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gstreamer/sys/decklink/gstdecklinkaudiosink.cpp
Sebastian Dröge 3e70f00209 decklinkaudiosink: Check also against the clock if we run ahead of the clock too much 
The buffer level as reported by the Decklink driver is completely
unreliable, and we could otherwise easily run ahead >1s if we're
unlucky.

https://bugzilla.gnome.org/show_bug.cgi?id=790114
2017-12-14 10:37:20 +02:00

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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.
*/
#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)
enum
{
PROP_0,
PROP_DEVICE_NUMBER,
PROP_HW_SERIAL_NUMBER,
PROP_ALIGNMENT_THRESHOLD,
PROP_DISCONT_WAIT,
PROP_BUFFER_TIME,
};
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);
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)));
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", "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;
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;
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;
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;
}
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;
GstClockTime 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;
if (clock_now < running_time)
clock_ahead = running_time - clock_now;
}
}
GST_DEBUG_OBJECT (self,
"Ahead %" GST_TIME_FORMAT " of the clock running time",
GST_TIME_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);
// We start waiting once we have more than buffer-time buffered
if (buffered_time > self->buffer_time || 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");
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);
if (!self->output->started) {
GST_LOG_OBJECT (self, "Writing audio frame synchronously because PAUSED");
ret =
self->output->output->WriteAudioSamplesSync ((void *) data, len,
&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_WARNING (self, STREAM, FAILED,
(NULL), ("Failed to write audio frame synchronously: 0x%08lx",
(unsigned long) ret));
flow_ret = GST_FLOW_ERROR;
break;
} else {
flow_ret = GST_FLOW_FLUSHING;
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;
} else {
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 {
flow_ret = GST_FLOW_FLUSHING;
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);
return flow_ret;
}
static gboolean
gst_decklink_audio_sink_open (GstBaseSink * bsink)
{
GstDecklinkAudioSink *self = GST_DECKLINK_AUDIO_SINK_CAST (bsink);
GST_DEBUG_OBJECT (self, "Stopping");
self->output =
gst_decklink_acquire_nth_output (self->device_number,
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,
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);
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_sink_stop (self);
break;
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:{
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;
}
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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