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gstreamer/subprojects/gst-plugins-bad/sys/decklink/gstdecklinkaudiosink.cpp
Seungha Yang 7117cba831 decklinkaudiosink: Fix playback when video caps is configured before audio 
Scheduled playback starts on videosink's state change or on caps
but it's possible that audiosink is configure without caps yet.
Try start scheduled playback on audiosink's caps event as well

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4364>
2023-04-07 18:59:17 +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;
}
g_mutex_lock (&self->output->lock);
ret = self->output->output->EnableAudioOutput (bmdAudioSampleRate48kHz,
sample_depth, info.channels, bmdAudioOutputStreamContinuous);
if (ret != S_OK) {
g_mutex_unlock (&self->output->lock);
GST_WARNING_OBJECT (self, "Failed to enable audio output 0x%08lx",
(unsigned long) ret);
return FALSE;
}
self->output->audio_enabled = TRUE;
self->info = info;
if (self->output->start_scheduled_playback && self->output->videosink)
self->output->start_scheduled_playback (self->output->videosink);
g_mutex_unlock (&self->output->lock);
// 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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