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gstreamer/gstajasink.cpp

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/* GStreamer
* Copyright (C) 2021 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 <ajaanc/includes/ancillarydata_cea708.h>
#include <ajaanc/includes/ancillarylist.h>
#include <ajantv2/includes/ntv2rp188.h>
#include "gstajacommon.h"
#include "gstajasink.h"
GST_DEBUG_CATEGORY_STATIC(gst_aja_sink_debug);
#define GST_CAT_DEFAULT gst_aja_sink_debug
#define DEFAULT_DEVICE_IDENTIFIER ("0")
#define DEFAULT_CHANNEL (::NTV2_CHANNEL1)
#define DEFAULT_AUDIO_SYSTEM (GST_AJA_AUDIO_SYSTEM_AUTO)
#define DEFAULT_OUTPUT_DESTINATION (GST_AJA_OUTPUT_DESTINATION_AUTO)
#define DEFAULT_TIMECODE_INDEX (GST_AJA_TIMECODE_INDEX_AUTO)
#define DEFAULT_REFERENCE_SOURCE (GST_AJA_REFERENCE_SOURCE_AUTO)
#define DEFAULT_QUEUE_SIZE (16)
#define DEFAULT_OUTPUT_CPU_CORE (G_MAXUINT)
enum {
PROP_0,
PROP_DEVICE_IDENTIFIER,
PROP_CHANNEL,
PROP_AUDIO_SYSTEM,
PROP_OUTPUT_DESTINATION,
PROP_TIMECODE_INDEX,
PROP_REFERENCE_SOURCE,
PROP_QUEUE_SIZE,
PROP_OUTPUT_CPU_CORE,
};
typedef enum {
QUEUE_ITEM_TYPE_FRAME,
} QueueItemType;
typedef struct {
QueueItemType type;
// For FRAME
GstVideoFrame frame;
GstBuffer *audio_buffer;
GstMapInfo audio_map;
NTV2_RP188 tc;
AJAAncillaryList *anc_packet_list;
} QueueItem;
static void gst_aja_sink_set_property(GObject *object, guint property_id,
const GValue *value, GParamSpec *pspec);
static void gst_aja_sink_get_property(GObject *object, guint property_id,
GValue *value, GParamSpec *pspec);
static void gst_aja_sink_finalize(GObject *object);
static gboolean gst_aja_sink_set_caps(GstBaseSink *bsink, GstCaps *caps);
static GstCaps *gst_aja_sink_get_caps(GstBaseSink *bsink, GstCaps *filter);
static gboolean gst_aja_sink_event(GstBaseSink *bsink, GstEvent *event);
static gboolean gst_aja_sink_propose_allocation(GstBaseSink *bsink,
GstQuery *query);
static GstFlowReturn gst_aja_sink_render(GstBaseSink *bsink, GstBuffer *buffer);
static gboolean gst_aja_sink_open(GstAjaSink *sink);
static gboolean gst_aja_sink_close(GstAjaSink *sink);
static gboolean gst_aja_sink_start(GstAjaSink *sink);
static gboolean gst_aja_sink_stop(GstAjaSink *sink);
static GstStateChangeReturn gst_aja_sink_change_state(
GstElement *element, GstStateChange transition);
static void output_thread_func(AJAThread *thread, void *data);
#define parent_class gst_aja_sink_parent_class
G_DEFINE_TYPE(GstAjaSink, gst_aja_sink, GST_TYPE_BASE_SINK);
static void gst_aja_sink_class_init(GstAjaSinkClass *klass) {
GObjectClass *gobject_class = G_OBJECT_CLASS(klass);
GstElementClass *element_class = GST_ELEMENT_CLASS(klass);
GstBaseSinkClass *basesink_class = GST_BASE_SINK_CLASS(klass);
GstCaps *templ_caps;
gobject_class->set_property = gst_aja_sink_set_property;
gobject_class->get_property = gst_aja_sink_get_property;
gobject_class->finalize = gst_aja_sink_finalize;
g_object_class_install_property(
gobject_class, PROP_DEVICE_IDENTIFIER,
g_param_spec_string(
"device-identifier", "Device identifier",
"Input device instance to use", DEFAULT_DEVICE_IDENTIFIER,
(GParamFlags)(G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS |
G_PARAM_CONSTRUCT)));
g_object_class_install_property(
gobject_class, PROP_CHANNEL,
g_param_spec_uint(
"channel", "Channel", "Channel to use", 0, NTV2_MAX_NUM_CHANNELS - 1,
DEFAULT_CHANNEL,
(GParamFlags)(G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS |
G_PARAM_CONSTRUCT)));
g_object_class_install_property(
gobject_class, PROP_QUEUE_SIZE,
g_param_spec_uint(
"queue-size", "Queue Size",
"Size of internal queue in number of video frames. "
"Half of this is allocated as device buffers and equal to the "
"latency.",
1, G_MAXINT, DEFAULT_QUEUE_SIZE,
(GParamFlags)(G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)));
g_object_class_install_property(
gobject_class, PROP_AUDIO_SYSTEM,
g_param_spec_enum(
"audio-system", "Audio System", "Audio system to use",
GST_TYPE_AJA_AUDIO_SYSTEM, DEFAULT_AUDIO_SYSTEM,
(GParamFlags)(G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS |
G_PARAM_CONSTRUCT)));
g_object_class_install_property(
gobject_class, PROP_OUTPUT_DESTINATION,
g_param_spec_enum(
"output-destination", "Output Destination",
"Output destination to use", GST_TYPE_AJA_OUTPUT_DESTINATION,
DEFAULT_OUTPUT_DESTINATION,
(GParamFlags)(G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS |
G_PARAM_CONSTRUCT)));
g_object_class_install_property(
gobject_class, PROP_TIMECODE_INDEX,
g_param_spec_enum(
"timecode-index", "Timecode Index", "Timecode index to use",
GST_TYPE_AJA_TIMECODE_INDEX, DEFAULT_TIMECODE_INDEX,
(GParamFlags)(G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS |
G_PARAM_CONSTRUCT)));
g_object_class_install_property(
gobject_class, PROP_REFERENCE_SOURCE,
g_param_spec_enum(
"reference-source", "Reference Source", "Reference source to use",
GST_TYPE_AJA_REFERENCE_SOURCE, DEFAULT_REFERENCE_SOURCE,
(GParamFlags)(G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS |
G_PARAM_CONSTRUCT)));
g_object_class_install_property(
gobject_class, PROP_OUTPUT_CPU_CORE,
g_param_spec_uint(
"output-cpu-core", "Output CPU Core",
"Sets the affinity of the output thread to this CPU core "
"(-1=disabled)",
0, G_MAXUINT, DEFAULT_OUTPUT_CPU_CORE,
(GParamFlags)(G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS |
G_PARAM_CONSTRUCT)));
element_class->change_state = GST_DEBUG_FUNCPTR(gst_aja_sink_change_state);
basesink_class->set_caps = GST_DEBUG_FUNCPTR(gst_aja_sink_set_caps);
basesink_class->get_caps = GST_DEBUG_FUNCPTR(gst_aja_sink_get_caps);
basesink_class->event = GST_DEBUG_FUNCPTR(gst_aja_sink_event);
basesink_class->propose_allocation =
GST_DEBUG_FUNCPTR(gst_aja_sink_propose_allocation);
basesink_class->render = GST_DEBUG_FUNCPTR(gst_aja_sink_render);
templ_caps = gst_ntv2_supported_caps(DEVICE_ID_INVALID);
gst_element_class_add_pad_template(
element_class,
gst_pad_template_new("sink", GST_PAD_SINK, GST_PAD_ALWAYS, templ_caps));
gst_caps_unref(templ_caps);
gst_element_class_set_static_metadata(
element_class, "AJA audio/video sink", "Audio/Video/Sink",
"Outputs audio/video frames with AJA devices",
"Sebastian Dröge <sebastian@centricular.com>");
GST_DEBUG_CATEGORY_INIT(gst_aja_sink_debug, "ajasink", 0, "AJA sink");
}
static void gst_aja_sink_init(GstAjaSink *self) {
g_mutex_init(&self->queue_lock);
g_cond_init(&self->queue_cond);
g_cond_init(&self->drain_cond);
self->device_identifier = g_strdup(DEFAULT_DEVICE_IDENTIFIER);
self->channel = DEFAULT_CHANNEL;
self->queue_size = DEFAULT_QUEUE_SIZE;
self->audio_system_setting = DEFAULT_AUDIO_SYSTEM;
self->output_destination = DEFAULT_OUTPUT_DESTINATION;
self->timecode_index = DEFAULT_TIMECODE_INDEX;
self->reference_source = DEFAULT_REFERENCE_SOURCE;
self->output_cpu_core = DEFAULT_OUTPUT_CPU_CORE;
gst_base_sink_set_render_delay(GST_BASE_SINK(self),
(self->queue_size / 2) * GST_SECOND / 30);
self->queue =
gst_queue_array_new_for_struct(sizeof(QueueItem), self->queue_size);
}
void gst_aja_sink_set_property(GObject *object, guint property_id,
const GValue *value, GParamSpec *pspec) {
GstAjaSink *self = GST_AJA_SINK(object);
switch (property_id) {
case PROP_DEVICE_IDENTIFIER:
g_free(self->device_identifier);
self->device_identifier = g_value_dup_string(value);
break;
case PROP_CHANNEL:
self->channel = (NTV2Channel)g_value_get_uint(value);
break;
case PROP_QUEUE_SIZE:
self->queue_size = g_value_get_uint(value);
break;
case PROP_AUDIO_SYSTEM:
self->audio_system_setting = (GstAjaAudioSystem)g_value_get_enum(value);
break;
case PROP_OUTPUT_DESTINATION:
self->output_destination =
(GstAjaOutputDestination)g_value_get_enum(value);
break;
case PROP_TIMECODE_INDEX:
self->timecode_index = (GstAjaTimecodeIndex)g_value_get_enum(value);
break;
case PROP_REFERENCE_SOURCE:
self->reference_source = (GstAjaReferenceSource)g_value_get_enum(value);
break;
case PROP_OUTPUT_CPU_CORE:
self->output_cpu_core = g_value_get_uint(value);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID(object, property_id, pspec);
break;
}
}
void gst_aja_sink_get_property(GObject *object, guint property_id,
GValue *value, GParamSpec *pspec) {
GstAjaSink *self = GST_AJA_SINK(object);
switch (property_id) {
case PROP_DEVICE_IDENTIFIER:
g_value_set_string(value, self->device_identifier);
break;
case PROP_CHANNEL:
g_value_set_uint(value, self->channel);
break;
case PROP_QUEUE_SIZE:
g_value_set_uint(value, self->queue_size);
break;
case PROP_AUDIO_SYSTEM:
g_value_set_enum(value, self->audio_system_setting);
break;
case PROP_OUTPUT_DESTINATION:
g_value_set_enum(value, self->output_destination);
break;
case PROP_TIMECODE_INDEX:
g_value_set_enum(value, self->timecode_index);
break;
case PROP_REFERENCE_SOURCE:
g_value_set_enum(value, self->reference_source);
break;
case PROP_OUTPUT_CPU_CORE:
g_value_set_uint(value, self->output_cpu_core);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID(object, property_id, pspec);
break;
}
}
void gst_aja_sink_finalize(GObject *object) {
GstAjaSink *self = GST_AJA_SINK(object);
g_assert(self->device == NULL);
g_assert(gst_queue_array_get_length(self->queue) == 0);
g_clear_pointer(&self->queue, gst_queue_array_free);
g_mutex_clear(&self->queue_lock);
g_cond_clear(&self->queue_cond);
g_cond_clear(&self->drain_cond);
G_OBJECT_CLASS(parent_class)->finalize(object);
}
static gboolean gst_aja_sink_open(GstAjaSink *self) {
GST_DEBUG_OBJECT(self, "Opening device");
g_assert(self->device == NULL);
self->device = gst_aja_device_obtain(self->device_identifier);
if (!self->device) {
GST_ERROR_OBJECT(self, "Failed to open device");
return FALSE;
}
if (!self->device->device->IsDeviceReady(false)) {
g_clear_pointer(&self->device, gst_aja_device_unref);
return FALSE;
}
self->device->device->SetEveryFrameServices(::NTV2_OEM_TASKS);
self->device_id = self->device->device->GetDeviceID();
std::string serial_number;
if (!self->device->device->GetSerialNumberString(serial_number))
serial_number = "none";
GST_DEBUG_OBJECT(self,
"Opened device with ID %d at index %d (%s, version %s, "
"serial number %s, can do VANC %d)",
self->device_id, self->device->device->GetIndexNumber(),
self->device->device->GetDisplayName().c_str(),
self->device->device->GetDeviceVersionString().c_str(),
serial_number.c_str(),
::NTV2DeviceCanDoCustomAnc(self->device_id));
GST_DEBUG_OBJECT(self,
"Using SDK version %d.%d.%d.%d (%s) and driver version %s",
AJA_NTV2_SDK_VERSION_MAJOR, AJA_NTV2_SDK_VERSION_MINOR,
AJA_NTV2_SDK_VERSION_POINT, AJA_NTV2_SDK_BUILD_NUMBER,
AJA_NTV2_SDK_BUILD_DATETIME,
self->device->device->GetDriverVersionString().c_str());
self->device->device->SetMultiFormatMode(true);
self->allocator = gst_aja_allocator_new(self->device);
GST_DEBUG_OBJECT(self, "Opened device");
return TRUE;
}
static gboolean gst_aja_sink_close(GstAjaSink *self) {
gst_clear_object(&self->allocator);
g_clear_pointer(&self->device, gst_aja_device_unref);
self->device_id = DEVICE_ID_INVALID;
GST_DEBUG_OBJECT(self, "Closed device");
return TRUE;
}
static gboolean gst_aja_sink_start(GstAjaSink *self) {
GST_DEBUG_OBJECT(self, "Starting");
self->output_thread = new AJAThread();
self->output_thread->Attach(output_thread_func, self);
self->output_thread->SetPriority(AJA_ThreadPriority_High);
self->output_thread->Start();
g_mutex_lock(&self->queue_lock);
self->shutdown = FALSE;
self->playing = FALSE;
self->eos = FALSE;
g_cond_signal(&self->queue_cond);
g_mutex_unlock(&self->queue_lock);
return TRUE;
}
static gboolean gst_aja_sink_stop(GstAjaSink *self) {
QueueItem *item;
GST_DEBUG_OBJECT(self, "Stopping");
g_mutex_lock(&self->queue_lock);
self->shutdown = TRUE;
self->playing = FALSE;
g_cond_signal(&self->queue_cond);
g_mutex_unlock(&self->queue_lock);
if (self->output_thread) {
self->output_thread->Stop();
delete self->output_thread;
self->output_thread = NULL;
}
GST_OBJECT_LOCK(self);
gst_clear_caps(&self->configured_caps);
self->configured_audio_channels = 0;
GST_OBJECT_UNLOCK(self);
while ((item = (QueueItem *)gst_queue_array_pop_head_struct(self->queue))) {
if (item->type == QUEUE_ITEM_TYPE_FRAME) {
gst_video_frame_unmap(&item->frame);
if (item->audio_buffer) {
gst_buffer_unmap(item->audio_buffer, &item->audio_map);
gst_buffer_unref(item->audio_buffer);
}
if (item->anc_packet_list) {
delete item->anc_packet_list;
}
}
}
if (self->buffer_pool) {
gst_buffer_pool_set_active(self->buffer_pool, FALSE);
gst_clear_object(&self->buffer_pool);
}
if (self->audio_buffer_pool) {
gst_buffer_pool_set_active(self->audio_buffer_pool, FALSE);
gst_clear_object(&self->audio_buffer_pool);
}
GST_DEBUG_OBJECT(self, "Stopped");
return TRUE;
}
static GstStateChangeReturn gst_aja_sink_change_state(
GstElement *element, GstStateChange transition) {
GstAjaSink *self = GST_AJA_SINK(element);
GstStateChangeReturn ret;
switch (transition) {
case GST_STATE_CHANGE_NULL_TO_READY:
if (!gst_aja_sink_open(self)) return GST_STATE_CHANGE_FAILURE;
break;
case GST_STATE_CHANGE_READY_TO_PAUSED:
if (!gst_aja_sink_start(self)) return GST_STATE_CHANGE_FAILURE;
break;
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
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_PLAYING_TO_PAUSED:
g_mutex_lock(&self->queue_lock);
self->playing = FALSE;
g_cond_signal(&self->queue_cond);
g_mutex_unlock(&self->queue_lock);
break;
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
g_mutex_lock(&self->queue_lock);
self->playing = TRUE;
g_cond_signal(&self->queue_cond);
g_mutex_unlock(&self->queue_lock);
break;
case GST_STATE_CHANGE_PAUSED_TO_READY:
if (!gst_aja_sink_stop(self)) return GST_STATE_CHANGE_FAILURE;
break;
case GST_STATE_CHANGE_READY_TO_NULL:
if (!gst_aja_sink_close(self)) return GST_STATE_CHANGE_FAILURE;
break;
default:
break;
}
return ret;
}
static gboolean gst_aja_sink_set_caps(GstBaseSink *bsink, GstCaps *caps) {
GstAjaSink *self = GST_AJA_SINK(bsink);
const GstStructure *s;
NTV2VideoFormat video_format = ::NTV2_FORMAT_UNKNOWN;
GST_DEBUG_OBJECT(self, "Configuring caps %" GST_PTR_FORMAT, caps);
GST_OBJECT_LOCK(self);
if (self->configured_caps) {
if (!gst_caps_can_intersect(self->configured_caps, caps)) {
GST_DEBUG_OBJECT(self, "Need to reconfigure, waiting for draining");
GST_OBJECT_UNLOCK(self);
g_mutex_lock(&self->queue_lock);
self->draining = TRUE;
g_cond_signal(&self->queue_cond);
while (self->draining && !self->flushing && !self->shutdown) {
g_cond_wait(&self->drain_cond, &self->queue_lock);
}
if (self->flushing || self->shutdown) {
g_mutex_unlock(&self->queue_lock);
GST_DEBUG_OBJECT(self, "Flushing");
return FALSE;
}
g_mutex_unlock(&self->queue_lock);
GST_OBJECT_LOCK(self);
} else {
GST_OBJECT_UNLOCK(self);
GST_DEBUG_OBJECT(self,
"Compatible caps with previous caps, not reconfiguring");
return TRUE;
}
}
if (!gst_video_info_from_caps(&self->configured_info, caps)) {
GST_OBJECT_UNLOCK(self);
GST_FIXME_OBJECT(self, "Failed to parse caps");
return FALSE;
}
self->configured_audio_channels = 0;
s = gst_caps_get_structure(caps, 0);
gst_structure_get_int(s, "audio-channels", &self->configured_audio_channels);
gst_caps_replace(&self->configured_caps, caps);
GST_OBJECT_UNLOCK(self);
video_format = gst_ntv2_video_format_from_caps(caps);
if (video_format == NTV2_FORMAT_UNKNOWN) {
GST_ERROR_OBJECT(self, "Unsupported caps %" GST_PTR_FORMAT, caps);
return FALSE;
}
self->video_format = video_format;
// Configure render delay based on the framerate and queue size
gst_base_sink_set_render_delay(
GST_BASE_SINK(self),
gst_util_uint64_scale(self->queue_size / 2,
self->configured_info.fps_d * GST_SECOND,
self->configured_info.fps_n));
g_assert(self->device != NULL);
// Make sure to globally lock here as the routing settings and others are
// global shared state
ShmMutexLocker locker;
if (!::NTV2DeviceCanDoVideoFormat(self->device_id, video_format)) {
GST_ERROR_OBJECT(self, "Device does not support mode %d",
(int)video_format);
return FALSE;
}
self->device->device->SetMode(self->channel, NTV2_MODE_DISPLAY, false);
GST_DEBUG_OBJECT(self, "Configuring video format %d on channel %d",
(int)video_format, (int)self->channel);
self->device->device->SetVideoFormat(video_format, false, false,
self->channel);
if (!::NTV2DeviceCanDoFrameBufferFormat(self->device_id,
::NTV2_FBF_10BIT_YCBCR)) {
GST_ERROR_OBJECT(self, "Device does not support frame buffer format %d",
(int)::NTV2_FBF_10BIT_YCBCR);
return FALSE;
}
self->device->device->SetFrameBufferFormat(self->channel,
::NTV2_FBF_10BIT_YCBCR);
NTV2ReferenceSource reference_source;
switch (self->reference_source) {
case GST_AJA_REFERENCE_SOURCE_EXTERNAL:
reference_source = ::NTV2_REFERENCE_EXTERNAL;
break;
case GST_AJA_REFERENCE_SOURCE_FREERUN:
case GST_AJA_REFERENCE_SOURCE_AUTO:
reference_source = ::NTV2_REFERENCE_FREERUN;
break;
case GST_AJA_REFERENCE_SOURCE_INPUT_1:
reference_source = ::NTV2_REFERENCE_INPUT1;
break;
case GST_AJA_REFERENCE_SOURCE_INPUT_2:
reference_source = ::NTV2_REFERENCE_INPUT2;
break;
case GST_AJA_REFERENCE_SOURCE_INPUT_3:
reference_source = ::NTV2_REFERENCE_INPUT3;
break;
case GST_AJA_REFERENCE_SOURCE_INPUT_4:
reference_source = ::NTV2_REFERENCE_INPUT4;
break;
case GST_AJA_REFERENCE_SOURCE_INPUT_5:
reference_source = ::NTV2_REFERENCE_INPUT5;
break;
case GST_AJA_REFERENCE_SOURCE_INPUT_6:
reference_source = ::NTV2_REFERENCE_INPUT6;
break;
case GST_AJA_REFERENCE_SOURCE_INPUT_7:
reference_source = ::NTV2_REFERENCE_INPUT7;
break;
case GST_AJA_REFERENCE_SOURCE_INPUT_8:
reference_source = ::NTV2_REFERENCE_INPUT8;
break;
default:
g_assert_not_reached();
break;
}
GST_DEBUG_OBJECT(self, "Configuring reference source %d",
(int)reference_source);
self->device->device->SetFramePulseReference(reference_source);
if (!self->device->device->EnableChannel(self->channel)) {
GST_ERROR_OBJECT(self, "Failed to enable channel");
return FALSE;
}
self->device->device->DMABufferAutoLock(false, true, 0);
if (::NTV2DeviceHasBiDirectionalSDI(self->device_id))
self->device->device->SetSDITransmitEnable(self->channel, true);
const NTV2Standard standard(::GetNTV2StandardFromVideoFormat(video_format));
self->device->device->SetSDIOutputStandard(self->channel, standard);
const NTV2FrameGeometry geometry =
::GetNTV2FrameGeometryFromVideoFormat(video_format);
self->device->device->SetVANCMode(::NTV2_VANCMODE_OFF, standard, geometry,
self->channel);
NTV2SmpteLineNumber smpte_line_num_info = ::GetSmpteLineNumber(standard);
self->f2_start_line =
(smpte_line_num_info.GetLastLine(
smpte_line_num_info.firstFieldTop ? NTV2_FIELD0 : NTV2_FIELD1) +
1);
if (self->configured_audio_channels) {
switch (self->audio_system_setting) {
case GST_AJA_AUDIO_SYSTEM_1:
self->audio_system = ::NTV2_AUDIOSYSTEM_1;
break;
case GST_AJA_AUDIO_SYSTEM_2:
self->audio_system = ::NTV2_AUDIOSYSTEM_2;
break;
case GST_AJA_AUDIO_SYSTEM_3:
self->audio_system = ::NTV2_AUDIOSYSTEM_3;
break;
case GST_AJA_AUDIO_SYSTEM_4:
self->audio_system = ::NTV2_AUDIOSYSTEM_4;
break;
case GST_AJA_AUDIO_SYSTEM_5:
self->audio_system = ::NTV2_AUDIOSYSTEM_5;
break;
case GST_AJA_AUDIO_SYSTEM_6:
self->audio_system = ::NTV2_AUDIOSYSTEM_6;
break;
case GST_AJA_AUDIO_SYSTEM_7:
self->audio_system = ::NTV2_AUDIOSYSTEM_7;
break;
case GST_AJA_AUDIO_SYSTEM_8:
self->audio_system = ::NTV2_AUDIOSYSTEM_8;
break;
case GST_AJA_AUDIO_SYSTEM_AUTO:
self->audio_system = ::NTV2_AUDIOSYSTEM_1;
if (::NTV2DeviceGetNumAudioSystems(self->device_id) > 1)
self->audio_system = ::NTV2ChannelToAudioSystem(self->channel);
if (!::NTV2DeviceCanDoFrameStore1Display(self->device_id))
self->audio_system = ::NTV2_AUDIOSYSTEM_1;
break;
default:
g_assert_not_reached();
break;
}
GST_DEBUG_OBJECT(self, "Using audio system %d", self->audio_system);
self->device->device->SetNumberAudioChannels(
self->configured_audio_channels, self->audio_system);
self->device->device->SetAudioRate(::NTV2_AUDIO_48K, self->audio_system);
self->device->device->SetAudioBufferSize(::NTV2_AUDIO_BUFFER_BIG,
self->audio_system);
self->device->device->SetSDIOutputAudioSystem(self->channel,
self->audio_system);
self->device->device->SetSDIOutputDS2AudioSystem(self->channel,
self->audio_system);
self->device->device->SetAudioLoopBack(::NTV2_AUDIO_LOOPBACK_OFF,
self->audio_system);
} else {
self->audio_system = ::NTV2_AUDIOSYSTEM_INVALID;
}
CNTV2SignalRouter router;
self->device->device->GetRouting(router);
// Always use the framebuffer associated with the channel
NTV2OutputCrosspointID framebuffer_id =
::GetFrameBufferOutputXptFromChannel(self->channel, false, false);
NTV2InputCrosspointID output_destination_id;
switch (self->output_destination) {
case GST_AJA_OUTPUT_DESTINATION_AUTO:
output_destination_id = ::GetSDIOutputInputXpt(self->channel, false);
break;
case GST_AJA_OUTPUT_DESTINATION_SDI1:
output_destination_id = ::NTV2_XptSDIOut1Input;
break;
case GST_AJA_OUTPUT_DESTINATION_SDI2:
output_destination_id = ::NTV2_XptSDIOut2Input;
break;
case GST_AJA_OUTPUT_DESTINATION_SDI3:
output_destination_id = ::NTV2_XptSDIOut3Input;
break;
case GST_AJA_OUTPUT_DESTINATION_SDI4:
output_destination_id = ::NTV2_XptSDIOut4Input;
break;
case GST_AJA_OUTPUT_DESTINATION_SDI5:
output_destination_id = ::NTV2_XptSDIOut5Input;
break;
case GST_AJA_OUTPUT_DESTINATION_SDI6:
output_destination_id = ::NTV2_XptSDIOut6Input;
break;
case GST_AJA_OUTPUT_DESTINATION_SDI7:
output_destination_id = ::NTV2_XptSDIOut7Input;
break;
case GST_AJA_OUTPUT_DESTINATION_SDI8:
output_destination_id = ::NTV2_XptSDIOut8Input;
break;
case GST_AJA_OUTPUT_DESTINATION_ANALOG:
output_destination_id = ::NTV2_XptAnalogOutInput;
break;
case GST_AJA_OUTPUT_DESTINATION_HDMI:
output_destination_id = ::NTV2_XptHDMIOutInput;
break;
default:
g_assert_not_reached();
break;
}
// Need to remove old routes for the output and framebuffer we're going to use
NTV2ActualConnections connections = router.GetConnections();
for (NTV2ActualConnectionsConstIter iter = connections.begin();
iter != connections.end(); iter++) {
if (iter->first == output_destination_id || iter->second == framebuffer_id)
router.RemoveConnection(iter->first, iter->second);
}
GST_DEBUG_OBJECT(self, "Creating connection %d - %d", output_destination_id,
framebuffer_id);
router.AddConnection(output_destination_id, framebuffer_id);
{
std::stringstream os;
CNTV2SignalRouter oldRouter;
self->device->device->GetRouting(oldRouter);
oldRouter.Print(os);
GST_DEBUG_OBJECT(self, "Previous routing:\n%s", os.str().c_str());
}
self->device->device->ApplySignalRoute(router, true);
{
std::stringstream os;
CNTV2SignalRouter currentRouter;
self->device->device->GetRouting(currentRouter);
currentRouter.Print(os);
GST_DEBUG_OBJECT(self, "New routing:\n%s", os.str().c_str());
}
return TRUE;
}
static GstCaps *gst_aja_sink_get_caps(GstBaseSink *bsink, GstCaps *filter) {
GstAjaSink *self = GST_AJA_SINK(bsink);
GstCaps *caps;
if (self->device) {
caps = gst_ntv2_supported_caps(self->device_id);
} else {
caps = gst_pad_get_pad_template_caps(GST_BASE_SINK_PAD(self));
}
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_aja_sink_event(GstBaseSink *bsink, GstEvent *event) {
GstAjaSink *self = GST_AJA_SINK(bsink);
switch (GST_EVENT_TYPE(event)) {
case GST_EVENT_EOS: {
GST_DEBUG_OBJECT(self, "Signalling EOS");
g_mutex_lock(&self->queue_lock);
self->eos = TRUE;
g_cond_signal(&self->queue_cond);
g_mutex_unlock(&self->queue_lock);
break;
}
case GST_EVENT_FLUSH_START: {
g_mutex_lock(&self->queue_lock);
self->flushing = TRUE;
self->draining = FALSE;
g_cond_signal(&self->drain_cond);
g_mutex_unlock(&self->queue_lock);
break;
}
case GST_EVENT_FLUSH_STOP: {
QueueItem *item;
g_mutex_lock(&self->queue_lock);
while (
(item = (QueueItem *)gst_queue_array_pop_head_struct(self->queue))) {
if (item->type == QUEUE_ITEM_TYPE_FRAME) {
gst_video_frame_unmap(&item->frame);
if (item->audio_buffer) {
gst_buffer_unmap(item->audio_buffer, &item->audio_map);
gst_buffer_unref(item->audio_buffer);
}
if (item->anc_packet_list) {
delete item->anc_packet_list;
}
}
}
g_cond_signal(&self->queue_cond);
self->flushing = FALSE;
g_cond_signal(&self->drain_cond);
g_mutex_unlock(&self->queue_lock);
break;
}
default:
break;
}
return GST_BASE_SINK_CLASS(parent_class)->event(bsink, event);
}
static gboolean gst_aja_sink_propose_allocation(GstBaseSink *bsink,
GstQuery *query) {
GstAjaSink *self = GST_AJA_SINK(bsink);
if (self->allocator) {
GstAllocationParams params;
gst_allocation_params_init(&params);
params.prefix = 0;
params.padding = 0;
params.align = 4095;
gst_query_add_allocation_param(query, self->allocator, &params);
}
return TRUE;
}
static GstFlowReturn gst_aja_sink_render(GstBaseSink *bsink,
GstBuffer *buffer) {
GstAjaSink *self = GST_AJA_SINK(bsink);
GstFlowReturn flow_ret = GST_FLOW_OK;
GstAjaAudioMeta *meta;
GstBuffer *item_buffer = NULL, *item_audio_buffer = NULL;
GstVideoTimeCodeMeta *tc_meta;
QueueItem item = {
.type = QUEUE_ITEM_TYPE_FRAME,
.frame =
{
{0},
},
.audio_buffer = NULL,
.audio_map = GST_MAP_INFO_INIT,
.tc = NTV2_RP188(),
.anc_packet_list = NULL,
};
guint video_buffer_size = ::GetVideoActiveSize(
self->video_format, ::NTV2_FBF_10BIT_YCBCR, ::NTV2_VANCMODE_OFF);
meta = gst_buffer_get_aja_audio_meta(buffer);
tc_meta = gst_buffer_get_video_time_code_meta(buffer);
if (gst_buffer_n_memory(buffer) == 1) {
GstMemory *mem = gst_buffer_peek_memory(buffer, 0);
if (gst_memory_get_sizes(mem, NULL, NULL) == video_buffer_size &&
strcmp(mem->allocator->mem_type, GST_AJA_ALLOCATOR_MEMTYPE) == 0 &&
GST_AJA_ALLOCATOR(mem->allocator)->device->device->GetIndexNumber() ==
self->device->device->GetIndexNumber()) {
item_buffer = gst_buffer_ref(buffer);
}
}
if (!item_buffer) {
GstVideoFrame in_frame;
GST_DEBUG_OBJECT(self, "Allocating new video buffer");
if (!self->buffer_pool) {
self->buffer_pool = gst_buffer_pool_new();
GstStructure *config = gst_buffer_pool_get_config(self->buffer_pool);
gst_buffer_pool_config_set_params(config, NULL, video_buffer_size,
self->queue_size, 0);
gst_buffer_pool_config_set_allocator(config, self->allocator, NULL);
gst_buffer_pool_set_config(self->buffer_pool, config);
gst_buffer_pool_set_active(self->buffer_pool, TRUE);
}
if (!gst_video_frame_map(&in_frame, &self->configured_info, buffer,
GST_MAP_READ)) {
GST_ERROR_OBJECT(self, "Failed to map buffer");
return GST_FLOW_ERROR;
}
flow_ret =
gst_buffer_pool_acquire_buffer(self->buffer_pool, &item_buffer, NULL);
if (flow_ret != GST_FLOW_OK) {
gst_video_frame_unmap(&in_frame);
return flow_ret;
}
item.type = QUEUE_ITEM_TYPE_FRAME;
gst_video_frame_map(&item.frame, &self->configured_info, item_buffer,
GST_MAP_READWRITE);
gst_video_frame_copy(&item.frame, &in_frame);
gst_video_frame_unmap(&in_frame);
gst_buffer_unref(item_buffer);
} else {
item.type = QUEUE_ITEM_TYPE_FRAME;
gst_video_frame_map(&item.frame, &self->configured_info, item_buffer,
GST_MAP_READ);
gst_buffer_unref(item_buffer);
}
if (meta) {
if (gst_buffer_n_memory(meta->buffer) == 1) {
GstMemory *mem = gst_buffer_peek_memory(meta->buffer, 0);
if (strcmp(mem->allocator->mem_type, GST_AJA_ALLOCATOR_MEMTYPE) == 0 &&
GST_AJA_ALLOCATOR(mem->allocator)->device->device->GetIndexNumber() ==
self->device->device->GetIndexNumber()) {
item_audio_buffer = gst_buffer_ref(meta->buffer);
}
}
if (!item_audio_buffer) {
GstMapInfo audio_map;
GST_DEBUG_OBJECT(self, "Allocating new audio buffer");
if (!self->audio_buffer_pool) {
guint audio_buffer_size = 1UL * 1024UL * 1024UL;
self->audio_buffer_pool = gst_buffer_pool_new();
GstStructure *config =
gst_buffer_pool_get_config(self->audio_buffer_pool);
gst_buffer_pool_config_set_params(config, NULL, audio_buffer_size,
self->queue_size, 0);
gst_buffer_pool_config_set_allocator(config, self->allocator, NULL);
gst_buffer_pool_set_config(self->audio_buffer_pool, config);
gst_buffer_pool_set_active(self->audio_buffer_pool, TRUE);
}
flow_ret = gst_buffer_pool_acquire_buffer(self->audio_buffer_pool,
&item_audio_buffer, NULL);
if (flow_ret != GST_FLOW_OK) {
gst_video_frame_unmap(&item.frame);
return flow_ret;
}
gst_buffer_set_size(item_audio_buffer, gst_buffer_get_size(meta->buffer));
gst_buffer_map(meta->buffer, &audio_map, GST_MAP_READ);
gst_buffer_map(item_audio_buffer, &item.audio_map, GST_MAP_READWRITE);
memcpy(item.audio_map.data, audio_map.data, audio_map.size);
gst_buffer_unmap(meta->buffer, &audio_map);
item.audio_buffer = item_audio_buffer;
} else {
gst_buffer_map(item_audio_buffer, &item.audio_map, GST_MAP_READ);
item.audio_buffer = item_audio_buffer;
}
} else {
item.audio_buffer = NULL;
}
if (tc_meta) {
TimecodeFormat tc_format = ::kTCFormatUnknown;
if (tc_meta->tc.config.fps_n == 24 && tc_meta->tc.config.fps_d == 1) {
tc_format = kTCFormat24fps;
} else if (tc_meta->tc.config.fps_n == 25 &&
tc_meta->tc.config.fps_d == 1) {
tc_format = kTCFormat25fps;
} else if (tc_meta->tc.config.fps_n == 30 &&
tc_meta->tc.config.fps_d == 1) {
tc_format = kTCFormat30fps;
} else if (tc_meta->tc.config.fps_n == 30000 &&
tc_meta->tc.config.fps_d == 1001) {
tc_format = kTCFormat30fpsDF;
} else if (tc_meta->tc.config.fps_n == 48 &&
tc_meta->tc.config.fps_d == 1) {
tc_format = kTCFormat48fps;
} else if (tc_meta->tc.config.fps_n == 50 &&
tc_meta->tc.config.fps_d == 1) {
tc_format = kTCFormat50fps;
} else if (tc_meta->tc.config.fps_n == 60 &&
tc_meta->tc.config.fps_d == 1) {
tc_format = kTCFormat60fps;
} else if (tc_meta->tc.config.fps_n == 60000 &&
tc_meta->tc.config.fps_d == 1001) {
tc_format = kTCFormat60fpsDF;
}
const CRP188 rp188(tc_meta->tc.frames, tc_meta->tc.seconds,
tc_meta->tc.minutes, tc_meta->tc.hours, tc_format);
rp188.GetRP188Reg(item.tc);
} else {
item.tc.fDBB = 0xffffffff;
}
// TODO: Handle AFD/Bar meta
#if 0
if (bar_meta || afd_meta) {
const uint16_t kF1PktLineNumAFDBAR(11);
const AJAAncillaryDataLocation kAFDBARLocF1(
AJAAncillaryDataLink_A, AJAAncillaryDataVideoStream_Y,
AJAAncillaryDataSpace_VANC, kF1PktLineNumAFDBAR,
AJAAncDataHorizOffset_AnyVanc);
const uint16_t kF2PktLineNumAFDBAR(self->f2_start_line + 11);
const AJAAncillaryDataLocation kAFDBARLocF2(
AJAAncillaryDataLink_A, AJAAncillaryDataVideoStream_Y,
AJAAncillaryDataSpace_VANC, kF2PktLineNumAFDBAR,
AJAAncDataHorizOffset_AnyVanc);
AJAAncillaryData pkt;
pkt.SetFromSMPTE334(NULL, 0, kAFDBARLocF1);
item.anc_packet_list->AddAncillaryData(pkt);
if (self->configured_info.interlace_mode != GST_VIDEO_INTERLACE_MODE_PROGRESSIVE) {
AJAAncillaryData pkt2;
pkt.SetFromSMPTE334(NULL, 0, kAFDBARLocF2);
item.anc_packet_list->AddAncillaryData(pkt);
}
}
#endif
GstVideoCaptionMeta *caption_meta;
gpointer iter = NULL;
while (
(caption_meta = (GstVideoCaptionMeta *)gst_buffer_iterate_meta_filtered(
buffer, &iter, GST_VIDEO_CAPTION_META_API_TYPE))) {
if (!item.anc_packet_list) item.anc_packet_list = new AJAAncillaryList;
if (caption_meta->caption_type == GST_VIDEO_CAPTION_TYPE_CEA708_CDP) {
const uint16_t kF1PktLineNumCEA708(9);
const AJAAncillaryDataLocation kCEA708LocF1(
AJAAncillaryDataLink_A, AJAAncillaryDataVideoStream_Y,
AJAAncillaryDataSpace_VANC, kF1PktLineNumCEA708,
AJAAncDataHorizOffset_AnyVanc);
AJAAncillaryData_Cea708 pkt;
pkt.SetDID(GST_VIDEO_ANCILLARY_DID16_S334_EIA_708 >> 8);
pkt.SetSID(GST_VIDEO_ANCILLARY_DID16_S334_EIA_708 & 0xff);
pkt.SetDataLocation(kCEA708LocF1);
pkt.SetDataCoding(AJAAncillaryDataCoding_Digital);
pkt.SetPayloadData(caption_meta->data, caption_meta->size);
item.anc_packet_list->AddAncillaryData(pkt);
} else {
GST_WARNING_OBJECT(self, "Unhandled caption type %d",
caption_meta->caption_type);
}
}
g_mutex_lock(&self->queue_lock);
while (gst_queue_array_get_length(self->queue) >= self->queue_size) {
QueueItem *tmp = (QueueItem *)gst_queue_array_pop_head_struct(self->queue);
if (tmp->type == QUEUE_ITEM_TYPE_FRAME) {
GST_WARNING_OBJECT(self, "Element queue overrun, dropping old frame");
GstMessage *msg = gst_message_new_qos(
GST_OBJECT_CAST(self), TRUE, GST_CLOCK_TIME_NONE, GST_CLOCK_TIME_NONE,
GST_BUFFER_PTS(tmp->frame.buffer),
gst_util_uint64_scale(GST_SECOND, self->configured_info.fps_d,
self->configured_info.fps_n));
gst_element_post_message(GST_ELEMENT_CAST(self), msg);
gst_video_frame_unmap(&tmp->frame);
if (tmp->audio_buffer) {
gst_buffer_unmap(tmp->audio_buffer, &tmp->audio_map);
gst_buffer_unref(tmp->audio_buffer);
}
if (tmp->anc_packet_list) {
delete tmp->anc_packet_list;
}
}
}
GST_TRACE_OBJECT(self, "Queuing frame video %p audio %p",
GST_VIDEO_FRAME_PLANE_DATA(&item.frame, 0),
item.audio_buffer ? item.audio_map.data : NULL);
gst_queue_array_push_tail_struct(self->queue, &item);
GST_TRACE_OBJECT(self, "%u frames queued",
gst_queue_array_get_length(self->queue));
g_cond_signal(&self->queue_cond);
g_mutex_unlock(&self->queue_lock);
return flow_ret;
}
static void output_thread_func(AJAThread *thread, void *data) {
GstAjaSink *self = GST_AJA_SINK(data);
GstClock *clock = NULL;
guint64 frames_renderded_start = G_MAXUINT64;
GstClockTime frames_renderded_start_time = GST_CLOCK_TIME_NONE;
guint64 frames_dropped_last = G_MAXUINT64;
AUTOCIRCULATE_TRANSFER transfer;
if (self->output_cpu_core != G_MAXUINT) {
cpu_set_t mask;
pthread_t current_thread = pthread_self();
CPU_ZERO(&mask);
CPU_SET(self->output_cpu_core, &mask);
if (pthread_setaffinity_np(current_thread, sizeof(mask), &mask) != 0) {
GST_ERROR_OBJECT(self,
"Failed to set affinity for current thread to core %u",
self->output_cpu_core);
}
}
g_mutex_lock(&self->queue_lock);
restart:
if (self->draining && gst_queue_array_get_length(self->queue) == 0) {
GST_DEBUG_OBJECT(self, "Drained");
self->draining = FALSE;
g_cond_signal(&self->drain_cond);
}
GST_DEBUG_OBJECT(self, "Waiting for playing or shutdown");
while ((!self->playing && !self->shutdown) ||
(self->playing &&
gst_queue_array_get_length(self->queue) < self->queue_size / 2 &&
!self->eos))
g_cond_wait(&self->queue_cond, &self->queue_lock);
if (self->shutdown) {
GST_DEBUG_OBJECT(self, "Shutting down");
g_mutex_unlock(&self->queue_lock);
return;
}
GST_DEBUG_OBJECT(self, "Starting playing");
g_mutex_unlock(&self->queue_lock);
NTV2TCIndexes tc_indexes;
switch (self->timecode_index) {
case GST_AJA_TIMECODE_INDEX_AUTO:
tc_indexes.insert(::NTV2ChannelToTimecodeIndex(self->channel, false));
tc_indexes.insert(::NTV2ChannelToTimecodeIndex(self->channel, true));
if (self->configured_info.interlace_mode !=
GST_VIDEO_INTERLACE_MODE_PROGRESSIVE)
tc_indexes.insert(
::NTV2ChannelToTimecodeIndex(self->channel, false, true));
break;
case GST_AJA_TIMECODE_INDEX_SDI1:
tc_indexes.insert(::NTV2_TCINDEX_SDI1);
break;
case GST_AJA_TIMECODE_INDEX_SDI2:
tc_indexes.insert(::NTV2_TCINDEX_SDI2);
break;
case GST_AJA_TIMECODE_INDEX_SDI3:
tc_indexes.insert(::NTV2_TCINDEX_SDI3);
break;
case GST_AJA_TIMECODE_INDEX_SDI4:
tc_indexes.insert(::NTV2_TCINDEX_SDI4);
break;
case GST_AJA_TIMECODE_INDEX_SDI5:
tc_indexes.insert(::NTV2_TCINDEX_SDI5);
break;
case GST_AJA_TIMECODE_INDEX_SDI6:
tc_indexes.insert(::NTV2_TCINDEX_SDI6);
break;
case GST_AJA_TIMECODE_INDEX_SDI7:
tc_indexes.insert(::NTV2_TCINDEX_SDI7);
break;
case GST_AJA_TIMECODE_INDEX_SDI8:
tc_indexes.insert(::NTV2_TCINDEX_SDI8);
break;
case GST_AJA_TIMECODE_INDEX_SDI1_LTC:
tc_indexes.insert(::NTV2_TCINDEX_SDI1_LTC);
break;
case GST_AJA_TIMECODE_INDEX_SDI2_LTC:
tc_indexes.insert(::NTV2_TCINDEX_SDI2_LTC);
break;
case GST_AJA_TIMECODE_INDEX_SDI3_LTC:
tc_indexes.insert(::NTV2_TCINDEX_SDI3_LTC);
break;
case GST_AJA_TIMECODE_INDEX_SDI4_LTC:
tc_indexes.insert(::NTV2_TCINDEX_SDI4_LTC);
break;
case GST_AJA_TIMECODE_INDEX_SDI5_LTC:
tc_indexes.insert(::NTV2_TCINDEX_SDI5_LTC);
break;
case GST_AJA_TIMECODE_INDEX_SDI6_LTC:
tc_indexes.insert(::NTV2_TCINDEX_SDI6_LTC);
break;
case GST_AJA_TIMECODE_INDEX_SDI7_LTC:
tc_indexes.insert(::NTV2_TCINDEX_SDI7_LTC);
break;
case GST_AJA_TIMECODE_INDEX_SDI8_LTC:
tc_indexes.insert(::NTV2_TCINDEX_SDI8_LTC);
break;
case GST_AJA_TIMECODE_INDEX_LTC1:
tc_indexes.insert(::NTV2_TCINDEX_LTC1);
break;
case GST_AJA_TIMECODE_INDEX_LTC2:
tc_indexes.insert(::NTV2_TCINDEX_LTC2);
break;
default:
g_assert_not_reached();
break;
}
{
// Make sure to globally lock here as the routing settings and others are
// global shared state
ShmMutexLocker locker;
self->device->device->AutoCirculateStop(self->channel);
self->device->device->EnableOutputInterrupt(self->channel);
self->device->device->SubscribeOutputVerticalEvent(self->channel);
if (!self->device->device->AutoCirculateInitForOutput(
self->channel, self->queue_size / 2, self->audio_system,
AUTOCIRCULATE_WITH_RP188 | AUTOCIRCULATE_WITH_ANC, 1)) {
GST_ELEMENT_ERROR(self, STREAM, FAILED, (NULL),
("Failed to initialize autocirculate"));
goto out;
}
self->device->device->AutoCirculateStart(self->channel);
}
gst_clear_object(&clock);
clock = gst_element_get_clock(GST_ELEMENT_CAST(self));
frames_renderded_start = G_MAXUINT64;
frames_renderded_start_time = GST_CLOCK_TIME_NONE;
frames_dropped_last = G_MAXUINT64;
transfer.acANCBuffer.Allocate(2048);
if (self->configured_info.interlace_mode !=
GST_VIDEO_INTERLACE_MODE_INTERLEAVED)
transfer.acANCField2Buffer.Allocate(2048);
g_mutex_lock(&self->queue_lock);
while (self->playing && !self->shutdown &&
!(self->draining && gst_queue_array_get_length(self->queue) == 0)) {
AUTOCIRCULATE_STATUS status;
self->device->device->AutoCirculateGetStatus(self->channel, status);
GST_TRACE_OBJECT(self,
"Start frame %d "
"end frame %d "
"active frame %d "
"start time %" G_GUINT64_FORMAT
" "
"current time %" G_GUINT64_FORMAT
" "
"frames processed %u "
"frames dropped %u "
"buffer level %u",
status.acStartFrame, status.acEndFrame,
status.acActiveFrame, status.acRDTSCStartTime,
status.acRDTSCCurrentTime, status.acFramesProcessed,
status.acFramesDropped, status.acBufferLevel);
// Trivial drift calculation
//
// TODO: Should probably take averages over a timespan (say 1 minute) into a
// ringbuffer and calculate a linear regression over them
// FIXME: Add some compensation by dropping/duplicating frames as needed
// but make this configurable
// FIXME: Should use transfer.acTransferStatus.acFrameStamp after
// AutoCirculateTransfer()
if (frames_renderded_start_time == GST_CLOCK_TIME_NONE &&
status.acRDTSCStartTime != 0 &&
status.acFramesProcessed + status.acFramesDropped > self->queue_size &&
clock) {
frames_renderded_start =
status.acFramesProcessed + status.acFramesDropped;
frames_renderded_start_time = gst_clock_get_time(clock);
}
if (clock && frames_renderded_start_time != GST_CLOCK_TIME_NONE) {
GstClockTime now = gst_clock_get_time(clock);
GstClockTime diff = now - frames_renderded_start_time;
guint64 frames_rendered =
(status.acFramesProcessed + status.acFramesDropped) -
frames_renderded_start;
guint64 frames_produced =
gst_util_uint64_scale(diff, self->configured_info.fps_n,
self->configured_info.fps_d * GST_SECOND);
gdouble fps_rendered = ((gdouble)frames_rendered * GST_SECOND) / diff;
GST_TRACE_OBJECT(self,
"Frames rendered %" G_GUINT64_FORMAT
", frames produced %" G_GUINT64_FORMAT
", FPS rendered %lf",
frames_rendered, frames_produced, fps_rendered);
}
// Detect if we were too slow with providing frames and report if that was
// the case together with the amount of frames dropped
if (frames_dropped_last == G_MAXUINT64) {
frames_dropped_last = status.acFramesDropped;
} else if (frames_dropped_last < status.acFramesDropped) {
GST_WARNING_OBJECT(self, "Dropped %" G_GUINT64_FORMAT " frames",
status.acFramesDropped - frames_dropped_last);
GstClockTime timestamp =
gst_util_uint64_scale(status.acFramesProcessed + frames_dropped_last,
self->configured_info.fps_n,
self->configured_info.fps_d * GST_SECOND);
GstClockTime timestamp_end = gst_util_uint64_scale(
status.acFramesProcessed + status.acFramesDropped,
self->configured_info.fps_n,
self->configured_info.fps_d * GST_SECOND);
GstMessage *msg = gst_message_new_qos(
GST_OBJECT_CAST(self), TRUE, GST_CLOCK_TIME_NONE, GST_CLOCK_TIME_NONE,
timestamp, timestamp_end - timestamp);
gst_element_post_message(GST_ELEMENT_CAST(self), msg);
frames_dropped_last = status.acFramesDropped;
}
if (status.GetNumAvailableOutputFrames() > 1) {
QueueItem item, *item_p;
while ((item_p = (QueueItem *)gst_queue_array_pop_head_struct(
self->queue)) == NULL &&
self->playing && !self->shutdown && !self->draining) {
GST_DEBUG_OBJECT(
self,
"Element queue underrun, waiting for more frames or shutdown");
g_cond_wait(&self->queue_cond, &self->queue_lock);
}
if (!self->playing || self->shutdown || (!item_p && self->draining)) {
if (item_p && item_p->type == QUEUE_ITEM_TYPE_FRAME) {
gst_video_frame_unmap(&item_p->frame);
if (item_p->audio_buffer) {
gst_buffer_unmap(item_p->audio_buffer, &item_p->audio_map);
gst_buffer_unref(item_p->audio_buffer);
}
if (item_p->anc_packet_list) {
delete item_p->anc_packet_list;
}
}
break;
}
if (item_p && item_p->type != QUEUE_ITEM_TYPE_FRAME) {
continue;
}
GST_TRACE_OBJECT(self, "%u frames queued",
gst_queue_array_get_length(self->queue));
item = *item_p;
g_mutex_unlock(&self->queue_lock);
GST_TRACE_OBJECT(self,
"Transferring frame: "
"Video %p %" G_GSIZE_FORMAT
" "
"Audio %p %" G_GSIZE_FORMAT,
GST_VIDEO_FRAME_PLANE_DATA(&item.frame, 0),
GST_VIDEO_FRAME_SIZE(&item.frame),
item.audio_buffer ? item.audio_map.data : NULL,
item.audio_buffer ? item.audio_map.size : 0);
// Set timecodes if provided by upstream
if (item.tc.IsValid() && item.tc.fDBB != 0xffffffff) {
NTV2TimeCodes timecodes;
for (const auto &tc_index : tc_indexes) {
timecodes[tc_index] = item.tc;
}
transfer.SetOutputTimeCodes(timecodes);
}
transfer.SetVideoBuffer(
(guint *)GST_VIDEO_FRAME_PLANE_DATA(&item.frame, 0),
GST_VIDEO_FRAME_SIZE(&item.frame));
if (item.audio_buffer) {
transfer.SetAudioBuffer((guint *)item.audio_map.data,
item.audio_map.size);
}
// Clear VANC and fill in captions as needed
transfer.acANCBuffer.Fill(ULWord(0));
transfer.acANCField2Buffer.Fill(ULWord(0));
if (item.anc_packet_list) {
item.anc_packet_list->GetTransmitData(
transfer.acANCBuffer, transfer.acANCField2Buffer,
self->configured_info.interlace_mode !=
GST_VIDEO_INTERLACE_MODE_PROGRESSIVE,
self->f2_start_line);
}
if (!self->device->device->AutoCirculateTransfer(self->channel,
transfer)) {
GST_WARNING_OBJECT(self, "Failed to transfer frame");
}
gst_video_frame_unmap(&item.frame);
if (item.audio_buffer) {
gst_buffer_unmap(item.audio_buffer, &item.audio_map);
gst_buffer_unref(item.audio_buffer);
}
if (item.anc_packet_list) {
delete item.anc_packet_list;
}
g_mutex_lock(&self->queue_lock);
} else {
g_mutex_unlock(&self->queue_lock);
self->device->device->WaitForOutputVerticalInterrupt(self->channel);
g_mutex_lock(&self->queue_lock);
}
}
out : {
// Make sure to globally lock here as the routing settings and others are
// global shared state
ShmMutexLocker locker;
self->device->device->AutoCirculateStop(self->channel);
self->device->device->UnsubscribeOutputVerticalEvent(self->channel);
self->device->device->DisableOutputInterrupt(self->channel);
}
if ((!self->playing || self->draining) && !self->shutdown) goto restart;
g_mutex_unlock(&self->queue_lock);
gst_clear_object(&clock);
GST_DEBUG_OBJECT(self, "Stopped");
}
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