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gstreamer/sys/decklink/gstdecklinkaudiosrc.cpp
Vivia Nikolaidou b5ff71fc3c decklinkaudiosrc: Extrapolate stream/hw reference timestamps when video frame is missing 
Sometimes we might get an audio packet without a corresponding video
frame. In these cases, the stream and hardware reference timestamps
would be missing, because they're called on the video frame. Instead of
potentially breaking stuff downstream that might depend on these, we now
extrapolate them.

https://bugzilla.gnome.org/show_bug.cgi?id=792042
2018-01-04 15:51:16 +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 "gstdecklinkaudiosrc.h"
#include "gstdecklinkvideosrc.h"
#include <string.h>
GST_DEBUG_CATEGORY_STATIC (gst_decklink_audio_src_debug);
#define GST_CAT_DEFAULT gst_decklink_audio_src_debug
#define DEFAULT_CONNECTION (GST_DECKLINK_AUDIO_CONNECTION_AUTO)
#define DEFAULT_BUFFER_SIZE (5)
#define DEFAULT_ALIGNMENT_THRESHOLD (40 * GST_MSECOND)
#define DEFAULT_DISCONT_WAIT (1 * GST_SECOND)
#define DEFAULT_CHANNELS (GST_DECKLINK_AUDIO_CHANNELS_2)
#ifndef ABSDIFF
#define ABSDIFF(x, y) ( (x) > (y) ? ((x) - (y)) : ((y) - (x)) )
#endif
enum
{
PROP_0,
PROP_CONNECTION,
PROP_DEVICE_NUMBER,
PROP_ALIGNMENT_THRESHOLD,
PROP_DISCONT_WAIT,
PROP_BUFFER_SIZE,
PROP_CHANNELS,
PROP_HW_SERIAL_NUMBER
};
static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS
("audio/x-raw, format={S16LE,S32LE}, channels=2, rate=48000, "
"layout=interleaved;"
"audio/x-raw, format={S16LE,S32LE}, channels={8,16}, channel-mask=(bitmask)0, rate=48000, "
"layout=interleaved")
);
typedef struct
{
IDeckLinkAudioInputPacket *packet;
GstClockTime timestamp;
GstClockTime stream_timestamp;
GstClockTime stream_duration;
GstClockTime hardware_timestamp;
GstClockTime hardware_duration;
gboolean no_signal;
} CapturePacket;
static void
capture_packet_clear (CapturePacket * packet)
{
packet->packet->Release ();
memset (packet, 0, sizeof (*packet));
}
typedef struct
{
IDeckLinkAudioInputPacket *packet;
IDeckLinkInput *input;
} AudioPacket;
static void
audio_packet_free (void *data)
{
AudioPacket *packet = (AudioPacket *) data;
packet->packet->Release ();
packet->input->Release ();
g_free (packet);
}
static void gst_decklink_audio_src_set_property (GObject * object,
guint property_id, const GValue * value, GParamSpec * pspec);
static void gst_decklink_audio_src_get_property (GObject * object,
guint property_id, GValue * value, GParamSpec * pspec);
static void gst_decklink_audio_src_finalize (GObject * object);
static GstStateChangeReturn
gst_decklink_audio_src_change_state (GstElement * element,
GstStateChange transition);
static gboolean gst_decklink_audio_src_set_caps (GstBaseSrc * bsrc,
GstCaps * caps);
static GstCaps *gst_decklink_audio_src_get_caps (GstBaseSrc * bsrc,
GstCaps * filter);
static gboolean gst_decklink_audio_src_unlock (GstBaseSrc * bsrc);
static gboolean gst_decklink_audio_src_unlock_stop (GstBaseSrc * bsrc);
static gboolean gst_decklink_audio_src_query (GstBaseSrc * bsrc,
GstQuery * query);
static GstFlowReturn gst_decklink_audio_src_create (GstPushSrc * psrc,
GstBuffer ** buffer);
static gboolean gst_decklink_audio_src_open (GstDecklinkAudioSrc * self);
static gboolean gst_decklink_audio_src_close (GstDecklinkAudioSrc * self);
static gboolean gst_decklink_audio_src_stop (GstDecklinkAudioSrc * self);
#define parent_class gst_decklink_audio_src_parent_class
G_DEFINE_TYPE (GstDecklinkAudioSrc, gst_decklink_audio_src, GST_TYPE_PUSH_SRC);
static void
gst_decklink_audio_src_class_init (GstDecklinkAudioSrcClass * klass)
{
GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
GstBaseSrcClass *basesrc_class = GST_BASE_SRC_CLASS (klass);
GstPushSrcClass *pushsrc_class = GST_PUSH_SRC_CLASS (klass);
gobject_class->set_property = gst_decklink_audio_src_set_property;
gobject_class->get_property = gst_decklink_audio_src_get_property;
gobject_class->finalize = gst_decklink_audio_src_finalize;
element_class->change_state =
GST_DEBUG_FUNCPTR (gst_decklink_audio_src_change_state);
basesrc_class->get_caps = GST_DEBUG_FUNCPTR (gst_decklink_audio_src_get_caps);
basesrc_class->set_caps = GST_DEBUG_FUNCPTR (gst_decklink_audio_src_set_caps);
basesrc_class->query = GST_DEBUG_FUNCPTR (gst_decklink_audio_src_query);
basesrc_class->unlock = GST_DEBUG_FUNCPTR (gst_decklink_audio_src_unlock);
basesrc_class->unlock_stop =
GST_DEBUG_FUNCPTR (gst_decklink_audio_src_unlock_stop);
pushsrc_class->create = GST_DEBUG_FUNCPTR (gst_decklink_audio_src_create);
g_object_class_install_property (gobject_class, PROP_CONNECTION,
g_param_spec_enum ("connection", "Connection",
"Audio input connection to use",
GST_TYPE_DECKLINK_AUDIO_CONNECTION, DEFAULT_CONNECTION,
(GParamFlags) (G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS |
G_PARAM_CONSTRUCT)));
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)));
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)));
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)));
g_object_class_install_property (gobject_class, PROP_BUFFER_SIZE,
g_param_spec_uint ("buffer-size", "Buffer Size",
"Size of internal buffer in number of video frames", 1,
G_MAXINT, DEFAULT_BUFFER_SIZE,
(GParamFlags) (G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)));
g_object_class_install_property (gobject_class, PROP_CHANNELS,
g_param_spec_enum ("channels", "Channels",
"Audio channels",
GST_TYPE_DECKLINK_AUDIO_CHANNELS, DEFAULT_CHANNELS,
(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)));
gst_element_class_add_static_pad_template (element_class, &sink_template);
gst_element_class_set_static_metadata (element_class, "Decklink Audio Source",
"Audio/Src", "Decklink Source", "David Schleef <ds@entropywave.com>, "
"Sebastian Dröge <sebastian@centricular.com>");
GST_DEBUG_CATEGORY_INIT (gst_decklink_audio_src_debug, "decklinkaudiosrc",
0, "debug category for decklinkaudiosrc element");
}
static void
gst_decklink_audio_src_init (GstDecklinkAudioSrc * self)
{
self->device_number = 0;
self->alignment_threshold = DEFAULT_ALIGNMENT_THRESHOLD;
self->discont_wait = DEFAULT_DISCONT_WAIT;
self->buffer_size = DEFAULT_BUFFER_SIZE;
self->channels = DEFAULT_CHANNELS;
gst_base_src_set_live (GST_BASE_SRC (self), TRUE);
gst_base_src_set_format (GST_BASE_SRC (self), GST_FORMAT_TIME);
g_mutex_init (&self->lock);
g_cond_init (&self->cond);
self->current_packets =
gst_queue_array_new_for_struct (sizeof (CapturePacket),
DEFAULT_BUFFER_SIZE);
}
void
gst_decklink_audio_src_set_property (GObject * object, guint property_id,
const GValue * value, GParamSpec * pspec)
{
GstDecklinkAudioSrc *self = GST_DECKLINK_AUDIO_SRC_CAST (object);
switch (property_id) {
case PROP_CONNECTION:
self->connection =
(GstDecklinkAudioConnectionEnum) g_value_get_enum (value);
break;
case PROP_DEVICE_NUMBER:
self->device_number = g_value_get_int (value);
break;
case PROP_ALIGNMENT_THRESHOLD:
self->alignment_threshold = g_value_get_uint64 (value);
break;
case PROP_DISCONT_WAIT:
self->discont_wait = g_value_get_uint64 (value);
break;
case PROP_BUFFER_SIZE:
self->buffer_size = g_value_get_uint (value);
break;
case PROP_CHANNELS:
self->channels = (GstDecklinkAudioChannelsEnum) g_value_get_enum (value);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, property_id, pspec);
break;
}
}
void
gst_decklink_audio_src_get_property (GObject * object, guint property_id,
GValue * value, GParamSpec * pspec)
{
GstDecklinkAudioSrc *self = GST_DECKLINK_AUDIO_SRC_CAST (object);
switch (property_id) {
case PROP_CONNECTION:
g_value_set_enum (value, self->connection);
break;
case PROP_DEVICE_NUMBER:
g_value_set_int (value, self->device_number);
break;
case PROP_ALIGNMENT_THRESHOLD:
g_value_set_uint64 (value, self->alignment_threshold);
break;
case PROP_DISCONT_WAIT:
g_value_set_uint64 (value, self->discont_wait);
break;
case PROP_BUFFER_SIZE:
g_value_set_uint (value, self->buffer_size);
break;
case PROP_CHANNELS:
g_value_set_enum (value, self->channels);
break;
case PROP_HW_SERIAL_NUMBER:
if (self->input)
g_value_set_string (value, self->input->hw_serial_number);
else
g_value_set_string (value, NULL);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, property_id, pspec);
break;
}
}
void
gst_decklink_audio_src_finalize (GObject * object)
{
GstDecklinkAudioSrc *self = GST_DECKLINK_AUDIO_SRC_CAST (object);
g_mutex_clear (&self->lock);
g_cond_clear (&self->cond);
if (self->current_packets) {
while (gst_queue_array_get_length (self->current_packets) > 0) {
CapturePacket *tmp = (CapturePacket *)
gst_queue_array_pop_head_struct (self->current_packets);
capture_packet_clear (tmp);
}
gst_queue_array_free (self->current_packets);
self->current_packets = NULL;
}
G_OBJECT_CLASS (parent_class)->finalize (object);
}
static gboolean
gst_decklink_audio_src_set_caps (GstBaseSrc * bsrc, GstCaps * caps)
{
GstDecklinkAudioSrc *self = GST_DECKLINK_AUDIO_SRC_CAST (bsrc);
BMDAudioSampleType sample_depth;
GstCaps *current_caps;
HRESULT ret;
BMDAudioConnection conn = (BMDAudioConnection) - 1;
GST_DEBUG_OBJECT (self, "Setting caps %" GST_PTR_FORMAT, caps);
if ((current_caps = gst_pad_get_current_caps (GST_BASE_SRC_PAD (bsrc)))) {
GstCaps *curcaps_cp;
GstStructure *cur_st, *caps_st;
GST_DEBUG_OBJECT (self, "Pad already has caps %" GST_PTR_FORMAT, caps);
curcaps_cp = gst_caps_make_writable (current_caps);
cur_st = gst_caps_get_structure (curcaps_cp, 0);
caps_st = gst_caps_get_structure (caps, 0);
gst_structure_remove_field (cur_st, "channel-mask");
if (!gst_structure_can_intersect (caps_st, cur_st)) {
GST_ERROR_OBJECT (self, "New caps are not compatible with old caps");
gst_caps_unref (current_caps);
gst_caps_unref (curcaps_cp);
return FALSE;
} else {
gst_caps_unref (current_caps);
gst_caps_unref (curcaps_cp);
return TRUE;
}
}
if (!gst_audio_info_from_caps (&self->info, caps))
return FALSE;
if (self->info.finfo->format == GST_AUDIO_FORMAT_S16LE) {
sample_depth = bmdAudioSampleType16bitInteger;
} else {
sample_depth = bmdAudioSampleType32bitInteger;
}
switch (self->connection) {
case GST_DECKLINK_AUDIO_CONNECTION_AUTO:{
GstElement *videosrc = NULL;
GstDecklinkConnectionEnum vconn;
// Try to get the connection from the videosrc and try
// to select a sensible audio connection based on that
g_mutex_lock (&self->input->lock);
if (self->input->videosrc)
videosrc = GST_ELEMENT_CAST (gst_object_ref (self->input->videosrc));
g_mutex_unlock (&self->input->lock);
if (videosrc) {
g_object_get (videosrc, "connection", &vconn, NULL);
gst_object_unref (videosrc);
switch (vconn) {
case GST_DECKLINK_CONNECTION_SDI:
conn = bmdAudioConnectionEmbedded;
break;
case GST_DECKLINK_CONNECTION_HDMI:
conn = bmdAudioConnectionEmbedded;
break;
case GST_DECKLINK_CONNECTION_OPTICAL_SDI:
conn = bmdAudioConnectionEmbedded;
break;
case GST_DECKLINK_CONNECTION_COMPONENT:
conn = bmdAudioConnectionAnalog;
break;
case GST_DECKLINK_CONNECTION_COMPOSITE:
conn = bmdAudioConnectionAnalog;
break;
case GST_DECKLINK_CONNECTION_SVIDEO:
conn = bmdAudioConnectionAnalog;
break;
default:
// Use default
break;
}
}
break;
}
case GST_DECKLINK_AUDIO_CONNECTION_EMBEDDED:
conn = bmdAudioConnectionEmbedded;
break;
case GST_DECKLINK_AUDIO_CONNECTION_AES_EBU:
conn = bmdAudioConnectionAESEBU;
break;
case GST_DECKLINK_AUDIO_CONNECTION_ANALOG:
conn = bmdAudioConnectionAnalog;
break;
case GST_DECKLINK_AUDIO_CONNECTION_ANALOG_XLR:
conn = bmdAudioConnectionAnalogXLR;
break;
case GST_DECKLINK_AUDIO_CONNECTION_ANALOG_RCA:
conn = bmdAudioConnectionAnalogRCA;
break;
default:
g_assert_not_reached ();
break;
}
if (conn != (BMDAudioConnection) - 1) {
ret =
self->input->config->SetInt (bmdDeckLinkConfigAudioInputConnection,
conn);
if (ret != S_OK) {
GST_ERROR ("set configuration (audio input connection): 0x%08lx",
(unsigned long) ret);
return FALSE;
}
}
ret = self->input->input->EnableAudioInput (bmdAudioSampleRate48kHz,
sample_depth, self->info.channels);
if (ret != S_OK) {
GST_WARNING_OBJECT (self, "Failed to enable audio input: 0x%08lx",
(unsigned long) ret);
return FALSE;
}
g_mutex_lock (&self->input->lock);
self->input->audio_enabled = TRUE;
if (self->input->start_streams && self->input->videosrc)
self->input->start_streams (self->input->videosrc);
g_mutex_unlock (&self->input->lock);
return TRUE;
}
static GstCaps *
gst_decklink_audio_src_get_caps (GstBaseSrc * bsrc, GstCaps * filter)
{
GstDecklinkAudioSrc *self = GST_DECKLINK_AUDIO_SRC_CAST (bsrc);
GstCaps *caps;
// We don't support renegotiation
caps = gst_pad_get_current_caps (GST_BASE_SRC_PAD (bsrc));
if (!caps) {
GstCaps *channel_filter, *templ;
templ = gst_pad_get_pad_template_caps (GST_BASE_SRC_PAD (bsrc));
if (self->channels_found > 0) {
channel_filter =
gst_caps_new_simple ("audio/x-raw", "channels", G_TYPE_INT,
self->channels_found, NULL);
} else if (self->channels > 0) {
channel_filter =
gst_caps_new_simple ("audio/x-raw", "channels", G_TYPE_INT,
self->channels, NULL);
} else {
channel_filter = gst_caps_new_empty_simple ("audio/x-raw");
}
caps = gst_caps_intersect (channel_filter, templ);
gst_caps_unref (channel_filter);
gst_caps_unref (templ);
}
if (filter) {
GstCaps *tmp =
gst_caps_intersect_full (filter, caps, GST_CAPS_INTERSECT_FIRST);
gst_caps_unref (caps);
caps = tmp;
}
return caps;
}
static void
gst_decklink_audio_src_got_packet (GstElement * element,
IDeckLinkAudioInputPacket * packet, GstClockTime capture_time,
GstClockTime stream_time, GstClockTime stream_duration,
GstClockTime hardware_time, GstClockTime hardware_duration,
gboolean no_signal)
{
GstDecklinkAudioSrc *self = GST_DECKLINK_AUDIO_SRC_CAST (element);
GstClockTime timestamp;
GST_LOG_OBJECT (self,
"Got audio packet at %" GST_TIME_FORMAT " / %" GST_TIME_FORMAT
", no signal %d", GST_TIME_ARGS (capture_time),
GST_TIME_ARGS (stream_time), no_signal);
g_mutex_lock (&self->input->lock);
if (self->input->videosrc) {
GstDecklinkVideoSrc *videosrc =
GST_DECKLINK_VIDEO_SRC_CAST (gst_object_ref (self->input->videosrc));
if (videosrc->drop_no_signal_frames && no_signal) {
g_mutex_unlock (&self->input->lock);
return;
}
if (videosrc->first_time == GST_CLOCK_TIME_NONE)
videosrc->first_time = stream_time;
if (videosrc->skip_first_time > 0
&& stream_time - videosrc->first_time < videosrc->skip_first_time) {
GST_DEBUG_OBJECT (self,
"Skipping frame as requested: %" GST_TIME_FORMAT " < %"
GST_TIME_FORMAT, GST_TIME_ARGS (stream_time),
GST_TIME_ARGS (videosrc->skip_first_time + videosrc->first_time));
g_mutex_unlock (&self->input->lock);
return;
}
if (videosrc->output_stream_time)
timestamp = stream_time;
else
timestamp = gst_clock_adjust_with_calibration (NULL, stream_time,
videosrc->current_time_mapping.xbase,
videosrc->current_time_mapping.b, videosrc->current_time_mapping.num,
videosrc->current_time_mapping.den);
} else {
timestamp = capture_time;
}
g_mutex_unlock (&self->input->lock);
GST_LOG_OBJECT (self, "Converted times to %" GST_TIME_FORMAT,
GST_TIME_ARGS (timestamp));
g_mutex_lock (&self->lock);
if (!self->flushing) {
CapturePacket p;
guint skipped_packets = 0;
GstClockTime from_timestamp = GST_CLOCK_TIME_NONE;
GstClockTime to_timestamp = GST_CLOCK_TIME_NONE;
while (gst_queue_array_get_length (self->current_packets) >=
self->buffer_size) {
CapturePacket *tmp = (CapturePacket *)
gst_queue_array_pop_head_struct (self->current_packets);
if (skipped_packets == 0)
from_timestamp = tmp->timestamp;
skipped_packets++;
to_timestamp = tmp->timestamp;
capture_packet_clear (tmp);
}
if (skipped_packets > 0)
GST_WARNING_OBJECT (self,
"Dropped %u old packets from %" GST_TIME_FORMAT " to %"
GST_TIME_FORMAT, skipped_packets, GST_TIME_ARGS (from_timestamp),
GST_TIME_ARGS (to_timestamp));
memset (&p, 0, sizeof (p));
p.packet = packet;
p.timestamp = timestamp;
p.stream_timestamp = stream_time;
p.stream_duration = stream_duration;
p.hardware_timestamp = hardware_time;
p.hardware_duration = hardware_duration;
p.no_signal = no_signal;
packet->AddRef ();
gst_queue_array_push_tail_struct (self->current_packets, &p);
g_cond_signal (&self->cond);
}
g_mutex_unlock (&self->lock);
}
static GstFlowReturn
gst_decklink_audio_src_create (GstPushSrc * bsrc, GstBuffer ** buffer)
{
GstDecklinkAudioSrc *self = GST_DECKLINK_AUDIO_SRC_CAST (bsrc);
GstFlowReturn flow_ret = GST_FLOW_OK;
const guint8 *data;
glong sample_count;
gsize data_size;
CapturePacket p;
AudioPacket *ap;
GstClockTime timestamp, duration;
GstClockTime start_time, end_time;
guint64 start_offset, end_offset;
gboolean discont = FALSE;
static GstStaticCaps stream_reference =
GST_STATIC_CAPS ("timestamp/x-decklink-stream");
static GstStaticCaps hardware_reference =
GST_STATIC_CAPS ("timestamp/x-decklink-hardware");
retry:
g_mutex_lock (&self->lock);
while (gst_queue_array_is_empty (self->current_packets) && !self->flushing) {
g_cond_wait (&self->cond, &self->lock);
}
if (self->flushing) {
GST_DEBUG_OBJECT (self, "Flushing");
g_mutex_unlock (&self->lock);
return GST_FLOW_FLUSHING;
}
p = *(CapturePacket *)
gst_queue_array_pop_head_struct (self->current_packets);
g_mutex_unlock (&self->lock);
p.packet->GetBytes ((gpointer *) & data);
sample_count = p.packet->GetSampleFrameCount ();
data_size = self->info.bpf * sample_count;
if (p.timestamp == GST_CLOCK_TIME_NONE && self->next_offset == (guint64) - 1) {
GST_DEBUG_OBJECT (self,
"Got packet without timestamp before initial "
"timestamp after discont - dropping");
capture_packet_clear (&p);
goto retry;
}
ap = (AudioPacket *) g_malloc0 (sizeof (AudioPacket));
*buffer =
gst_buffer_new_wrapped_full ((GstMemoryFlags) GST_MEMORY_FLAG_READONLY,
(gpointer) data, data_size, 0, data_size, ap,
(GDestroyNotify) audio_packet_free);
ap->packet = p.packet;
p.packet->AddRef ();
ap->input = self->input->input;
ap->input->AddRef ();
timestamp = p.timestamp;
// Jitter and discontinuity handling, based on audiobasesrc
start_time = timestamp;
// Convert to the sample numbers
start_offset =
gst_util_uint64_scale (start_time, self->info.rate, GST_SECOND);
end_offset = start_offset + sample_count;
end_time = gst_util_uint64_scale_int (end_offset, GST_SECOND,
self->info.rate);
duration = end_time - start_time;
if (self->next_offset == (guint64) - 1) {
discont = TRUE;
} else {
guint64 diff, max_sample_diff;
// Check discont
if (start_offset <= self->next_offset)
diff = self->next_offset - start_offset;
else
diff = start_offset - self->next_offset;
max_sample_diff =
gst_util_uint64_scale_int (self->alignment_threshold, self->info.rate,
GST_SECOND);
// Discont!
if (G_UNLIKELY (diff >= max_sample_diff)) {
if (self->discont_wait > 0) {
if (self->discont_time == GST_CLOCK_TIME_NONE) {
self->discont_time = start_time;
} else if (start_time - self->discont_time >= self->discont_wait) {
discont = TRUE;
self->discont_time = GST_CLOCK_TIME_NONE;
}
} else {
discont = TRUE;
}
} else if (G_UNLIKELY (self->discont_time != GST_CLOCK_TIME_NONE)) {
// we have had a discont, but are now back on track!
self->discont_time = GST_CLOCK_TIME_NONE;
}
}
if (discont) {
// Have discont, need resync and use the capture timestamps
if (self->next_offset != (guint64) - 1)
GST_INFO_OBJECT (self, "Have discont. Expected %"
G_GUINT64_FORMAT ", got %" G_GUINT64_FORMAT,
self->next_offset, start_offset);
GST_BUFFER_FLAG_SET (*buffer, GST_BUFFER_FLAG_DISCONT);
self->next_offset = end_offset;
// Got a discont and adjusted, reset the discont_time marker.
self->discont_time = GST_CLOCK_TIME_NONE;
} else {
// No discont, just keep counting
timestamp =
gst_util_uint64_scale (self->next_offset, GST_SECOND, self->info.rate);
self->next_offset += sample_count;
duration =
gst_util_uint64_scale (self->next_offset, GST_SECOND,
self->info.rate) - timestamp;
}
// Detect gaps in stream time
self->processed += sample_count;
if (self->expected_stream_time != GST_CLOCK_TIME_NONE
&& p.stream_timestamp == GST_CLOCK_TIME_NONE) {
/* We missed a frame. Extrapolate the timestamps */
p.stream_timestamp = self->expected_stream_time;
p.stream_duration =
gst_util_uint64_scale_int (sample_count, GST_SECOND, self->info.rate);
}
if (self->last_hardware_time != GST_CLOCK_TIME_NONE
&& p.hardware_timestamp == GST_CLOCK_TIME_NONE) {
/* This should always happen when the previous one also does, but let's
* have two separate checks just in case */
GstClockTime start_hw_offset, end_hw_offset;
start_hw_offset =
gst_util_uint64_scale (self->last_hardware_time, self->info.rate,
GST_SECOND);
end_hw_offset = start_hw_offset + sample_count;
p.hardware_timestamp =
gst_util_uint64_scale_int (end_hw_offset, GST_SECOND, self->info.rate);
/* Will be the same as the stream duration - reuse it */
p.hardware_duration = p.stream_duration;
}
if (p.stream_timestamp != GST_CLOCK_TIME_NONE) {
GstClockTime start_stream_time, end_stream_time;
start_stream_time = p.stream_timestamp;
start_offset =
gst_util_uint64_scale (start_stream_time, self->info.rate, GST_SECOND);
end_offset = start_offset + sample_count;
end_stream_time = gst_util_uint64_scale_int (end_offset, GST_SECOND,
self->info.rate);
/* With drop-frame we have gaps of 1 sample every now and then (rounding
* errors because of the samples-per-frame pattern which is not 100%
* accurate), and due to rounding errors in the calculations these can be
* 2>x>1 */
if (self->expected_stream_time != GST_CLOCK_TIME_NONE &&
ABSDIFF (self->expected_stream_time, p.stream_timestamp) >
gst_util_uint64_scale (2, GST_SECOND, self->info.rate)) {
GstMessage *msg;
GstClockTime running_time;
self->dropped +=
gst_util_uint64_scale (ABSDIFF (self->expected_stream_time,
p.stream_timestamp), self->info.rate, GST_SECOND);
running_time =
gst_segment_to_running_time (&GST_BASE_SRC (self)->segment,
GST_FORMAT_TIME, timestamp);
msg =
gst_message_new_qos (GST_OBJECT (self), TRUE, running_time,
p.stream_timestamp, timestamp, duration);
gst_message_set_qos_stats (msg, GST_FORMAT_DEFAULT, self->processed,
self->dropped);
gst_element_post_message (GST_ELEMENT (self), msg);
}
self->expected_stream_time = end_stream_time;
}
self->last_hardware_time = p.hardware_timestamp;
if (p.no_signal)
GST_BUFFER_FLAG_SET (*buffer, GST_BUFFER_FLAG_GAP);
GST_BUFFER_TIMESTAMP (*buffer) = timestamp;
GST_BUFFER_DURATION (*buffer) = duration;
gst_buffer_add_reference_timestamp_meta (*buffer,
gst_static_caps_get (&stream_reference), p.stream_timestamp,
p.stream_duration);
gst_buffer_add_reference_timestamp_meta (*buffer,
gst_static_caps_get (&hardware_reference), p.hardware_timestamp,
p.hardware_duration);
GST_DEBUG_OBJECT (self,
"Outputting buffer %p with timestamp %" GST_TIME_FORMAT " and duration %"
GST_TIME_FORMAT, *buffer, GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (*buffer)),
GST_TIME_ARGS (GST_BUFFER_DURATION (*buffer)));
capture_packet_clear (&p);
return flow_ret;
}
static gboolean
gst_decklink_audio_src_query (GstBaseSrc * bsrc, GstQuery * query)
{
GstDecklinkAudioSrc *self = GST_DECKLINK_AUDIO_SRC_CAST (bsrc);
gboolean ret = TRUE;
switch (GST_QUERY_TYPE (query)) {
case GST_QUERY_LATENCY:{
if (self->input) {
g_mutex_lock (&self->input->lock);
if (self->input->mode) {
GstClockTime min, max;
min =
gst_util_uint64_scale_ceil (GST_SECOND, self->input->mode->fps_d,
self->input->mode->fps_n);
max = self->buffer_size * min;
gst_query_set_latency (query, TRUE, min, max);
ret = TRUE;
} else {
ret = FALSE;
}
g_mutex_unlock (&self->input->lock);
} else {
ret = FALSE;
}
break;
}
default:
ret = GST_BASE_SRC_CLASS (parent_class)->query (bsrc, query);
break;
}
return ret;
}
static gboolean
gst_decklink_audio_src_unlock (GstBaseSrc * bsrc)
{
GstDecklinkAudioSrc *self = GST_DECKLINK_AUDIO_SRC_CAST (bsrc);
g_mutex_lock (&self->lock);
self->flushing = TRUE;
g_cond_signal (&self->cond);
g_mutex_unlock (&self->lock);
return TRUE;
}
static gboolean
gst_decklink_audio_src_unlock_stop (GstBaseSrc * bsrc)
{
GstDecklinkAudioSrc *self = GST_DECKLINK_AUDIO_SRC_CAST (bsrc);
g_mutex_lock (&self->lock);
self->flushing = FALSE;
while (gst_queue_array_get_length (self->current_packets) > 0) {
CapturePacket *tmp = (CapturePacket *)
gst_queue_array_pop_head_struct (self->current_packets);
capture_packet_clear (tmp);
}
g_mutex_unlock (&self->lock);
return TRUE;
}
static gboolean
gst_decklink_audio_src_open (GstDecklinkAudioSrc * self)
{
GST_DEBUG_OBJECT (self, "Opening");
self->input =
gst_decklink_acquire_nth_input (self->device_number,
GST_ELEMENT_CAST (self), TRUE);
if (!self->input) {
GST_ERROR_OBJECT (self, "Failed to acquire input");
return FALSE;
}
g_object_notify (G_OBJECT (self), "hw-serial-number");
g_mutex_lock (&self->input->lock);
if (self->channels > 0) {
self->channels_found = self->channels;
} else {
if (self->input->attributes) {
int64_t channels_found;
HRESULT ret = self->input->attributes->GetInt
(BMDDeckLinkMaximumAudioChannels, &channels_found);
self->channels_found = channels_found;
/* Sometimes the card may report an invalid number of channels. In
* that case, we should (empirically) use 8. */
if (ret != S_OK ||
self->channels_found == 0 || g_enum_get_value ((GEnumClass *)
g_type_class_peek (GST_TYPE_DECKLINK_AUDIO_CHANNELS),
self->channels_found)
== NULL) {
self->channels_found = GST_DECKLINK_AUDIO_CHANNELS_8;
}
}
}
self->input->got_audio_packet = gst_decklink_audio_src_got_packet;
g_mutex_unlock (&self->input->lock);
return TRUE;
}
static gboolean
gst_decklink_audio_src_close (GstDecklinkAudioSrc * self)
{
GST_DEBUG_OBJECT (self, "Closing");
if (self->input) {
g_mutex_lock (&self->input->lock);
self->input->got_audio_packet = NULL;
g_mutex_unlock (&self->input->lock);
gst_decklink_release_nth_input (self->device_number,
GST_ELEMENT_CAST (self), TRUE);
self->input = NULL;
}
return TRUE;
}
static gboolean
gst_decklink_audio_src_stop (GstDecklinkAudioSrc * self)
{
GST_DEBUG_OBJECT (self, "Stopping");
while (gst_queue_array_get_length (self->current_packets) > 0) {
CapturePacket *tmp = (CapturePacket *)
gst_queue_array_pop_head_struct (self->current_packets);
capture_packet_clear (tmp);
}
if (self->input && self->input->audio_enabled) {
g_mutex_lock (&self->input->lock);
self->input->audio_enabled = FALSE;
g_mutex_unlock (&self->input->lock);
self->input->input->DisableAudioInput ();
}
return TRUE;
}
#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 GstStateChangeReturn
gst_decklink_audio_src_change_state (GstElement * element,
GstStateChange transition)
{
GstDecklinkAudioSrc *self = GST_DECKLINK_AUDIO_SRC_CAST (element);
GstStateChangeReturn ret;
switch (transition) {
case GST_STATE_CHANGE_NULL_TO_READY:
self->processed = 0;
self->dropped = 0;
self->expected_stream_time = GST_CLOCK_TIME_NONE;
if (!gst_decklink_audio_src_open (self)) {
ret = GST_STATE_CHANGE_FAILURE;
goto out;
}
break;
case GST_STATE_CHANGE_READY_TO_PAUSED:{
GstElement *videosrc = NULL;
// Check if there is a video src for this input too and if it
// is actually in the same pipeline
g_mutex_lock (&self->input->lock);
if (self->input->videosrc)
videosrc = GST_ELEMENT_CAST (gst_object_ref (self->input->videosrc));
g_mutex_unlock (&self->input->lock);
if (!videosrc) {
GST_ELEMENT_ERROR (self, STREAM, FAILED,
(NULL), ("Audio src needs a video src for its operation"));
ret = GST_STATE_CHANGE_FAILURE;
goto out;
}
// FIXME: This causes deadlocks sometimes
#if 0
else if (!in_same_pipeline (GST_ELEMENT_CAST (self), videosrc)) {
GST_ELEMENT_ERROR (self, STREAM, FAILED,
(NULL),
("Audio src and video src need to be in the same pipeline"));
ret = GST_STATE_CHANGE_FAILURE;
gst_object_unref (videosrc);
goto out;
}
#endif
if (videosrc)
gst_object_unref (videosrc);
self->flushing = FALSE;
self->next_offset = -1;
break;
}
default:
break;
}
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
if (ret == GST_STATE_CHANGE_FAILURE)
return ret;
switch (transition) {
case GST_STATE_CHANGE_PAUSED_TO_READY:
gst_decklink_audio_src_stop (self);
break;
case GST_STATE_CHANGE_READY_TO_NULL:
gst_decklink_audio_src_close (self);
break;
default:
break;
}
out:
return ret;
}
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