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gstreamer/sys/decklink/gstdecklinkaudiosrc.cpp
Seungha Yang ba83f29cec decklinkaudiosrc: Don't assume that stream time is always valid 
As per SDK doc, IDeckLinkInputCallback::VideoInputFrameArrived
method might not provide video frame and it can be null.
In that case, given stream_time can be invalid.
So, we should not try to convert GST_CLOCK_TIME_NONE
by using gst_clock_adjust_with_calibration()

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2337>
2021-06-21 09:58:46 +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-decklinkaudiosrc
* @short_description: Inputs Audio from a BlackMagic DeckLink Device
* @see_also: decklinkvideosrc
*
* Capture Video and Audio from a BlackMagic DeckLink Device. Can only be used
* in conjunction with decklinkvideosink.
*
* ## Sample pipeline
* |[
* gst-launch-1.0 \
* decklinkvideosrc device-number=0 mode=1080p25 ! autovideosink \
* decklinkaudiosrc device-number=0 ! autoaudiosink
* ]|
* Capturing 1080p25 video and audio from the SDI-In of Card 0. Devices are numbered
* starting with 0.
*/
#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_unlock (GstBaseSrc * bsrc);
static gboolean gst_decklink_audio_src_unlock_stop (GstBaseSrc * bsrc);
static GstCaps *gst_decklink_audio_src_get_caps (GstBaseSrc * bsrc,
GstCaps * filter);
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);
GST_ELEMENT_REGISTER_DEFINE_WITH_CODE (decklinkaudiosrc, "decklinkaudiosrc", GST_RANK_NONE,
GST_TYPE_DECKLINK_AUDIO_SRC, decklink_element_init (plugin));
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->query = GST_DEBUG_FUNCPTR (gst_decklink_audio_src_query);
basesrc_class->negotiate = NULL;
basesrc_class->get_caps = GST_DEBUG_FUNCPTR (gst_decklink_audio_src_get_caps);
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/Source/Hardware", "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);
gst_pad_use_fixed_caps (GST_BASE_SRC_PAD (self));
g_mutex_init (&self->lock);
g_cond_init (&self->cond);
self->current_packets =
gst_queue_array_new_for_struct (sizeof (CapturePacket),
DEFAULT_BUFFER_SIZE);
self->skipped_last = 0;
self->skip_from_timestamp = GST_CLOCK_TIME_NONE;
self->skip_to_timestamp = GST_CLOCK_TIME_NONE;
}
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_start (GstDecklinkAudioSrc * self)
{
BMDAudioSampleType sample_depth;
HRESULT ret;
BMDAudioConnection conn = (BMDAudioConnection) - 1;
GstCaps *allowed_caps, *caps;
g_mutex_lock (&self->input->lock);
if (self->input->audio_enabled) {
g_mutex_unlock (&self->input->lock);
return TRUE;
}
g_mutex_unlock (&self->input->lock);
/* Negotiate the format / sample depth with downstream */
allowed_caps = gst_pad_get_allowed_caps (GST_BASE_SRC_PAD (self));
if (!allowed_caps)
allowed_caps = gst_pad_get_pad_template_caps (GST_BASE_SRC_PAD (self));
sample_depth = bmdAudioSampleType32bitInteger;
if (!gst_caps_is_empty (allowed_caps)) {
GstStructure *s;
allowed_caps = gst_caps_simplify (allowed_caps);
s = gst_caps_get_structure (allowed_caps, 0);
/* If it's not a string then both formats are supported */
if (gst_structure_has_field_typed (s, "format", G_TYPE_STRING)) {
const gchar *format = gst_structure_get_string (s, "format");
if (g_str_equal (format, "S16LE")) {
sample_depth = bmdAudioSampleType16bitInteger;
}
}
}
gst_caps_unref (allowed_caps);
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->channels_found);
if (ret != S_OK) {
GST_WARNING_OBJECT (self, "Failed to enable audio input: 0x%08lx",
(unsigned long) ret);
return FALSE;
}
gst_audio_info_set_format (&self->info,
sample_depth ==
bmdAudioSampleType16bitInteger ? GST_AUDIO_FORMAT_S16LE :
GST_AUDIO_FORMAT_S32LE, 48000, self->channels_found, NULL);
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);
caps = gst_audio_info_to_caps (&self->info);
if (!gst_base_src_set_caps (GST_BASE_SRC (self), caps)) {
gst_caps_unref (caps);
GST_WARNING_OBJECT (self, "Failed to set caps");
return FALSE;
}
gst_caps_unref (caps);
self->skipped_last = 0;
self->skip_from_timestamp = GST_CLOCK_TIME_NONE;
self->skip_to_timestamp = GST_CLOCK_TIME_NONE;
return TRUE;
}
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_CLOCK_TIME_NONE;
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 (GST_CLOCK_TIME_IS_VALID (videosrc->first_time) &&
GST_CLOCK_TIME_IS_VALID (stream_time) &&
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 if (GST_CLOCK_TIME_IS_VALID (stream_time))
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;
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 && self->skipped_last == 0)
self->skip_from_timestamp = tmp->timestamp;
skipped_packets++;
self->skip_to_timestamp = tmp->timestamp;
capture_packet_clear (tmp);
}
if (self->skipped_last == 0 && skipped_packets > 0) {
GST_WARNING_OBJECT (self, "Starting to drop audio packets");
}
if (skipped_packets == 0 && self->skipped_last > 0) {
GST_ELEMENT_WARNING_WITH_DETAILS (self,
STREAM, FAILED,
("Dropped %u old packets from %" GST_TIME_FORMAT " to %"
GST_TIME_FORMAT, self->skipped_last,
GST_TIME_ARGS (self->skip_from_timestamp),
GST_TIME_ARGS (self->skip_to_timestamp)),
(NULL),
("dropped", G_TYPE_UINT, self->skipped_last,
"from", G_TYPE_UINT64, self->skip_from_timestamp,
"to", G_TYPE_UINT64, self->skip_to_timestamp, NULL));
self->skipped_last = 0;
}
self->skipped_last += skipped_packets;
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");
if (!gst_decklink_audio_src_start (self)) {
return GST_FLOW_NOT_NEGOTIATED;
}
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;
timestamp = p.timestamp;
if (!GST_CLOCK_TIME_IS_VALID (timestamp)) {
if (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;
} else {
GST_INFO_OBJECT (self, "Unknown timestamp value");
/* Likely the case where IDeckLinkInputCallback::VideoInputFrameArrived()
* didn't provide video frame, so no reference video stream timestamp
* is available. It can happen as per SDK documentation
* under the following circumstances:
* - On Intensity Pro with progressive NTSC only, every video frame will
* have two audio packets.
* - With 3:2 pulldown there are five audio packets for each four
* video frames.
* - If video processing is not fast enough, audio will still be delivered
*
* Assume there was no packet drop from previous valid packet, and use
* previously calculated expected timestamp here */
timestamp = gst_util_uint64_scale (self->next_offset,
GST_SECOND, self->info.rate);
}
}
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 ();
// 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);
// Convert back to round down to a sample multiple and get rid of rounding errors
start_time = gst_util_uint64_scale (start_offset, GST_SECOND, self->info.rate);
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 (self->alignment_threshold > 0
&& self->alignment_threshold != GST_CLOCK_TIME_NONE
&& 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 if (self->alignment_threshold == 0) {
// Don't align, just pass through timestamps
} 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 GstCaps *
gst_decklink_audio_src_get_caps (GstBaseSrc * bsrc, GstCaps * filter)
{
GstDecklinkAudioSrc *self = GST_DECKLINK_AUDIO_SRC_CAST (bsrc);
GstCaps *caps, *template_caps;
const GstStructure *s;
gint channels;
channels = self->channels;
if (channels == 0)
channels = self->channels_found;
template_caps = gst_pad_get_pad_template_caps (GST_BASE_SRC_PAD (bsrc));
if (channels == 0) {
caps = template_caps;
} else {
if (channels > 2)
s = gst_caps_get_structure (template_caps, 1);
else
s = gst_caps_get_structure (template_caps, 0);
caps = gst_caps_new_full (gst_structure_copy (s), NULL);
gst_caps_set_simple (caps, "channels", G_TYPE_INT, channels, NULL);
gst_caps_unref (template_caps);
}
if (filter) {
GstCaps *tmp =
gst_caps_intersect_full (filter, caps, GST_CAPS_INTERSECT_FIRST);
gst_caps_unref (caps);
caps = tmp;
}
return caps;
}
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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