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gstreamer/subprojects/gst-plugins-bad/sys/decklink/gstdecklinkaudiosink.cpp
Wojciech Kapsa 505f48f237 decklink: Add new persistent-id property and sort devices by persistent ID 
The order of the devices iterator from the SDK is undefined and can
randomly change.

Keep the device-number property for backwards compatibility and
simplicity but prefer the persistent-id property and also use it for the
device provider implementation.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3078>
2022-10-06 15:50:11 +00:00

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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.
*/
/**
* SECTION:element-decklinkaudiosink
* @short_description: Outputs Audio to a BlackMagic DeckLink Device
* @see_also: decklinkvideosink
*
* Playout Video and Audio to a BlackMagic DeckLink Device. Can only be used
* in conjunction with decklinkvideosink.
*
* ## Sample pipeline
* |[
* gst-launch-1.0 \
* videotestsrc ! decklinkvideosink device-number=0 mode=1080p25 \
* audiotestsrc ! decklinkaudiosink device-number=0
* ]|
* Playout a 1080p25 test-video with a test-audio signal to the SDI-Out of Card 0.
* Devices are numbered starting with 0.
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include "gstdecklinkaudiosink.h"
#include "gstdecklinkvideosink.h"
#include <string.h>
GST_DEBUG_CATEGORY_STATIC (gst_decklink_audio_sink_debug);
#define GST_CAT_DEFAULT gst_decklink_audio_sink_debug
#define DEFAULT_DEVICE_NUMBER (0)
#define DEFAULT_ALIGNMENT_THRESHOLD (40 * GST_MSECOND)
#define DEFAULT_DISCONT_WAIT (1 * GST_SECOND)
// Microseconds for audiobasesink compatibility...
#define DEFAULT_BUFFER_TIME (50 * GST_MSECOND / 1000)
#define DEFAULT_PERSISTENT_ID (-1)
enum
{
PROP_0,
PROP_DEVICE_NUMBER,
PROP_HW_SERIAL_NUMBER,
PROP_ALIGNMENT_THRESHOLD,
PROP_DISCONT_WAIT,
PROP_BUFFER_TIME,
PROP_PERSISTENT_ID
};
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);
static GstClock *gst_decklink_audio_sink_provide_clock (GstElement * element);
static GstCaps *gst_decklink_audio_sink_get_caps (GstBaseSink * bsink,
GstCaps * filter);
static gboolean gst_decklink_audio_sink_set_caps (GstBaseSink * bsink,
GstCaps * caps);
static GstFlowReturn gst_decklink_audio_sink_render (GstBaseSink * bsink,
GstBuffer * buffer);
static gboolean gst_decklink_audio_sink_open (GstBaseSink * bsink);
static gboolean gst_decklink_audio_sink_close (GstBaseSink * bsink);
static gboolean gst_decklink_audio_sink_stop (GstDecklinkAudioSink * self);
static gboolean gst_decklink_audio_sink_unlock_stop (GstBaseSink * bsink);
static void gst_decklink_audio_sink_get_times (GstBaseSink * bsink,
GstBuffer * buffer, GstClockTime * start, GstClockTime * end);
static gboolean gst_decklink_audio_sink_query (GstBaseSink * bsink,
GstQuery * query);
static gboolean gst_decklink_audio_sink_event (GstBaseSink * bsink,
GstEvent * event);
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")
);
#define parent_class gst_decklink_audio_sink_parent_class
G_DEFINE_TYPE (GstDecklinkAudioSink, gst_decklink_audio_sink,
GST_TYPE_BASE_SINK);
GST_ELEMENT_REGISTER_DEFINE_WITH_CODE (decklinkaudiosink, "decklinkaudiosink", GST_RANK_NONE,
GST_TYPE_DECKLINK_AUDIO_SINK, decklink_element_init (plugin));
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);
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);
element_class->provide_clock =
GST_DEBUG_FUNCPTR (gst_decklink_audio_sink_provide_clock);
basesink_class->get_caps =
GST_DEBUG_FUNCPTR (gst_decklink_audio_sink_get_caps);
basesink_class->set_caps =
GST_DEBUG_FUNCPTR (gst_decklink_audio_sink_set_caps);
basesink_class->render = GST_DEBUG_FUNCPTR (gst_decklink_audio_sink_render);
// FIXME: These are misnamed in basesink!
basesink_class->start = GST_DEBUG_FUNCPTR (gst_decklink_audio_sink_open);
basesink_class->stop = GST_DEBUG_FUNCPTR (gst_decklink_audio_sink_close);
basesink_class->unlock_stop =
GST_DEBUG_FUNCPTR (gst_decklink_audio_sink_unlock_stop);
basesink_class->get_times =
GST_DEBUG_FUNCPTR (gst_decklink_audio_sink_get_times);
basesink_class->query = GST_DEBUG_FUNCPTR (gst_decklink_audio_sink_query);
basesink_class->event = GST_DEBUG_FUNCPTR (gst_decklink_audio_sink_event);
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, DEFAULT_DEVICE_NUMBER,
(GParamFlags) (G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS |
G_PARAM_CONSTRUCT)));
/**
* GstDecklinkAudioSink:persistent-id
*
* Decklink device to use. Higher priority than "device-number".
* BMDDeckLinkPersistentID is a device specific, 32-bit unique identifier.
* It is stable even when the device is plugged in a different connector,
* across reboots, and when plugged into different computers.
*
* Since: 1.22
*/
g_object_class_install_property (gobject_class, PROP_PERSISTENT_ID,
g_param_spec_int64 ("persistent-id", "Persistent id",
"Output device instance to use. Higher priority than \"device-number\".",
DEFAULT_PERSISTENT_ID, G_MAXINT64, DEFAULT_PERSISTENT_ID,
(GParamFlags) (G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS |
G_PARAM_CONSTRUCT)));
g_object_class_install_property (gobject_class, PROP_HW_SERIAL_NUMBER,
g_param_spec_string ("hw-serial-number", "Hardware serial number",
"The serial number (hardware ID) of the Decklink card",
NULL, (GParamFlags) (G_PARAM_READABLE | G_PARAM_STATIC_STRINGS)));
g_object_class_install_property (gobject_class, PROP_ALIGNMENT_THRESHOLD,
g_param_spec_uint64 ("alignment-threshold", "Alignment Threshold",
"Timestamp alignment threshold in nanoseconds", 0,
G_MAXUINT64 - 1, DEFAULT_ALIGNMENT_THRESHOLD,
(GParamFlags) (G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS |
GST_PARAM_MUTABLE_READY)));
g_object_class_install_property (gobject_class, PROP_DISCONT_WAIT,
g_param_spec_uint64 ("discont-wait", "Discont Wait",
"Window of time in nanoseconds to wait before "
"creating a discontinuity", 0,
G_MAXUINT64 - 1, DEFAULT_DISCONT_WAIT,
(GParamFlags) (G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS |
GST_PARAM_MUTABLE_READY)));
g_object_class_install_property (gobject_class, PROP_BUFFER_TIME,
g_param_spec_uint64 ("buffer-time", "Buffer Time",
"Size of audio buffer in microseconds, this is the minimum latency that the sink reports",
0, G_MAXUINT64, DEFAULT_BUFFER_TIME,
(GParamFlags) (G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS |
GST_PARAM_MUTABLE_READY)));
gst_element_class_add_static_pad_template (element_class, &sink_template);
gst_element_class_set_static_metadata (element_class, "Decklink Audio Sink",
"Audio/Sink/Hardware", "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 = DEFAULT_DEVICE_NUMBER;
self->stream_align =
gst_audio_stream_align_new (48000, DEFAULT_ALIGNMENT_THRESHOLD,
DEFAULT_DISCONT_WAIT);
self->buffer_time = DEFAULT_BUFFER_TIME * 1000;
self->persistent_id = DEFAULT_PERSISTENT_ID;
gst_base_sink_set_max_lateness (GST_BASE_SINK_CAST (self), 20 * GST_MSECOND);
}
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;
case PROP_ALIGNMENT_THRESHOLD:
GST_OBJECT_LOCK (self);
gst_audio_stream_align_set_alignment_threshold (self->stream_align,
g_value_get_uint64 (value));
GST_OBJECT_UNLOCK (self);
break;
case PROP_DISCONT_WAIT:
GST_OBJECT_LOCK (self);
gst_audio_stream_align_set_discont_wait (self->stream_align,
g_value_get_uint64 (value));
GST_OBJECT_UNLOCK (self);
break;
case PROP_BUFFER_TIME:
GST_OBJECT_LOCK (self);
self->buffer_time = g_value_get_uint64 (value) * 1000;
GST_OBJECT_UNLOCK (self);
break;
case PROP_PERSISTENT_ID:
self->persistent_id = g_value_get_int64 (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;
case PROP_HW_SERIAL_NUMBER:
if (self->output)
g_value_set_string (value, self->output->hw_serial_number);
else
g_value_set_string (value, NULL);
break;
case PROP_ALIGNMENT_THRESHOLD:
GST_OBJECT_LOCK (self);
g_value_set_uint64 (value,
gst_audio_stream_align_get_alignment_threshold (self->stream_align));
GST_OBJECT_UNLOCK (self);
break;
case PROP_DISCONT_WAIT:
GST_OBJECT_LOCK (self);
g_value_set_uint64 (value,
gst_audio_stream_align_get_discont_wait (self->stream_align));
GST_OBJECT_UNLOCK (self);
break;
case PROP_BUFFER_TIME:
GST_OBJECT_LOCK (self);
g_value_set_uint64 (value, self->buffer_time / 1000);
GST_OBJECT_UNLOCK (self);
break;
case PROP_PERSISTENT_ID:
g_value_set_int64 (value, self->persistent_id);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, property_id, pspec);
break;
}
}
void
gst_decklink_audio_sink_finalize (GObject * object)
{
GstDecklinkAudioSink *self = GST_DECKLINK_AUDIO_SINK_CAST (object);
if (self->stream_align) {
gst_audio_stream_align_free (self->stream_align);
self->stream_align = NULL;
}
G_OBJECT_CLASS (parent_class)->finalize (object);
}
static gboolean
gst_decklink_audio_sink_set_caps (GstBaseSink * bsink, GstCaps * caps)
{
GstDecklinkAudioSink *self = GST_DECKLINK_AUDIO_SINK_CAST (bsink);
HRESULT ret;
BMDAudioSampleType sample_depth;
GstAudioInfo info;
GST_DEBUG_OBJECT (self, "Setting caps %" GST_PTR_FORMAT, caps);
if (!gst_audio_info_from_caps (&info, caps))
return FALSE;
if (self->output->audio_enabled
&& (self->info.finfo->format != info.finfo->format
|| self->info.channels != info.channels)) {
GST_ERROR_OBJECT (self, "Reconfiguration not supported");
return FALSE;
} else if (self->output->audio_enabled) {
return TRUE;
}
if (info.finfo->format == GST_AUDIO_FORMAT_S16LE) {
sample_depth = bmdAudioSampleType16bitInteger;
} else {
sample_depth = bmdAudioSampleType32bitInteger;
}
ret = self->output->output->EnableAudioOutput (bmdAudioSampleRate48kHz,
sample_depth, info.channels, bmdAudioOutputStreamContinuous);
if (ret != S_OK) {
GST_WARNING_OBJECT (self, "Failed to enable audio output 0x%08lx",
(unsigned long) ret);
return FALSE;
}
self->output->audio_enabled = TRUE;
self->info = info;
// Create a new resampler as needed
if (self->resampler)
gst_audio_resampler_free (self->resampler);
self->resampler = NULL;
return TRUE;
}
static GstCaps *
gst_decklink_audio_sink_get_caps (GstBaseSink * bsink, GstCaps * filter)
{
GstDecklinkAudioSink *self = GST_DECKLINK_AUDIO_SINK_CAST (bsink);
GstCaps *caps;
if ((caps = gst_pad_get_current_caps (GST_BASE_SINK_PAD (bsink))))
return caps;
caps = gst_pad_get_pad_template_caps (GST_BASE_SINK_PAD (bsink));
GST_OBJECT_LOCK (self);
if (self->output && self->output->attributes) {
int64_t max_channels = 0;
HRESULT ret;
GstStructure *s;
GValue arr = G_VALUE_INIT;
GValue v = G_VALUE_INIT;
ret =
self->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 (self);
if (filter) {
GstCaps *intersection =
gst_caps_intersect_full (filter, caps, GST_CAPS_INTERSECT_FIRST);
gst_caps_unref (caps);
caps = intersection;
}
return caps;
}
static gboolean
gst_decklink_audio_sink_query (GstBaseSink * bsink, GstQuery * query)
{
GstDecklinkAudioSink *self = GST_DECKLINK_AUDIO_SINK (bsink);
gboolean res = FALSE;
switch (GST_QUERY_TYPE (query)) {
case GST_QUERY_LATENCY:
{
gboolean live, us_live;
GstClockTime min_l, max_l;
GST_DEBUG_OBJECT (self, "latency query");
/* ask parent first, it will do an upstream query for us. */
if ((res =
gst_base_sink_query_latency (GST_BASE_SINK_CAST (self), &live,
&us_live, &min_l, &max_l))) {
GstClockTime base_latency, min_latency, max_latency;
/* we and upstream are both live, adjust the min_latency */
if (live && us_live) {
GST_OBJECT_LOCK (self);
if (!self->info.rate) {
GST_OBJECT_UNLOCK (self);
GST_DEBUG_OBJECT (self,
"we are not negotiated, can't report latency yet");
res = FALSE;
goto done;
}
base_latency = self->buffer_time * 1000;
GST_OBJECT_UNLOCK (self);
/* we cannot go lower than the buffer size and the min peer latency */
min_latency = base_latency + min_l;
/* the max latency is the max of the peer, we can delay an infinite
* amount of time. */
max_latency =
(max_l ==
GST_CLOCK_TIME_NONE) ? GST_CLOCK_TIME_NONE : (base_latency +
max_l);
GST_DEBUG_OBJECT (self,
"peer min %" GST_TIME_FORMAT ", our min latency: %"
GST_TIME_FORMAT, GST_TIME_ARGS (min_l),
GST_TIME_ARGS (min_latency));
GST_DEBUG_OBJECT (self,
"peer max %" GST_TIME_FORMAT ", our max latency: %"
GST_TIME_FORMAT, GST_TIME_ARGS (max_l),
GST_TIME_ARGS (max_latency));
} else {
GST_DEBUG_OBJECT (self,
"peer or we are not live, don't care about latency");
min_latency = min_l;
max_latency = max_l;
}
gst_query_set_latency (query, live, min_latency, max_latency);
}
break;
}
default:
res = GST_BASE_SINK_CLASS (parent_class)->query (bsink, query);
break;
}
done:
return res;
}
static gboolean
gst_decklink_audio_sink_event (GstBaseSink * bsink, GstEvent * event)
{
GstDecklinkAudioSink *self = GST_DECKLINK_AUDIO_SINK_CAST (bsink);
if (GST_EVENT_TYPE (event) == GST_EVENT_SEGMENT) {
const GstSegment *new_segment;
gst_event_parse_segment (event, &new_segment);
if (ABS (new_segment->rate) != 1.0) {
guint out_rate = self->info.rate / ABS (new_segment->rate);
if (self->resampler && (self->resampler_out_rate != out_rate
|| self->resampler_in_rate != (guint) self->info.rate))
gst_audio_resampler_update (self->resampler, self->info.rate, out_rate,
NULL);
else if (!self->resampler)
self->resampler =
gst_audio_resampler_new (GST_AUDIO_RESAMPLER_METHOD_LINEAR,
GST_AUDIO_RESAMPLER_FLAG_NONE, self->info.finfo->format,
self->info.channels, self->info.rate, out_rate, NULL);
self->resampler_in_rate = self->info.rate;
self->resampler_out_rate = out_rate;
} else if (self->resampler) {
gst_audio_resampler_free (self->resampler);
self->resampler = NULL;
}
if (new_segment->rate < 0)
gst_audio_stream_align_set_rate (self->stream_align, -48000);
}
return GST_BASE_SINK_CLASS (parent_class)->event (bsink, event);
}
static GstFlowReturn
gst_decklink_audio_sink_render (GstBaseSink * bsink, GstBuffer * buffer)
{
GstDecklinkAudioSink *self = GST_DECKLINK_AUDIO_SINK_CAST (bsink);
GstDecklinkVideoSink *video_sink;
GstFlowReturn flow_ret;
HRESULT ret;
GstClockTime timestamp, duration;
GstClockTime running_time, running_time_duration;
GstClockTime schedule_time, schedule_time_duration;
GstClockTime latency, render_delay;
GstClockTimeDiff ts_offset;
GstMapInfo map_info;
const guint8 *data;
gsize len, written_all;
gboolean discont;
GST_DEBUG_OBJECT (self, "Rendering buffer %p", buffer);
// FIXME: Handle no timestamps
if (!GST_BUFFER_TIMESTAMP_IS_VALID (buffer)) {
return GST_FLOW_ERROR;
}
if (GST_BASE_SINK_CAST (self)->flushing) {
return GST_FLOW_FLUSHING;
}
// If we're called before output is actually started, start pre-rolling
if (!self->output->started) {
self->output->output->BeginAudioPreroll ();
}
video_sink =
GST_DECKLINK_VIDEO_SINK (gst_object_ref (self->output->videosink));
timestamp = GST_BUFFER_TIMESTAMP (buffer);
duration = GST_BUFFER_DURATION (buffer);
discont = gst_audio_stream_align_process (self->stream_align,
GST_BUFFER_IS_DISCONT (buffer), timestamp,
gst_buffer_get_size (buffer) / self->info.bpf, &timestamp, &duration,
NULL);
if (discont && self->resampler)
gst_audio_resampler_reset (self->resampler);
if (GST_BASE_SINK_CAST (self)->segment.rate < 0.0) {
GstMapInfo out_map;
gint out_frames = gst_buffer_get_size (buffer) / self->info.bpf;
buffer = gst_buffer_make_writable (gst_buffer_ref (buffer));
gst_buffer_map (buffer, &out_map, GST_MAP_READWRITE);
if (self->info.finfo->format == GST_AUDIO_FORMAT_S16) {
gint16 *swap_data = (gint16 *) out_map.data;
gint16 *swap_data_end =
swap_data + (out_frames - 1) * self->info.channels;
gint16 swap_tmp[16];
while (out_frames > 0) {
memcpy (&swap_tmp, swap_data, self->info.bpf);
memcpy (swap_data, swap_data_end, self->info.bpf);
memcpy (swap_data_end, &swap_tmp, self->info.bpf);
swap_data += self->info.channels;
swap_data_end -= self->info.channels;
out_frames -= 2;
}
} else {
gint32 *swap_data = (gint32 *) out_map.data;
gint32 *swap_data_end =
swap_data + (out_frames - 1) * self->info.channels;
gint32 swap_tmp[16];
while (out_frames > 0) {
memcpy (&swap_tmp, swap_data, self->info.bpf);
memcpy (swap_data, swap_data_end, self->info.bpf);
memcpy (swap_data_end, &swap_tmp, self->info.bpf);
swap_data += self->info.channels;
swap_data_end -= self->info.channels;
out_frames -= 2;
}
}
gst_buffer_unmap (buffer, &out_map);
} else {
gst_buffer_ref (buffer);
}
if (self->resampler) {
gint in_frames = gst_buffer_get_size (buffer) / self->info.bpf;
gint out_frames =
gst_audio_resampler_get_out_frames (self->resampler, in_frames);
GstBuffer *out_buf = gst_buffer_new_and_alloc (out_frames * self->info.bpf);
GstMapInfo out_map;
gst_buffer_map (buffer, &map_info, GST_MAP_READ);
gst_buffer_map (out_buf, &out_map, GST_MAP_READWRITE);
gst_audio_resampler_resample (self->resampler, (gpointer *) & map_info.data,
in_frames, (gpointer *) & out_map.data, out_frames);
gst_buffer_unmap (out_buf, &out_map);
gst_buffer_unmap (buffer, &map_info);
buffer = out_buf;
}
gst_buffer_map (buffer, &map_info, GST_MAP_READ);
data = map_info.data;
len = map_info.size / self->info.bpf;
written_all = 0;
do {
GstClockTime timestamp_now =
timestamp + gst_util_uint64_scale (written_all, GST_SECOND,
self->info.rate);
guint32 buffered_samples;
GstClockTime buffered_time;
guint32 written = 0;
GstClock *clock;
GstClockTimeDiff clock_ahead;
if (GST_BASE_SINK_CAST (self)->flushing) {
flow_ret = GST_FLOW_FLUSHING;
break;
}
running_time =
gst_segment_to_running_time (&GST_BASE_SINK_CAST (self)->segment,
GST_FORMAT_TIME, timestamp_now);
running_time_duration =
gst_segment_to_running_time (&GST_BASE_SINK_CAST (self)->segment,
GST_FORMAT_TIME, timestamp_now + duration) - running_time;
/* See gst_base_sink_adjust_time() */
latency = gst_base_sink_get_latency (bsink);
render_delay = gst_base_sink_get_render_delay (bsink);
ts_offset = gst_base_sink_get_ts_offset (bsink);
running_time += latency;
if (ts_offset < 0) {
ts_offset = -ts_offset;
if ((GstClockTime) ts_offset < running_time)
running_time -= ts_offset;
else
running_time = 0;
} else {
running_time += ts_offset;
}
if (running_time > render_delay)
running_time -= render_delay;
else
running_time = 0;
clock = gst_element_get_clock (GST_ELEMENT_CAST (self));
clock_ahead = 0;
if (clock) {
GstClockTime clock_now = gst_clock_get_time (clock);
GstClockTime base_time =
gst_element_get_base_time (GST_ELEMENT_CAST (self));
gst_object_unref (clock);
clock = NULL;
if (clock_now != GST_CLOCK_TIME_NONE && base_time != GST_CLOCK_TIME_NONE) {
GST_DEBUG_OBJECT (self,
"Clock time %" GST_TIME_FORMAT ", base time %" GST_TIME_FORMAT
", target running time %" GST_TIME_FORMAT,
GST_TIME_ARGS (clock_now), GST_TIME_ARGS (base_time),
GST_TIME_ARGS (running_time));
if (clock_now > base_time)
clock_now -= base_time;
else
clock_now = 0;
clock_ahead = running_time - clock_now;
}
}
GST_DEBUG_OBJECT (self,
"Ahead %" GST_STIME_FORMAT " of the clock running time",
GST_STIME_ARGS (clock_ahead));
if (self->output->
output->GetBufferedAudioSampleFrameCount (&buffered_samples) != S_OK)
buffered_samples = 0;
buffered_time =
gst_util_uint64_scale (buffered_samples, GST_SECOND, self->info.rate);
buffered_time /= ABS (GST_BASE_SINK_CAST (self)->segment.rate);
GST_DEBUG_OBJECT (self,
"Buffered %" GST_TIME_FORMAT " in the driver (%u samples)",
GST_TIME_ARGS (buffered_time), buffered_samples);
{
GstClockTimeDiff buffered_ahead_of_clock_ahead =
GST_CLOCK_DIFF (clock_ahead, buffered_time);
GST_DEBUG_OBJECT (self, "driver is %" GST_STIME_FORMAT " ahead of the "
"expected clock", GST_STIME_ARGS (buffered_ahead_of_clock_ahead));
/* we don't want to store too much data in the driver as decklink
* doesn't seem to actually use our provided timestamps to perform its
* own synchronisation. It seems to count samples instead. */
/* FIXME: do we need to split buffers? */
if (buffered_ahead_of_clock_ahead > 0 &&
buffered_ahead_of_clock_ahead >
gst_base_sink_get_max_lateness (bsink)) {
GST_DEBUG_OBJECT (self,
"Dropping buffer that is %" GST_STIME_FORMAT " too late",
GST_STIME_ARGS (buffered_ahead_of_clock_ahead));
if (self->resampler)
gst_audio_resampler_reset (self->resampler);
flow_ret = GST_FLOW_OK;
break;
}
}
// We start waiting once we have more than buffer-time buffered
if (((GstClockTime) clock_ahead) > self->buffer_time) {
GstClockReturn clock_ret;
GstClockTime wait_time = running_time;
GST_DEBUG_OBJECT (self,
"Buffered enough, wait for preroll or the clock or flushing. "
"Configured buffer time: %" GST_TIME_FORMAT,
GST_TIME_ARGS (self->buffer_time));
if (wait_time < self->buffer_time)
wait_time = 0;
else
wait_time -= self->buffer_time;
flow_ret =
gst_base_sink_do_preroll (GST_BASE_SINK_CAST (self),
GST_MINI_OBJECT_CAST (buffer));
if (flow_ret != GST_FLOW_OK)
break;
clock_ret =
gst_base_sink_wait_clock (GST_BASE_SINK_CAST (self), wait_time, NULL);
if (GST_BASE_SINK_CAST (self)->flushing) {
flow_ret = GST_FLOW_FLUSHING;
break;
}
// Rerun the whole loop again
if (clock_ret == GST_CLOCK_UNSCHEDULED)
continue;
}
schedule_time = running_time;
schedule_time_duration = running_time_duration;
gst_decklink_video_sink_convert_to_internal_clock (video_sink,
&schedule_time, &schedule_time_duration);
GST_LOG_OBJECT (self, "Scheduling audio samples at %" GST_TIME_FORMAT
" with duration %" GST_TIME_FORMAT, GST_TIME_ARGS (schedule_time),
GST_TIME_ARGS (schedule_time_duration));
ret = self->output->output->ScheduleAudioSamples ((void *) data, len,
schedule_time, GST_SECOND, &written);
if (ret != S_OK) {
bool is_running = true;
self->output->output->IsScheduledPlaybackRunning (&is_running);
if (is_running && !GST_BASE_SINK_CAST (self)->flushing
&& self->output->started) {
GST_ELEMENT_ERROR (self, STREAM, FAILED, (NULL),
("Failed to schedule frame: 0x%08lx", (unsigned long) ret));
flow_ret = GST_FLOW_ERROR;
break;
} else {
// Ignore the error and go out of the loop here, we're shutting down
// or are not started yet and there's nothing we can do at this point
GST_INFO_OBJECT (self,
"Ignoring scheduling error 0x%08x because we're not started yet"
" or not anymore", (guint) ret);
flow_ret = GST_FLOW_OK;
break;
}
}
len -= written;
data += written * self->info.bpf;
if (self->resampler)
written_all += written * ABS (GST_BASE_SINK_CAST (self)->segment.rate);
else
written_all += written;
flow_ret = GST_FLOW_OK;
} while (len > 0);
gst_buffer_unmap (buffer, &map_info);
gst_buffer_unref (buffer);
GST_DEBUG_OBJECT (self, "Returning %s", gst_flow_get_name (flow_ret));
return flow_ret;
}
static gboolean
gst_decklink_audio_sink_open (GstBaseSink * bsink)
{
GstDecklinkAudioSink *self = GST_DECKLINK_AUDIO_SINK_CAST (bsink);
GST_DEBUG_OBJECT (self, "Starting");
self->output =
gst_decklink_acquire_nth_output (self->device_number, self->persistent_id,
GST_ELEMENT_CAST (self), TRUE);
if (!self->output) {
GST_ERROR_OBJECT (self, "Failed to acquire output");
return FALSE;
}
g_object_notify (G_OBJECT (self), "hw-serial-number");
return TRUE;
}
static gboolean
gst_decklink_audio_sink_close (GstBaseSink * bsink)
{
GstDecklinkAudioSink *self = GST_DECKLINK_AUDIO_SINK_CAST (bsink);
GST_DEBUG_OBJECT (self, "Closing");
if (self->output) {
g_mutex_lock (&self->output->lock);
self->output->mode = NULL;
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 ();
gst_decklink_release_nth_output (self->device_number, self->persistent_id,
GST_ELEMENT_CAST (self), TRUE);
self->output = NULL;
}
return TRUE;
}
static gboolean
gst_decklink_audio_sink_stop (GstDecklinkAudioSink * self)
{
GST_DEBUG_OBJECT (self, "Stopping");
if (self->output && self->output->audio_enabled) {
g_mutex_lock (&self->output->lock);
self->output->audio_enabled = FALSE;
g_mutex_unlock (&self->output->lock);
self->output->output->DisableAudioOutput ();
}
if (self->resampler) {
gst_audio_resampler_free (self->resampler);
self->resampler = NULL;
}
return TRUE;
}
static gboolean
gst_decklink_audio_sink_unlock_stop (GstBaseSink * bsink)
{
GstDecklinkAudioSink *self = GST_DECKLINK_AUDIO_SINK (bsink);
if (self->output) {
self->output->output->FlushBufferedAudioSamples ();
}
return TRUE;
}
static void
gst_decklink_audio_sink_get_times (GstBaseSink * bsink, GstBuffer * buffer,
GstClockTime * start, GstClockTime * end)
{
/* our clock sync is a bit too much for the base class to handle so
* we implement it ourselves. */
*start = GST_CLOCK_TIME_NONE;
*end = GST_CLOCK_TIME_NONE;
}
static GstStateChangeReturn
gst_decklink_audio_sink_change_state (GstElement * element,
GstStateChange transition)
{
GstDecklinkAudioSink *self = GST_DECKLINK_AUDIO_SINK_CAST (element);
GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
switch (transition) {
case GST_STATE_CHANGE_READY_TO_PAUSED:
GST_OBJECT_LOCK (self);
gst_audio_stream_align_mark_discont (self->stream_align);
GST_OBJECT_UNLOCK (self);
g_mutex_lock (&self->output->lock);
if (self->output->start_scheduled_playback)
self->output->start_scheduled_playback (self->output->videosink);
g_mutex_unlock (&self->output->lock);
break;
case GST_STATE_CHANGE_PAUSED_TO_READY:
gst_decklink_audio_sink_stop (self);
break;
default:
break;
}
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
if (ret == GST_STATE_CHANGE_FAILURE)
return ret;
switch (transition) {
default:
break;
}
return ret;
}
static GstClock *
gst_decklink_audio_sink_provide_clock (GstElement * element)
{
GstDecklinkAudioSink *self = GST_DECKLINK_AUDIO_SINK_CAST (element);
if (!self->output)
return NULL;
return GST_CLOCK_CAST (gst_object_ref (self->output->clock));
}
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