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WebKitMediaSourceGStreamer.cpp
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/*
* Copyright (C) 2009, 2010 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* Copyright (C) 2013 Collabora Ltd.
* Copyright (C) 2013 Orange
* 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 Lesser 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
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with this library; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include "config.h"
#include "WebKitMediaSourceGStreamer.h"
#if ENABLE(VIDEO) && ENABLE(MEDIA_SOURCE) && USE(GSTREAMER)
#include "AudioTrackPrivateGStreamer.h"
#include "GStreamerUtilities.h"
#include "MediaDescription.h"
#include "MediaSample.h"
#include "MediaSourceGStreamer.h"
#include "NotImplemented.h"
#include "SourceBufferPrivateGStreamer.h"
#include "TimeRanges.h"
#include "VideoTrackPrivateGStreamer.h"
#include <gst/app/app.h>
#include <gst/app/gstappsrc.h>
#include <gst/gst.h>
#include <gst/pbutils/missing-plugins.h>
#include <gst/pbutils/pbutils.h>
#include <gst/video/video.h>
#include <wtf/text/CString.h>
#include <wtf/gobject/GUniquePtr.h>
namespace WebCore
{
class GStreamerMediaDescription : public MediaDescription {
private:
GstCaps* m_caps;
public:
static PassRefPtr<GStreamerMediaDescription> create(GstCaps* caps)
{
return adoptRef(new GStreamerMediaDescription(caps));
}
virtual ~GStreamerMediaDescription()
{
gst_caps_unref(m_caps);
}
AtomicString codec() const override
{
gchar* description = gst_pb_utils_get_codec_description(m_caps);
AtomicString codecName(description);
g_free(description);
return codecName;
}
bool isVideo() const override
{
GstStructure* s = gst_caps_get_structure(m_caps, 0);
const gchar* name = gst_structure_get_name(s);
return g_str_has_prefix(name, "video/");
}
bool isAudio() const override
{
GstStructure* s = gst_caps_get_structure(m_caps, 0);
const gchar* name = gst_structure_get_name(s);
return g_str_has_prefix(name, "audio/");
}
bool isText() const override
{
// TODO
return false;
}
private:
GStreamerMediaDescription(GstCaps* caps)
: MediaDescription()
, m_caps(gst_caps_ref(caps))
{
}
};
class GStreamerMediaSample : public MediaSample
{
private:
MediaTime m_pts, m_dts, m_duration;
AtomicString m_trackID;
size_t m_size;
GstBuffer* m_buffer;
FloatSize m_presentationSize;
MediaSample::SampleFlags m_flags;
GStreamerMediaSample(GstBuffer* buffer, const FloatSize& presentationSize, const AtomicString& trackID)
: MediaSample()
, m_pts(MediaTime::zeroTime())
, m_dts(MediaTime::zeroTime())
, m_duration(MediaTime::zeroTime())
, m_trackID(trackID)
, m_size(0)
, m_buffer(0)
, m_presentationSize(presentationSize)
, m_flags(MediaSample::IsSync)
{
if (!buffer)
return;
if (GST_BUFFER_PTS_IS_VALID(buffer))
m_pts = MediaTime(GST_BUFFER_PTS(buffer), GST_SECOND);
if (GST_BUFFER_DTS_IS_VALID(buffer))
m_dts = MediaTime(GST_BUFFER_DTS(buffer), GST_SECOND);
if (GST_BUFFER_DURATION_IS_VALID(buffer))
m_duration = MediaTime(GST_BUFFER_DURATION(buffer), GST_SECOND);
m_size = gst_buffer_get_size(buffer);
m_buffer = gst_buffer_ref(buffer);
if (GST_BUFFER_FLAG_IS_SET(buffer, GST_BUFFER_FLAG_DELTA_UNIT))
m_flags = MediaSample::None;
if (GST_BUFFER_FLAG_IS_SET(buffer, GST_BUFFER_FLAG_DECODE_ONLY))
m_flags = (MediaSample::SampleFlags) (m_flags | MediaSample::NonDisplaying);
}
public:
static PassRefPtr<GStreamerMediaSample> create(GstBuffer* buffer, const FloatSize& presentationSize, const AtomicString& trackID)
{
return adoptRef(new GStreamerMediaSample(buffer, presentationSize, trackID));
}
static PassRefPtr<GStreamerMediaSample> createFakeSample(MediaTime pts, MediaTime dts, MediaTime duration, const FloatSize& presentationSize, const AtomicString& trackID)
{
GStreamerMediaSample* s = new GStreamerMediaSample(0, presentationSize, trackID);
s->m_pts = pts;
s->m_dts = dts;
s->m_duration = duration;
s->m_flags = MediaSample::NonDisplaying;
return adoptRef(s);
}
virtual ~GStreamerMediaSample()
{
if (m_buffer)
gst_buffer_unref(m_buffer);
}
MediaTime presentationTime() const { return m_pts; }
MediaTime decodeTime() const { return m_dts; }
MediaTime duration() const { return m_duration; }
AtomicString trackID() const { return m_trackID; }
size_t sizeInBytes() const { return m_size; }
GstBuffer* buffer() const { return m_buffer; }
FloatSize presentationSize() const { return m_presentationSize; }
SampleFlags flags() const { return m_flags; }
PlatformSample platformSample() { return PlatformSample(); }
void dump(PrintStream&) const {}
};
};
typedef struct _Stream Stream;
typedef struct _Source Source;
typedef struct {
GstBuffer* buffer;
WebCore::FloatSize presentationSize;
} PendingReceiveSample;
struct _Stream
{
Source* parent;
int id;
StreamType type;
// Might be 0, e.g. for VP8/VP9
GstElement *parser;
GstPad* srcpad;
GstPad* demuxersrcpad;
GstPad* multiqueuesrcpad;
GstCaps* caps;
gulong bufferProbeId;
gulong bufferAfterMultiqueueProbeId;
#if ENABLE(VIDEO_TRACK)
RefPtr<WebCore::AudioTrackPrivateGStreamer> *audioTrack;
RefPtr<WebCore::VideoTrackPrivateGStreamer> *videoTrack;
#endif
WebCore::FloatSize presentationSize;
GList* pendingReceiveSample;
bool initSegmentAlreadyProcessed;
};
struct _Source {
WebKitMediaSrc* parent;
GstElement* src;
GstElement* typefind;
// May be 0 if elementary stream
GstElement* demuxer;
GstElement* multiqueue;
GList* streams;
// We expose everything when
// all sources are noMorePads
bool noMorePads;
// Just for identification
WebCore::SourceBufferPrivateGStreamer* sourceBuffer;
// Some appended data are only headers and don't generate any
// useful stream data for decoding. This is detected with a
// timeout and reported to the upper layers, so update/updateend
// can be generated and the append operation doesn't block.
guint noDataToDecodeTimeoutTag;
// Samples coming after the init segment arrive individually,
// we must detect when no more samples have arrived after a while
gint64 lastSampleTime;
guint pendingSamplesAfterInitSegment;
// To know how to construct the segment that will bind new data coming
// from appendBuffer.
MediaTime nextSamplePts;
};
struct _WebKitMediaSrcPrivate
{
GList* sources;
gchar* location;
int nAudio;
int nVideo;
int nText;
GstClockTime duration;
bool haveAppsrc;
bool asyncStart;
bool noMorePads;
int numberOfPads;
gboolean paused;
gboolean seekable;
guint64 offset;
guint64 requestedOffset;
MediaTime seekTime;
int flushAndReenqueueCount;
GstEvent* seekEvent;
gboolean isAppending;
WebCore::MediaPlayerPrivateGStreamer* mediaPlayerPrivate;
WebCore::MediaSourceClientGStreamer* mediaSourceClient;
};
enum
{
PROP_0,
PROP_LOCATION,
PROP_N_AUDIO,
PROP_N_VIDEO,
PROP_N_TEXT,
PROP_LAST
};
enum
{
SIGNAL_VIDEO_CHANGED,
SIGNAL_AUDIO_CHANGED,
SIGNAL_TEXT_CHANGED,
LAST_SIGNAL
};
static GstStaticPadTemplate srcTemplate = GST_STATIC_PAD_TEMPLATE("src_%u", GST_PAD_SRC,
GST_PAD_SOMETIMES, GST_STATIC_CAPS_ANY);
#define WEBKIT_MEDIA_SRC_GET_PRIVATE(obj) (G_TYPE_INSTANCE_GET_PRIVATE((obj), WEBKIT_TYPE_MEDIA_SRC, WebKitMediaSrcPrivate))
GST_DEBUG_CATEGORY_STATIC(webkit_media_src_debug);
#define GST_CAT_DEFAULT webkit_media_src_debug
static void webKitMediaSrcUriHandlerInit(gpointer gIface, gpointer ifaceData);
static void webKitMediaSrcFinalize(GObject*);
static void webKitMediaSrcSetProperty(GObject*, guint propertyId, const GValue*, GParamSpec*);
static void webKitMediaSrcGetProperty(GObject*, guint propertyId, GValue*, GParamSpec*);
static GstStateChangeReturn webKitMediaSrcChangeState(GstElement*, GstStateChange);
static gboolean webKitMediaSrcQueryWithParent(GstPad*, GstObject*, GstQuery*);
static gboolean webKitMediaSrcEventWithParent(GstPad*, GstObject*, GstEvent*);
static gboolean webKitMediaSrcDemuxerEventWithParent(GstPad*, GstObject*, GstEvent*);
static gboolean webKitMediaSrcSeekDataCb(GstAppSrc*, guint64 offset, gpointer userData);
static Stream* getStreamByDemuxerPad(WebKitMediaSrc* src, const GstPad* demuxersrcpad);
static GstClockTime toGstClockTime(float);
static void webkit_media_src_set_appending(WebKitMediaSrc*, gboolean);
static GstAppSrcCallbacks appsrcCallbacks = {
0,
0,
webKitMediaSrcSeekDataCb,
{ 0 }
};
#define webkit_media_src_parent_class parent_class
// We split this out into another macro to avoid a check-webkit-style error.
#define WEBKIT_MEDIA_SRC_CATEGORY_INIT GST_DEBUG_CATEGORY_INIT(webkit_media_src_debug, "webkitmediasrc", 0, "websrc element");
G_DEFINE_TYPE_WITH_CODE(WebKitMediaSrc, webkit_media_src, GST_TYPE_BIN,
G_IMPLEMENT_INTERFACE(GST_TYPE_URI_HANDLER, webKitMediaSrcUriHandlerInit);
WEBKIT_MEDIA_SRC_CATEGORY_INIT);
static guint webkit_media_src_signals[LAST_SIGNAL] = { 0 };
static void webkit_media_src_class_init(WebKitMediaSrcClass* klass)
{
GObjectClass* oklass = G_OBJECT_CLASS(klass);
GstElementClass* eklass = GST_ELEMENT_CLASS(klass);
oklass->finalize = webKitMediaSrcFinalize;
oklass->set_property = webKitMediaSrcSetProperty;
oklass->get_property = webKitMediaSrcGetProperty;
gst_element_class_add_pad_template(eklass, gst_static_pad_template_get(&srcTemplate));
gst_element_class_set_static_metadata(eklass, "WebKit Media source element", "Source", "Handles Blob uris", "Stephane Jadaud <sjadaud@sii.fr>, Sebastian Dröge <sebastian@centricular.com>");
/* Allows setting the uri using the 'location' property, which is used
* for example by gst_element_make_from_uri() */
g_object_class_install_property(oklass,
PROP_LOCATION,
g_param_spec_string("location", "location", "Location to read from", 0,
(GParamFlags) (G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)));
g_object_class_install_property (oklass,
PROP_N_AUDIO,
g_param_spec_int ("n-audio", "Number Audio", "Total number of audio streams",
0, G_MAXINT, 0, (GParamFlags) (G_PARAM_READABLE | G_PARAM_STATIC_STRINGS)));
g_object_class_install_property (oklass,
PROP_N_VIDEO,
g_param_spec_int ("n-video", "Number Video", "Total number of video streams",
0, G_MAXINT, 0, (GParamFlags) (G_PARAM_READABLE | G_PARAM_STATIC_STRINGS)));
g_object_class_install_property (oklass,
PROP_N_TEXT,
g_param_spec_int ("n-text", "Number Text", "Total number of text streams",
0, G_MAXINT, 0, (GParamFlags) (G_PARAM_READABLE | G_PARAM_STATIC_STRINGS)));
webkit_media_src_signals[SIGNAL_VIDEO_CHANGED] =
g_signal_new ("video-changed", G_TYPE_FROM_CLASS (oklass),
G_SIGNAL_RUN_LAST,
G_STRUCT_OFFSET (WebKitMediaSrcClass, video_changed), NULL, NULL,
g_cclosure_marshal_generic, G_TYPE_NONE, 0, G_TYPE_NONE);
webkit_media_src_signals[SIGNAL_AUDIO_CHANGED] =
g_signal_new ("audio-changed", G_TYPE_FROM_CLASS (oklass),
G_SIGNAL_RUN_LAST,
G_STRUCT_OFFSET (WebKitMediaSrcClass, audio_changed), NULL, NULL,
g_cclosure_marshal_generic, G_TYPE_NONE, 0, G_TYPE_NONE);
webkit_media_src_signals[SIGNAL_TEXT_CHANGED] =
g_signal_new ("text-changed", G_TYPE_FROM_CLASS (oklass),
G_SIGNAL_RUN_LAST,
G_STRUCT_OFFSET (WebKitMediaSrcClass, text_changed), NULL, NULL,
g_cclosure_marshal_generic, G_TYPE_NONE, 0, G_TYPE_NONE);
eklass->change_state = webKitMediaSrcChangeState;
g_type_class_add_private(klass, sizeof(WebKitMediaSrcPrivate));
}
static void webkit_media_src_init(WebKitMediaSrc* src)
{
src->priv = WEBKIT_MEDIA_SRC_GET_PRIVATE(src);
src->priv->seekTime = MediaTime::invalidTime();
src->priv->isAppending = FALSE;
}
static void webKitMediaSrcFinalize(GObject* object)
{
WebKitMediaSrc* src = WEBKIT_MEDIA_SRC(object);
WebKitMediaSrcPrivate* priv = src->priv;
// TODO: Free sources
g_free(priv->location);
priv->seekTime = MediaTime::invalidTime();
if (priv->seekEvent) {
gst_event_unref(priv->seekEvent);
priv->seekEvent = 0;
}
if (priv->mediaPlayerPrivate)
priv->mediaPlayerPrivate = 0;
GST_CALL_PARENT(G_OBJECT_CLASS, finalize, (object));
}
static void webKitMediaSrcSetProperty(GObject* object, guint propId, const GValue* value, GParamSpec* pspec)
{
WebKitMediaSrc* src = WEBKIT_MEDIA_SRC(object);
switch (propId) {
case PROP_LOCATION:
gst_uri_handler_set_uri(reinterpret_cast<GstURIHandler*>(src), g_value_get_string(value), 0);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID(object, propId, pspec);
break;
}
}
static void webKitMediaSrcGetProperty(GObject* object, guint propId, GValue* value, GParamSpec* pspec)
{
WebKitMediaSrc* src = WEBKIT_MEDIA_SRC(object);
WebKitMediaSrcPrivate* priv = src->priv;
GST_OBJECT_LOCK(src);
switch (propId) {
case PROP_LOCATION:
g_value_set_string(value, priv->location);
break;
case PROP_N_AUDIO:
g_value_set_int(value, priv->nAudio);
break;
case PROP_N_VIDEO:
g_value_set_int(value, priv->nVideo);
break;
case PROP_N_TEXT:
g_value_set_int(value, priv->nText);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID(object, propId, pspec);
break;
}
GST_OBJECT_UNLOCK(src);
}
static void webKitMediaSrcDoAsyncStart(WebKitMediaSrc* src)
{
WebKitMediaSrcPrivate* priv = src->priv;
priv->asyncStart = true;
GST_BIN_CLASS(parent_class)->handle_message(GST_BIN(src),
gst_message_new_async_start(GST_OBJECT(src)));
}
static void webKitMediaSrcDoAsyncDone(WebKitMediaSrc* src)
{
WebKitMediaSrcPrivate* priv = src->priv;
if (priv->asyncStart) {
GST_BIN_CLASS(parent_class)->handle_message(GST_BIN(src),
gst_message_new_async_done(GST_OBJECT(src), GST_CLOCK_TIME_NONE));
priv->asyncStart = false;
}
}
static GstStateChangeReturn webKitMediaSrcChangeState(GstElement* element, GstStateChange transition)
{
GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
WebKitMediaSrc* src = WEBKIT_MEDIA_SRC(element);
WebKitMediaSrcPrivate* priv = src->priv;
switch (transition) {
case GST_STATE_CHANGE_READY_TO_PAUSED:
priv->noMorePads = false;
webKitMediaSrcDoAsyncStart(src);
break;
default:
break;
}
ret = GST_ELEMENT_CLASS(parent_class)->change_state(element, transition);
if (G_UNLIKELY(ret == GST_STATE_CHANGE_FAILURE)) {
GST_DEBUG_OBJECT(src, "State change failed");
webKitMediaSrcDoAsyncDone(src);
return ret;
}
switch (transition) {
case GST_STATE_CHANGE_READY_TO_PAUSED:
ret = GST_STATE_CHANGE_ASYNC;
break;
case GST_STATE_CHANGE_PAUSED_TO_READY:
webKitMediaSrcDoAsyncDone(src);
priv->noMorePads = false;
break;
default:
break;
}
return ret;
}
static gboolean webKitMediaSrcQueryWithParent(GstPad* pad, GstObject* parent, GstQuery* query)
{
WebKitMediaSrc* src = WEBKIT_MEDIA_SRC(GST_ELEMENT(parent));
gboolean result = FALSE;
switch (GST_QUERY_TYPE(query)) {
case GST_QUERY_DURATION: {
GstFormat format;
gst_query_parse_duration(query, &format, NULL);
GST_DEBUG_OBJECT(src, "duration query in format %s", gst_format_get_name(format));
GST_OBJECT_LOCK(src);
if ((format == GST_FORMAT_TIME) && src && src->priv && (src->priv->duration > 0)) {
gst_query_set_duration(query, format, src->priv->duration);
result = TRUE;
}
GST_OBJECT_UNLOCK(src);
break;
}
case GST_QUERY_URI:
GST_OBJECT_LOCK(src);
if (src && src->priv)
gst_query_set_uri(query, src->priv->location);
GST_OBJECT_UNLOCK(src);
result = TRUE;
break;
default:{
GRefPtr<GstPad> target = adoptGRef(gst_ghost_pad_get_target(GST_GHOST_PAD_CAST(pad)));
// Forward the query to the proxy target pad.
if (target)
result = gst_pad_query(target.get(), query);
break;
}
}
return result;
}
#if ENABLE(VIDEO_TRACK)
static gboolean webKitMediaSrcDidReceiveInitializationSegment(gpointer data);
#endif
static gboolean webKitMediaSrcLastSampleTimeout(Source* source);
static gboolean webKitMediaSrcEventWithParent(GstPad* pad, GstObject* parent, GstEvent* event)
{
gboolean result = FALSE;
switch (GST_EVENT_TYPE(event)) {
#if ENABLE(VIDEO_TRACK)
case GST_EVENT_CUSTOM_UPSTREAM:{
const GstStructure* s = gst_event_get_structure(event);
if (gst_structure_has_name(s, "webKitAudioTrack")) {
Stream* stream = (Stream*) gst_pad_get_element_private(pad);
if (stream && stream->parent) {
RefPtr<WebCore::AudioTrackPrivateGStreamer>* audioTrack;
gst_structure_get(s, "track", G_TYPE_POINTER, &audioTrack, NULL);
stream->audioTrack = audioTrack;
g_timeout_add(0, GSourceFunc(webKitMediaSrcDidReceiveInitializationSegment), stream->parent);
}
result = TRUE;
gst_event_unref(event);
} else if (gst_structure_has_name(s, "webKitVideoTrack")) {
Stream* stream = (Stream*) gst_pad_get_element_private(pad);
if (stream && stream->parent) {
RefPtr<WebCore::VideoTrackPrivateGStreamer>* videoTrack;
gst_structure_get(s, "track", G_TYPE_POINTER, &videoTrack, NULL);
stream->videoTrack = videoTrack;
g_timeout_add(0, GSourceFunc(webKitMediaSrcDidReceiveInitializationSegment), stream->parent);
}
result = TRUE;
gst_event_unref(event);
} else {
result = gst_pad_event_default(pad, parent, event);
}
break;
}
#endif
default:
result = gst_pad_event_default(pad, parent, event);
break;
}
return result;
}
static gboolean webKitMediaSrcDemuxerSinkEventWithParent(GstPad* pad, GstObject* parent, GstEvent* event)
{
gboolean result = FALSE;
switch (GST_EVENT_TYPE(event)) {
case GST_EVENT_FLUSH_STOP: {
result = gst_pad_event_default(pad, parent, event);
if (!result)
break;
WebKitMediaSrc* src = NULL;
MediaTime nextSamplePts = MediaTime::invalidTime();
if (parent)
src = WEBKIT_MEDIA_SRC(GST_ELEMENT_PARENT(GST_ELEMENT(parent)));
if (src) {
GST_OBJECT_LOCK(src);
Source* source = NULL;
for (GList* sources = src->priv->sources; sources; sources = sources->next) {
Source* s = static_cast<Source*>(sources->data);
if (s->demuxer == GST_ELEMENT(parent)) {
source = s;
break;
}
}
if (source)
nextSamplePts = source->nextSamplePts;
GST_OBJECT_UNLOCK(src);
}
if (nextSamplePts.isInvalid())
nextSamplePts = MediaTime::createWithDouble(0);
// Insert a [nextSamplePts, Inf] segment to accomodate the new data
// and prevent a spureous [0, 0] segment to be generated by the demuxer.
GstSegment* segment = gst_segment_new();
gst_segment_init(segment, GST_FORMAT_TIME);
segment->start = toGstClockTime(nextSamplePts.toFloat());
segment->stop = GST_CLOCK_TIME_NONE;
gst_element_send_event(GST_ELEMENT(parent), gst_event_new_segment(segment));
gst_segment_free(segment);
break;
}
default:
result = gst_pad_event_default(pad, parent, event);
break;
}
return result;
}
static gboolean webKitMediaSrcDemuxerEventWithParent(GstPad* pad, GstObject* parent, GstEvent* event)
{
gboolean result = FALSE;
switch (GST_EVENT_TYPE(event)) {
case GST_EVENT_SEEK: {
gdouble rate;
GstFormat format;
GstSeekFlags flags;
GstSeekType start_type;
gint64 start;
GstSeekType stop_type;
gint64 stop;
gst_event_parse_seek(event, &rate, &format, &flags, &start_type, &start, &stop_type, &stop);
WebKitMediaSrc* src = NULL;
if (parent)
src = WEBKIT_MEDIA_SRC(GST_ELEMENT_PARENT(GST_ELEMENT(parent)));
if (src) {
GST_OBJECT_LOCK(src);
if (format == GST_FORMAT_TIME) {
gst_event_ref(event);
if (src->priv->seekEvent)
gst_event_unref(src->priv->seekEvent);
src->priv->seekEvent = event;
}
GST_OBJECT_UNLOCK(src);
}
// No break, will fall back to the "default" case on purpose
}
default:
result = gst_pad_event_default(pad, parent, event);
break;
}
return result;
}
static GstPad* get_internal_linked_pad(GstPad* pad)
{
GstIterator* it;
GstPad* otherpad;
GValue item = G_VALUE_INIT;
it = gst_pad_iterate_internal_links(pad);
if (!it || (gst_iterator_next(it, &item)) != GST_ITERATOR_OK
|| ((otherpad = GST_PAD(g_value_dup_object(&item))) == 0)) {
return 0;
}
g_value_unset(&item);
if (it)
gst_iterator_free(it);
return otherpad;
}
typedef struct {
Stream* stream;
WTF::RefPtr<WebCore::GStreamerMediaSample> sample;
} ReceiveSample;
#if ENABLE(VIDEO_TRACK)
static gboolean webKitWebSrcDidReceiveSample(gpointer userdata)
{
ReceiveSample* sample = (ReceiveSample*)userdata;
if (sample->stream->parent) {
MediaTime timestampOffset(MediaTime::createWithDouble(sample->stream->parent->sourceBuffer->timestampOffset()));
// Add a fake sample if a gap is detected before the first sample
if (sample->sample->presentationTime() >= timestampOffset &&
sample->sample->presentationTime() <= timestampOffset + MediaTime::createWithDouble(0.1)) {
RefPtr<WebCore::GStreamerMediaSample> fakeSample = WebCore::GStreamerMediaSample::createFakeSample(
timestampOffset, sample->sample->decodeTime(), sample->sample->presentationTime() - timestampOffset, sample->sample->presentationSize(),
sample->stream->audioTrack ? sample->stream->audioTrack->get()->id() : sample->stream->videoTrack->get()->id());
sample->stream->parent->parent->priv->mediaSourceClient->didReceiveSample(sample->stream->parent->sourceBuffer, fakeSample);
}
sample->stream->parent->parent->priv->mediaSourceClient->didReceiveSample(sample->stream->parent->sourceBuffer, sample->sample);
GST_OBJECT_LOCK(sample->stream->parent->parent);
sample->stream->parent->pendingSamplesAfterInitSegment--;
if (!sample->stream->parent->lastSampleTime) {
g_timeout_add(100, GSourceFunc(webKitMediaSrcLastSampleTimeout), sample->stream->parent);
}
sample->stream->parent->lastSampleTime = g_get_monotonic_time();
sample->stream->parent->nextSamplePts = sample->sample->presentationTime() + sample->sample->duration();
GST_OBJECT_UNLOCK(sample->stream->parent->parent);
}
sample->sample.release();
g_free(sample);
return G_SOURCE_REMOVE;
}
#endif
static gboolean webKitMediaSrcLastSampleTimeout(Source* source)
{
bool result;
bool callDidReceiveAllPendingSamples = false;
GST_OBJECT_LOCK(source->parent);
if (source->lastSampleTime) {
if (source->pendingSamplesAfterInitSegment == 0 && g_get_monotonic_time() - source->lastSampleTime > 250) {
source->lastSampleTime = 0;
callDidReceiveAllPendingSamples = true;
result = G_SOURCE_REMOVE;
} else
result = G_SOURCE_CONTINUE;
} else {
// The timer has been cancelled
result = G_SOURCE_REMOVE;
}
if (callDidReceiveAllPendingSamples)
webkit_media_src_set_appending(source->parent, FALSE);
GST_OBJECT_UNLOCK(source->parent);
if (callDidReceiveAllPendingSamples) {
source->parent->priv->mediaSourceClient->didReceiveAllPendingSamples(source->sourceBuffer);
}
return result;
}
static GstPadProbeReturn webKitWebSrcBufferProbe(GstPad*, GstPadProbeInfo* info, Stream* stream)
{
if (!(stream->parent)) return GST_PAD_PROBE_DROP;
GstBuffer* buffer = GST_BUFFER(info->data);
GST_OBJECT_LOCK(stream->parent->parent);
if (stream->parent->noDataToDecodeTimeoutTag) {
g_source_remove(stream->parent->noDataToDecodeTimeoutTag);
stream->parent->noDataToDecodeTimeoutTag = 0;
}
bool initSegmentAlreadyProcessed = stream->initSegmentAlreadyProcessed;
if (initSegmentAlreadyProcessed) {
ReceiveSample* sample = g_new0(ReceiveSample, 1);
sample->sample = WebCore::GStreamerMediaSample::create(buffer, stream->presentationSize, stream->audioTrack ? stream->audioTrack->get()->id() : stream->videoTrack->get()->id());
sample->stream = stream;
stream->parent->pendingSamplesAfterInitSegment++;
g_timeout_add(0, GSourceFunc(webKitWebSrcDidReceiveSample), sample);
} else {
PendingReceiveSample* sample = g_new0(PendingReceiveSample, 1);
sample->buffer = gst_buffer_ref(buffer);
sample->presentationSize = stream->presentationSize;
stream->pendingReceiveSample = g_list_append(stream->pendingReceiveSample, sample);
}
GST_OBJECT_UNLOCK(stream->parent->parent);
// This probe DROPS all the buffers. They will be reinserted in the
// pipeline by flushAndEnqueueNonDisplayingSamples() and enqueueSamples().
return GST_PAD_PROBE_DROP;
}
static GstPadProbeReturn webKitWebSrcBufferAfterMultiqueueProbe(GstPad* pad, GstPadProbeInfo* info, Stream* stream)
{
if (!(stream->parent)) return GST_PAD_PROBE_DROP;
GstPadProbeReturn result;
GstBuffer* buffer = GST_BUFFER(info->data);
GST_OBJECT_LOCK(stream->parent->parent);
GstClockTime duration = stream->parent->parent->priv->duration;
GST_OBJECT_UNLOCK(stream->parent->parent);
// If the presentation time of this buffer is beyond the "logical" duration, synthesize EOS.
// The "logical" duration may be shorter than the "physical" duration that the buffered data can provide,
// which would throw a natural EOS anyway
if (GST_BUFFER_PTS_IS_VALID(buffer) && duration && GST_BUFFER_PTS(buffer) > duration) {
GRefPtr<GstPad> peerPad = adoptGRef(gst_pad_get_peer(pad));
gst_pad_send_event(peerPad.get(), gst_event_new_eos());
result = GST_PAD_PROBE_DROP;
} else
result = GST_PAD_PROBE_OK;
return result;
}
static void webKitMediaSrcDemuxerNoMorePads(GstElement*, Source* source);
static void webKitMediaSrcUpdatePresentationSize(GstCaps* caps, Stream* stream)
{
GstStructure* s = gst_caps_get_structure(caps, 0);
const gchar* structureName = gst_structure_get_name(s);
GstVideoInfo info;
if (g_str_has_prefix(structureName, "video/") && gst_video_info_from_caps(&info, caps)) {
float width, height;
// TODO: correct?
width = info.width;
height = info.height * ((float) info.par_d / (float) info.par_n);
GST_OBJECT_LOCK(stream->parent->parent);
stream->presentationSize = WebCore::FloatSize(width, height);
GST_OBJECT_UNLOCK(stream->parent->parent);
} else {
GST_OBJECT_LOCK(stream->parent->parent);
stream->presentationSize = WebCore::FloatSize();
GST_OBJECT_UNLOCK(stream->parent->parent);
}
gst_caps_ref(caps);
GST_OBJECT_LOCK(stream->parent->parent);
if (stream->caps)
gst_caps_unref(stream->caps);
stream->caps = caps;
GST_OBJECT_UNLOCK(stream->parent->parent);
}
static void webKitMediaSrcLinkStreamToSrcPad(GstPad* srcpad, Stream* stream)
{
Source* source = stream->parent;
// TODO: Atomic ints, GRefPtr
GST_OBJECT_LOCK(stream->parent->parent);
int numberOfPads = source->parent->priv->numberOfPads++;
GST_OBJECT_UNLOCK(stream->parent->parent);
gchar* padName = g_strdup_printf("src_%u", numberOfPads);
GstPad* ghostpad = gst_ghost_pad_new_from_template(padName, srcpad, gst_static_pad_template_get(&srcTemplate));
gst_pad_set_query_function(ghostpad, webKitMediaSrcQueryWithParent);
gst_pad_set_event_function(ghostpad, webKitMediaSrcEventWithParent);
gst_pad_set_element_private(ghostpad, stream);
gst_pad_set_active(ghostpad, TRUE);
gst_element_add_pad(GST_ELEMENT(source->parent), ghostpad);
GST_OBJECT_LOCK(stream->parent->parent);
stream->srcpad = ghostpad;
GST_OBJECT_UNLOCK(stream->parent->parent);
}
static gboolean webKitMediaSrcNoDataToDecodeTimeout(Source* source)
{
GST_OBJECT_LOCK(source->parent);
source->noDataToDecodeTimeoutTag = 0;
webkit_media_src_set_appending(source->parent, FALSE);
GST_OBJECT_UNLOCK(source->parent);
source->parent->priv->mediaSourceClient->didReceiveAllPendingSamples(source->sourceBuffer);
return FALSE;
}
static void webKitMediaSrcParserNotifyCaps(GObject* object, GParamSpec*, Stream* stream)
{
GstPad* srcpad = GST_PAD(object);
GstCaps* caps = gst_pad_get_current_caps(srcpad);
if (!caps || !stream->parent) {
return;
}
GST_OBJECT_LOCK(stream->parent->parent);
if (stream->parent->noDataToDecodeTimeoutTag) {
g_source_remove(stream->parent->noDataToDecodeTimeoutTag);
stream->parent->noDataToDecodeTimeoutTag = 0;
}
GST_OBJECT_UNLOCK(stream->parent->parent);
webKitMediaSrcUpdatePresentationSize(caps, stream);
gst_caps_unref(caps);
// TODO
if (!gst_pad_is_linked(srcpad))
webKitMediaSrcLinkStreamToSrcPad(srcpad, stream);
webKitMediaSrcDemuxerNoMorePads(NULL, stream->parent);
}
static void webKitMediaSrcDemuxerPadAdded(GstElement*, GstPad* demuxersrcpad, Source* source)
{
GstCaps* demuxersrcpadcaps = gst_pad_get_current_caps(demuxersrcpad);
GstStructure* s = gst_caps_get_structure(demuxersrcpadcaps, 0);
Stream* stream = g_new0(Stream, 1);
gchar *parserBinName;
const gchar* demuxersrcpadtypename = gst_structure_get_name(s);
GST_OBJECT_LOCK(source->parent);
stream->id = source->parent->priv->numberOfPads; // Just informative
GST_OBJECT_UNLOCK(source->parent);
stream->parent = source;
stream->initSegmentAlreadyProcessed = false;
stream->type = STREAM_TYPE_UNKNOWN;
stream->demuxersrcpad = demuxersrcpad;
parserBinName = g_strdup_printf("streamparser%d", stream->id);
g_assert(demuxersrcpadcaps != 0);
if (gst_structure_has_name(s, "video/x-h264")) {
GstElement* parser;
GstElement* capsfilter;
GstPad* pad;
GstCaps* filtercaps;
filtercaps = gst_caps_new_simple("video/x-h264", "alignment", G_TYPE_STRING, "au", NULL);
parser = gst_element_factory_make("h264parse", 0);
capsfilter = gst_element_factory_make("capsfilter", 0);
g_object_set(capsfilter, "caps", filtercaps, NULL);
gst_caps_unref(filtercaps);
stream->parser = gst_bin_new(parserBinName);
gst_bin_add_many(GST_BIN(stream->parser), parser, capsfilter, NULL);
gst_element_link_pads(parser, "src", capsfilter, "sink");
pad = gst_element_get_static_pad(parser, "sink");
gst_element_add_pad(stream->parser, gst_ghost_pad_new("sink", pad));
gst_object_unref(pad);
pad = gst_element_get_static_pad(capsfilter, "src");
gst_element_add_pad(stream->parser, gst_ghost_pad_new("src", pad));
gst_object_unref(pad);
} else if (gst_structure_has_name(s, "video/x-h265")) {
GstElement* parser;
GstElement* capsfilter;
GstPad* pad;
GstCaps* filtercaps;
filtercaps = gst_caps_new_simple("video/x-h265", "alignment", G_TYPE_STRING, "au", NULL);
parser = gst_element_factory_make("h265parse", 0);
capsfilter = gst_element_factory_make("capsfilter", 0);
g_object_set(capsfilter, "caps", filtercaps, NULL);
gst_caps_unref(filtercaps);
stream->parser = gst_bin_new(parserBinName);
gst_bin_add_many(GST_BIN(stream->parser), parser, capsfilter, NULL);
gst_element_link_pads(parser, "src", capsfilter, "sink");
pad = gst_element_get_static_pad(parser, "sink");
gst_element_add_pad(stream->parser, gst_ghost_pad_new("sink", pad));
gst_object_unref(pad);
pad = gst_element_get_static_pad(capsfilter, "src");
gst_element_add_pad(stream->parser, gst_ghost_pad_new("src", pad));
gst_object_unref(pad);
} else if (gst_structure_has_name(s, "audio/mpeg")) {
gint mpegversion = -1;