type (void);
static void gst_decklink_audio_sink_ringbuffer_finalize (GObject * object);
static void gst_decklink_audio_sink_ringbuffer_clear_all (GstAudioRingBuffer *
rb);
static guint gst_decklink_audio_sink_ringbuffer_delay (GstAudioRingBuffer * rb);
static gboolean gst_decklink_audio_sink_ringbuffer_start (GstAudioRingBuffer *
rb);
static gboolean gst_decklink_audio_sink_ringbuffer_pause (GstAudioRingBuffer *
rb);
static gboolean gst_decklink_audio_sink_ringbuffer_stop (GstAudioRingBuffer *
rb);
static gboolean gst_decklink_audio_sink_ringbuffer_acquire (GstAudioRingBuffer *
rb, GstAudioRingBufferSpec * spec);
static gboolean gst_decklink_audio_sink_ringbuffer_release (GstAudioRingBuffer *
rb);
static gboolean
gst_decklink_audio_sink_ringbuffer_open_device (GstAudioRingBuffer * rb);
static gboolean
gst_decklink_audio_sink_ringbuffer_close_device (GstAudioRingBuffer * rb);
#define ringbuffer_parent_class gst_decklink_audio_sink_ringbuffer_parent_class
G_DEFINE_TYPE (GstDecklinkAudioSinkRingBuffer,
gst_decklink_audio_sink_ringbuffer, GST_TYPE_AUDIO_RING_BUFFER);
static void
gst_decklink_audio_sink_ringbuffer_class_init
(GstDecklinkAudioSinkRingBufferClass * klass)
{
GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
GstAudioRingBufferClass *gstringbuffer_class =
GST_AUDIO_RING_BUFFER_CLASS (klass);
gobject_class->finalize = gst_decklink_audio_sink_ringbuffer_finalize;
gstringbuffer_class->open_device =
GST_DEBUG_FUNCPTR (gst_decklink_audio_sink_ringbuffer_open_device);
gstringbuffer_class->close_device =
GST_DEBUG_FUNCPTR (gst_decklink_audio_sink_ringbuffer_close_device);
gstringbuffer_class->acquire =
GST_DEBUG_FUNCPTR (gst_decklink_audio_sink_ringbuffer_acquire);
gstringbuffer_class->release =
GST_DEBUG_FUNCPTR (gst_decklink_audio_sink_ringbuffer_release);
gstringbuffer_class->start =
GST_DEBUG_FUNCPTR (gst_decklink_audio_sink_ringbuffer_start);
gstringbuffer_class->pause =
GST_DEBUG_FUNCPTR (gst_decklink_audio_sink_ringbuffer_pause);
gstringbuffer_class->resume =
GST_DEBUG_FUNCPTR (gst_decklink_audio_sink_ringbuffer_start);
gstringbuffer_class->stop =
GST_DEBUG_FUNCPTR (gst_decklink_audio_sink_ringbuffer_stop);
gstringbuffer_class->delay =
GST_DEBUG_FUNCPTR (gst_decklink_audio_sink_ringbuffer_delay);
gstringbuffer_class->clear_all =
GST_DEBUG_FUNCPTR (gst_decklink_audio_sink_ringbuffer_clear_all);
}
static void
gst_decklink_audio_sink_ringbuffer_init (GstDecklinkAudioSinkRingBuffer * self)
{
g_mutex_init (&self->clock_id_lock);
}
static void
gst_decklink_audio_sink_ringbuffer_finalize (GObject * object)
{
GstDecklinkAudioSinkRingBuffer *self =
GST_DECKLINK_AUDIO_SINK_RING_BUFFER_CAST (object);
gst_object_unref (self->sink);
self->sink = NULL;
g_mutex_clear (&self->clock_id_lock);
G_OBJECT_CLASS (ringbuffer_parent_class)->finalize (object);
}
class GStreamerAudioOutputCallback:public IDeckLinkAudioOutputCallback
{
public:
GStreamerAudioOutputCallback (GstDecklinkAudioSinkRingBuffer * ringbuffer)
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:IDeckLinkAudioOutputCallback (), m_refcount (1)
{
m_ringbuffer =
GST_DECKLINK_AUDIO_SINK_RING_BUFFER_CAST (gst_object_ref (ringbuffer));
g_mutex_init (&m_mutex);
}
virtual HRESULT QueryInterface (REFIID, LPVOID *)
{
return E_NOINTERFACE;
}
virtual ULONG AddRef (void)
{
ULONG ret;
g_mutex_lock (&m_mutex);
m_refcount++;
ret = m_refcount;
g_mutex_unlock (&m_mutex);
return ret;
}
virtual ULONG Release (void)
{
ULONG ret;
g_mutex_lock (&m_mutex);
m_refcount--;
ret = m_refcount;
g_mutex_unlock (&m_mutex);
if (ret == 0) {
delete this;
}
return ret;
}
virtual ~ GStreamerAudioOutputCallback () {
gst_object_unref (m_ringbuffer);
g_mutex_clear (&m_mutex);
}
virtual HRESULT RenderAudioSamples (bool preroll)
{
guint8 *ptr;
gint seg;
gint len;
gint bpf;
guint written, written_sum;
HRESULT res;
const GstAudioRingBufferSpec *spec =
&GST_AUDIO_RING_BUFFER_CAST (m_ringbuffer)->spec;
guint delay, max_delay;
GST_LOG_OBJECT (m_ringbuffer->sink, "Writing audio samples (preroll: %d)",
preroll);
delay =
gst_audio_ring_buffer_delay (GST_AUDIO_RING_BUFFER_CAST (m_ringbuffer));
max_delay = MAX ((spec->segtotal * spec->segsize) / 2, spec->segsize);
max_delay /= GST_AUDIO_INFO_BPF (&spec->info);
if (delay > max_delay) {
GstClock *clock =
gst_element_get_clock (GST_ELEMENT_CAST (m_ringbuffer->sink));
GstClockTime wait_time;
GstClockID clock_id;
GstClockReturn clock_ret;
GST_DEBUG_OBJECT (m_ringbuffer->sink, "Delay %u > max delay %u", delay,
max_delay);
wait_time =
gst_util_uint64_scale (delay - max_delay, GST_SECOND,
GST_AUDIO_INFO_RATE (&spec->info));
GST_DEBUG_OBJECT (m_ringbuffer->sink, "Waiting for %" GST_TIME_FORMAT,
GST_TIME_ARGS (wait_time));
wait_time += gst_clock_get_time (clock);
g_mutex_lock (&m_ringbuffer->clock_id_lock);
if (!GST_AUDIO_RING_BUFFER_CAST (m_ringbuffer)->acquired) {
GST_DEBUG_OBJECT (m_ringbuffer->sink,
"Ringbuffer not acquired anymore");
g_mutex_unlock (&m_ringbuffer->clock_id_lock);
gst_object_unref (clock);
return S_OK;
}
clock_id = gst_clock_new_single_shot_id (clock, wait_time);
m_ringbuffer->clock_id = clock_id;
g_mutex_unlock (&m_ringbuffer->clock_id_lock);
gst_object_unref (clock);
clock_ret = gst_clock_id_wait (clock_id, NULL);
g_mutex_lock (&m_ringbuffer->clock_id_lock);
gst_clock_id_unref (clock_id);
m_ringbuffer->clock_id = NULL;
g_mutex_unlock (&m_ringbuffer->clock_id_lock);
if (clock_ret == GST_CLOCK_UNSCHEDULED) {
GST_DEBUG_OBJECT (m_ringbuffer->sink, "Flushing");
return S_OK;
}
}
if (!gst_audio_ring_buffer_prepare_read (GST_AUDIO_RING_BUFFER_CAST
(m_ringbuffer), &seg, &ptr, &len)) {
GST_WARNING_OBJECT (m_ringbuffer->sink, "No segment available");
return E_FAIL;
}
bpf =
GST_AUDIO_INFO_BPF (&GST_AUDIO_RING_BUFFER_CAST (m_ringbuffer)->
spec.info);
len /= bpf;
GST_LOG_OBJECT (m_ringbuffer->sink,
"Write audio samples: %p size %d segment: %d", ptr, len, seg);
written_sum = 0;
do {
res =
m_ringbuffer->output->output->ScheduleAudioSamples (ptr, len,
0, 0, &written);
len -= written;
ptr += written * bpf;
written_sum += written;
} while (len > 0 && res == S_OK);
GST_LOG_OBJECT (m_ringbuffer->sink, "Wrote %u samples: 0x%08x", written_sum,
res);
gst_audio_ring_buffer_clear (GST_AUDIO_RING_BUFFER_CAST (m_ringbuffer),
seg);
gst_audio_ring_buffer_advance (GST_AUDIO_RING_BUFFER_CAST (m_ringbuffer),
1);
return res;
}
private:
GstDecklinkAudioSinkRingBuffer * m_ringbuffer;
GMutex m_mutex;
gint m_refcount;
};
static void
gst_decklink_audio_sink_ringbuffer_clear_all (GstAudioRingBuffer * rb)
{
GstDecklinkAudioSinkRingBuffer *self =
GST_DECKLINK_AUDIO_SINK_RING_BUFFER_CAST (rb);
GST_DEBUG_OBJECT (self->sink, "Flushing");
if (self->output)
self->output->output->FlushBufferedAudioSamples ();
}
static guint
gst_decklink_audio_sink_ringbuffer_delay (GstAudioRingBuffer * rb)
{
GstDecklinkAudioSinkRingBuffer *self =
GST_DECKLINK_AUDIO_SINK_RING_BUFFER_CAST (rb);
guint ret = 0;
HRESULT res = S_OK;
if (self->output) {
if ((res =
self->output->output->GetBufferedAudioSampleFrameCount (&ret)) !=
S_OK)
ret = 0;
}
GST_DEBUG_OBJECT (self->sink, "Delay: %u (0x%08x)", ret, res);
return ret;
}
#if 0
static gboolean
in_same_pipeline (GstElement * a, GstElement * b)
{
GstObject *root = NULL, *tmp;
gboolean ret = FALSE;
tmp = gst_object_get_parent (GST_OBJECT_CAST (a));
while (tmp != NULL) {
if (root)
gst_object_unref (root);
root = tmp;
tmp = gst_object_get_parent (root);
}
ret = root && gst_object_has_ancestor (GST_OBJECT_CAST (b), root);
if (root)
gst_object_unref (root);
return ret;
}
#endif
static gboolean
gst_decklink_audio_sink_ringbuffer_start (GstAudioRingBuffer * rb)
{
GstDecklinkAudioSinkRingBuffer *self =
GST_DECKLINK_AUDIO_SINK_RING_BUFFER_CAST (rb);
GstElement *videosink = NULL;
gboolean ret = TRUE;
// Check if there is a video sink for this output too and if it
// is actually in the same pipeline
g_mutex_lock (&self->output->lock);
if (self->output->videosink)
videosink = GST_ELEMENT_CAST (gst_object_ref (self->output->videosink));
g_mutex_unlock (&self->output->lock);
if (!videosink) {
GST_ELEMENT_ERROR (self->sink, STREAM, FAILED,
(NULL), ("Audio sink needs a video sink for its operation"));
ret = FALSE;
}
// FIXME: This causes deadlocks sometimes
#if 0
else if (!in_same_pipeline (GST_ELEMENT_CAST (self->sink), videosink)) {
GST_ELEMENT_ERROR (self->sink, STREAM, FAILED,
(NULL), ("Audio sink and video sink need to be in the same pipeline"));
ret = FALSE;
}
#endif
if (videosink)
gst_object_unref (videosink);
return ret;
}
static gboolean
gst_decklink_audio_sink_ringbuffer_pause (GstAudioRingBuffer * rb)
{
return TRUE;
}
static gboolean
gst_decklink_audio_sink_ringbuffer_stop (GstAudioRingBuffer * rb)
{
return TRUE;
}
static gboolean
gst_decklink_audio_sink_ringbuffer_acquire (GstAudioRingBuffer * rb,
GstAudioRingBufferSpec * spec)
{
GstDecklinkAudioSinkRingBuffer *self =
GST_DECKLINK_AUDIO_SINK_RING_BUFFER_CAST (rb);
HRESULT ret;
BMDAudioSampleType sample_depth;
GST_DEBUG_OBJECT (self->sink, "Acquire");
if (spec->info.finfo->format == GST_AUDIO_FORMAT_S16LE) {
sample_depth = bmdAudioSampleType16bitInteger;
} else {
sample_depth = bmdAudioSampleType32bitInteger;
}
ret = self->output->output->EnableAudioOutput (bmdAudioSampleRate48kHz,
sample_depth, spec->info.channels, bmdAudioOutputStreamContinuous);
if (ret != S_OK) {
GST_WARNING_OBJECT (self->sink, "Failed to enable audio output 0x%08x",
ret);
return FALSE;
}
ret =
self->output->
output->SetAudioCallback (new GStreamerAudioOutputCallback (self));
if (ret != S_OK) {
GST_WARNING_OBJECT (self->sink,
"Failed to set audio output callback 0x%08x", ret);
return FALSE;
}
spec->segsize =
(spec->latency_time * GST_AUDIO_INFO_RATE (&spec->info) /
G_USEC_PER_SEC) * GST_AUDIO_INFO_BPF (&spec->info);
spec->segtotal = spec->buffer_time / spec->latency_time;
// set latency to one more segment as we need some headroom
spec->seglatency = spec->segtotal + 1;
rb->size = spec->segtotal * spec->segsize;
rb->memory = (guint8 *) g_malloc0 (rb->size);
return TRUE;
}
static gboolean
gst_decklink_audio_sink_ringbuffer_release (GstAudioRingBuffer * rb)
{
GstDecklinkAudioSinkRingBuffer *self =
GST_DECKLINK_AUDIO_SINK_RING_BUFFER_CAST (rb);
GST_DEBUG_OBJECT (self->sink, "Release");
if (self->output) {
g_mutex_lock (&self->clock_id_lock);
if (self->clock_id)
gst_clock_id_unschedule (self->clock_id);
g_mutex_unlock (&self->clock_id_lock);
g_mutex_lock (&self->output->lock);
self->output->audio_enabled = FALSE;
if (self->output->start_scheduled_playback && self->output->videosink)
self->output->start_scheduled_playback (self->output->videosink);
g_mutex_unlock (&self->output->lock);
self->output->output->DisableAudioOutput ();
}
// free the buffer
g_free (rb->memory);
rb->memory = NULL;
return TRUE;
}
static gboolean
gst_decklink_audio_sink_ringbuffer_open_device (GstAudioRingBuffer * rb)
{
GstDecklinkAudioSinkRingBuffer *self =
GST_DECKLINK_AUDIO_SINK_RING_BUFFER_CAST (rb);
GST_DEBUG_OBJECT (self->sink, "Open device");
self->output =
gst_decklink_acquire_nth_output (self->sink->device_number,
GST_ELEMENT_CAST (self), TRUE);
if (!self->output) {
GST_ERROR_OBJECT (self, "Failed to acquire output");
return FALSE;
}
gst_decklink_output_set_audio_clock (self->output,
GST_AUDIO_BASE_SINK_CAST (self->sink)->provided_clock);
return TRUE;
}
static gboolean
gst_decklink_audio_sink_ringbuffer_close_device (GstAudioRingBuffer * rb)
{
GstDecklinkAudioSinkRingBuffer *self =
GST_DECKLINK_AUDIO_SINK_RING_BUFFER_CAST (rb);
GST_DEBUG_OBJECT (self->sink, "Close device");
if (self->output) {
gst_decklink_output_set_audio_clock (self->output, NULL);
gst_decklink_release_nth_output (self->sink->device_number,
GST_ELEMENT_CAST (self), TRUE);
self->output = NULL;
}
return TRUE;
}
enum
{
PROP_0,
PROP_DEVICE_NUMBER
};
static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS
("audio/x-raw, format={S16LE,S32LE}, channels={2, 8, 16}, rate=48000, "
"layout=interleaved")
);
static void gst_decklink_audio_sink_set_property (GObject * object,
guint property_id, const GValue * value, GParamSpec * pspec);
static void gst_decklink_audio_sink_get_property (GObject * object,
guint property_id, GValue * value, GParamSpec * pspec);
static void gst_decklink_audio_sink_finalize (GObject * object);
static GstStateChangeReturn gst_decklink_audio_sink_change_state (GstElement *
element, GstStateChange transition);
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static GstCaps *gst_decklink_audio_sink_get_caps (GstBaseSink * bsink,
GstCaps * filter);
static GstAudioRingBuffer
* gst_decklink_audio_sink_create_ringbuffer (GstAudioBaseSink * absink);
#define parent_class gst_decklink_audio_sink_parent_class
G_DEFINE_TYPE (GstDecklinkAudioSink, gst_decklink_audio_sink,
GST_TYPE_AUDIO_BASE_SINK);
static void
gst_decklink_audio_sink_class_init (GstDecklinkAudioSinkClass * klass)
{
GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
GstBaseSinkClass *basesink_class = GST_BASE_SINK_CLASS (klass);
GstAudioBaseSinkClass *audiobasesink_class =
GST_AUDIO_BASE_SINK_CLASS (klass);
gobject_class->set_property = gst_decklink_audio_sink_set_property;
gobject_class->get_property = gst_decklink_audio_sink_get_property;
gobject_class->finalize = gst_decklink_audio_sink_finalize;
element_class->change_state =
GST_DEBUG_FUNCPTR (gst_decklink_audio_sink_change_state);
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basesink_class->get_caps =
GST_DEBUG_FUNCPTR (gst_decklink_audio_sink_get_caps);
audiobasesink_class->create_ringbuffer =
GST_DEBUG_FUNCPTR (gst_decklink_audio_sink_create_ringbuffer);
g_object_class_install_property (gobject_class, PROP_DEVICE_NUMBER,
g_param_spec_int ("device-number", "Device number",
"Output device instance to use", 0, G_MAXINT, 0,
(GParamFlags) (G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS |
G_PARAM_CONSTRUCT)));
gst_element_class_add_static_pad_template (element_class, &sink_template);
gst_element_class_set_static_metadata (element_class, "Decklink Audio Sink",
"Audio/Sink", "Decklink Sink", "David Schleef <ds@entropywave.com>, "
"Sebastian Dröge <sebastian@centricular.com>");
GST_DEBUG_CATEGORY_INIT (gst_decklink_audio_sink_debug, "decklinkaudiosink",
0, "debug category for decklinkaudiosink element");
}
static void
gst_decklink_audio_sink_init (GstDecklinkAudioSink * self)
{
self->device_number = 0;
// 25.000ms latency time seems to be needed at least,
// everything below can cause drop-outs
// TODO: This is probably related to the video mode that
// is selected, but not directly it seems. Choosing the
// duration of a frame does not work.
GST_AUDIO_BASE_SINK_CAST (self)->latency_time = 25000;
}
void
gst_decklink_audio_sink_set_property (GObject * object, guint property_id,
const GValue * value, GParamSpec * pspec)
{
GstDecklinkAudioSink *self = GST_DECKLINK_AUDIO_SINK_CAST (object);
switch (property_id) {
case PROP_DEVICE_NUMBER:
self->device_number = g_value_get_int (value);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, property_id, pspec);
break;
}
}
void
gst_decklink_audio_sink_get_property (GObject * object, guint property_id,
GValue * value, GParamSpec * pspec)
{
GstDecklinkAudioSink *self = GST_DECKLINK_AUDIO_SINK_CAST (object);
switch (property_id) {
case PROP_DEVICE_NUMBER:
g_value_set_int (value, self->device_number);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, property_id, pspec);
break;
}
}
void
gst_decklink_audio_sink_finalize (GObject * object)
{
G_OBJECT_CLASS (parent_class)->finalize (object);
}
static GstStateChangeReturn
gst_decklink_audio_sink_change_state (GstElement * element,
GstStateChange transition)
{
GstDecklinkAudioSink *self = GST_DECKLINK_AUDIO_SINK_CAST (element);
GstDecklinkAudioSinkRingBuffer *buf =
GST_DECKLINK_AUDIO_SINK_RING_BUFFER_CAST (GST_AUDIO_BASE_SINK_CAST
(self)->ringbuffer);
GstStateChangeReturn ret;
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_PLAYING:
g_mutex_lock (&buf->output->lock);
buf->output->audio_enabled = TRUE;
if (buf->output->start_scheduled_playback && buf->output->videosink)
buf->output->start_scheduled_playback (buf->output->videosink);
g_mutex_unlock (&buf->output->lock);
break;
default:
break;
}
return ret;
}
static GstCaps *
gst_decklink_audio_sink_get_caps (GstBaseSink * bsink, GstCaps * filter)
{
GstDecklinkAudioSink *self = GST_DECKLINK_AUDIO_SINK_CAST (bsink);
GstDecklinkAudioSinkRingBuffer *buf =
GST_DECKLINK_AUDIO_SINK_RING_BUFFER_CAST (GST_AUDIO_BASE_SINK_CAST
(self)->ringbuffer);
GstCaps *caps = gst_pad_get_pad_template_caps (GST_BASE_SINK_PAD (bsink));
if (buf) {
GST_OBJECT_LOCK (buf);
if (buf->output && buf->output->attributes) {
int64_t max_channels = 0;
HRESULT ret;
GstStructure *s;
GValue arr = G_VALUE_INIT;
GValue v = G_VALUE_INIT;
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ret =
buf->output->attributes->GetInt (BMDDeckLinkMaximumAudioChannels,
&max_channels);
/* 2 should always be supported */
if (ret != S_OK) {
max_channels = 2;
}
caps = gst_caps_make_writable (caps);
s = gst_caps_get_structure (caps, 0);
g_value_init (&arr, GST_TYPE_LIST);
g_value_init (&v, G_TYPE_INT);
if (max_channels >= 16) {
g_value_set_int (&v, 16);
gst_value_list_append_value (&arr, &v);
}
if (max_channels >= 8) {
g_value_set_int (&v, 8);
gst_value_list_append_value (&arr, &v);
}
g_value_set_int (&v, 2);
gst_value_list_append_value (&arr, &v);
gst_structure_set_value (s, "channels", &arr);
g_value_unset (&v);
g_value_unset (&arr);
}
GST_OBJECT_UNLOCK (buf);
}
if (filter) {
2016-09-01 11:18:33 +00:00
GstCaps *intersection =
gst_caps_intersect_full (filter, caps, GST_CAPS_INTERSECT_FIRST);
gst_caps_unref (caps);
caps = intersection;
}
return caps;
}
static GstAudioRingBuffer *
gst_decklink_audio_sink_create_ringbuffer (GstAudioBaseSink * absink)
{
GstAudioRingBuffer *ret;
GST_DEBUG_OBJECT (absink, "Creating ringbuffer");
ret =
GST_AUDIO_RING_BUFFER_CAST (g_object_new
(GST_TYPE_DECKLINK_AUDIO_SINK_RING_BUFFER, NULL));
GST_DECKLINK_AUDIO_SINK_RING_BUFFER_CAST (ret)->sink =
(GstDecklinkAudioSink *) gst_object_ref (absink);
return ret;
}