Diff
Modified: trunk/LayoutTests/ChangeLog (238950 => 238951)
--- trunk/LayoutTests/ChangeLog 2018-12-07 06:18:33 UTC (rev 238950)
+++ trunk/LayoutTests/ChangeLog 2018-12-07 10:48:56 UTC (rev 238951)
@@ -1,3 +1,15 @@
+2018-12-07 Thibault Saunier <[email protected]>
+
+ [WPE][GTK] Implement WebAudioSourceProviderGStreamer to allow bridging MediaStream and the WebAudio APIs
+ https://bugs.webkit.org/show_bug.cgi?id=186933
+
+ Reviewed by Philippe Normand.
+
+ Enabled all tests depending on that feature.
+
+ * platform/gtk/TestExpectations:
+ * webrtc/clone-audio-track.html:
+
2018-12-06 Yongjun Zhang <[email protected]>
We should ignore minimumEffectiveDeviceWidth if the page specifies device-width in viewport meta-tag.
Modified: trunk/LayoutTests/platform/gtk/TestExpectations (238950 => 238951)
--- trunk/LayoutTests/platform/gtk/TestExpectations 2018-12-07 06:18:33 UTC (rev 238950)
+++ trunk/LayoutTests/platform/gtk/TestExpectations 2018-12-07 10:48:56 UTC (rev 238951)
@@ -586,16 +586,6 @@
webkit.org/b/177533 webrtc/video-interruption.html
webkit.org/b/186933 webrtc/peer-connection-createMediaStreamDestination.html
-webkit.org/b/186933 webrtc/peer-connection-remote-audio-mute2.html
-webkit.org/b/186933 webrtc/peer-connection-audio-unmute.html
-webkit.org/b/186933 webrtc/peer-connection-audio-mute2.html
-webkit.org/b/186933 webrtc/peer-connection-audio-mute.html
-webkit.org/b/186933 webrtc/peer-connection-remote-audio-mute.html
-webkit.org/b/186933 webrtc/clone-audio-track.html
-webkit.org/b/186933 webrtc/audio-replace-track.html
-webkit.org/b/186933 webrtc/audio-peer-connection-webaudio.html
-webkit.org/b/186933 webrtc/audio-muted-stats.html
-webkit.org/b/186933 webrtc/getUserMedia-webaudio-autoplay.html
imported/w3c/web-platform-tests/webrtc/ [ Skip ]
http/tests/webrtc [ Skip ]
Modified: trunk/LayoutTests/webrtc/clone-audio-track.html (238950 => 238951)
--- trunk/LayoutTests/webrtc/clone-audio-track.html 2018-12-07 06:18:33 UTC (rev 238950)
+++ trunk/LayoutTests/webrtc/clone-audio-track.html 2018-12-07 10:48:56 UTC (rev 238951)
@@ -33,7 +33,7 @@
});
}).then(() => {
return analyseAudio(remoteStream, 200, context).then((results) => {
- assert_false(results.heardHum, "Did not heard hum from remote enabled track");
+ assert_false(results.heardHum, "Did not hear hum from remote disabled track");
});
}).then(() => {
return analyseAudio(new MediaStream([clonedTrack]), 200, context).then((results) => {
Modified: trunk/Source/WebCore/ChangeLog (238950 => 238951)
--- trunk/Source/WebCore/ChangeLog 2018-12-07 06:18:33 UTC (rev 238950)
+++ trunk/Source/WebCore/ChangeLog 2018-12-07 10:48:56 UTC (rev 238951)
@@ -1,3 +1,47 @@
+2018-12-07 Thibault Saunier <[email protected]>
+
+ [WPE][GTK] Implement WebAudioSourceProviderGStreamer to allow bridging MediaStream and the WebAudio APIs
+ https://bugs.webkit.org/show_bug.cgi?id=186933
+
+ Reusing the AudioSourceProviderGStreamer itself as it was doing almost everything we needed,
+ just added a constructor to be able to create it from a MediaStreamTrackPrivate and made it a
+ WebAudioSourceProvider which only means it is now a ThreadSafeRefCounted.
+
+ Sensibily refactored GStreamerMediaStreamSource so that we could reuse it to track a single
+ MediaStreamTrack.
+
+ Reviewed by Philippe Normand.
+
+ Enabled all tests depending on that feature.
+
+ * platform/audio/gstreamer/AudioSourceProviderGStreamer.cpp:
+ (WebCore::AudioSourceProviderGStreamer::AudioSourceProviderGStreamer):
+ (WebCore::AudioSourceProviderGStreamer::~AudioSourceProviderGStreamer):
+ (WebCore::AudioSourceProviderGStreamer::setClient):
+ * platform/audio/gstreamer/AudioSourceProviderGStreamer.h:
+ * platform/mediastream/MediaStreamTrackPrivate.cpp:
+ (WebCore::MediaStreamTrackPrivate::audioSourceProvider):
+ * platform/mediastream/gstreamer/GStreamerAudioCapturer.cpp:
+ (WebCore::GStreamerAudioCapturer::GStreamerAudioCapturer):
+ * platform/mediastream/gstreamer/GStreamerAudioStreamDescription.h:
+ * platform/mediastream/gstreamer/GStreamerMediaStreamSource.cpp:
+ (WebCore::webkitMediaStreamSrcSetupSrc):
+ (WebCore::webkitMediaStreamSrcSetupAppSrc):
+ (WebCore::webkitMediaStreamSrcAddTrack):
+ (WebCore::webkitMediaStreamSrcSetStream):
+ (WebCore::webkitMediaStreamSrcNew):
+ * platform/mediastream/gstreamer/GStreamerMediaStreamSource.h:
+ * platform/mediastream/gstreamer/MockGStreamerAudioCaptureSource.cpp:
+ (WebCore::WrappedMockRealtimeAudioSource::WrappedMockRealtimeAudioSource):
+ (WebCore::WrappedMockRealtimeAudioSource::start):
+ (WebCore::WrappedMockRealtimeAudioSource::addHum):
+ (WebCore::WrappedMockRealtimeAudioSource::render):
+ (WebCore::WrappedMockRealtimeAudioSource::settingsDidChange):
+ (WebCore::MockGStreamerAudioCaptureSource::startProducingData):
+ * platform/mediastream/gstreamer/RealtimeOutgoingAudioSourceLibWebRTC.cpp:
+ (WebCore::RealtimeOutgoingAudioSourceLibWebRTC::pullAudioData): Handle the case where input buffers
+ are "big" and process all the data we can for each runs of the method.
+
2018-12-06 Alexey Proskuryakov <[email protected]>
Move USE_NEW_THEME out of WebCore's config.h
Modified: trunk/Source/WebCore/platform/audio/gstreamer/AudioSourceProviderGStreamer.cpp (238950 => 238951)
--- trunk/Source/WebCore/platform/audio/gstreamer/AudioSourceProviderGStreamer.cpp 2018-12-07 06:18:33 UTC (rev 238950)
+++ trunk/Source/WebCore/platform/audio/gstreamer/AudioSourceProviderGStreamer.cpp 2018-12-07 10:48:56 UTC (rev 238951)
@@ -27,6 +27,10 @@
#include <gst/audio/audio-info.h>
#include <gst/base/gstadapter.h>
+#if ENABLE(MEDIA_STREAM) && USE(LIBWEBRTC)
+#include "GStreamerAudioData.h"
+#include "GStreamerMediaStreamSource.h"
+#endif
namespace WebCore {
@@ -94,6 +98,31 @@
m_frontRightAdapter = gst_adapter_new();
}
+#if ENABLE(MEDIA_STREAM) && USE(LIBWEBRTC)
+AudioSourceProviderGStreamer::AudioSourceProviderGStreamer(MediaStreamTrackPrivate& source)
+ : m_notifier(MainThreadNotifier<MainThreadNotification>::create())
+ , m_client(nullptr)
+ , m_deinterleaveSourcePads(0)
+ , m_deinterleavePadAddedHandlerId(0)
+ , m_deinterleaveNoMorePadsHandlerId(0)
+ , m_deinterleavePadRemovedHandlerId(0)
+{
+ m_frontLeftAdapter = gst_adapter_new();
+ m_frontRightAdapter = gst_adapter_new();
+ auto pipelineName = String::format("WebAudioProvider_MediaStreamTrack_%s", source.id().utf8().data());
+ m_pipeline = adoptGRef(GST_ELEMENT(g_object_ref_sink(gst_element_factory_make("pipeline", pipelineName.utf8().data()))));
+ auto src = ""
+ webkitMediaStreamSrcAddTrack(WEBKIT_MEDIA_STREAM_SRC(src), &source, true);
+
+ m_audioSinkBin = adoptGRef(GST_ELEMENT(g_object_ref_sink(gst_parse_bin_from_description("tee name=audioTee", true, nullptr))));
+
+ gst_bin_add_many(GST_BIN(m_pipeline.get()), src, m_audioSinkBin.get(), nullptr);
+ gst_element_link(src, m_audioSinkBin.get());
+
+ connectSimpleBusMessageCallback(m_pipeline.get());
+}
+#endif
+
AudioSourceProviderGStreamer::~AudioSourceProviderGStreamer()
{
m_notifier->invalidate();
@@ -105,6 +134,9 @@
g_signal_handler_disconnect(deinterleave.get(), m_deinterleavePadRemovedHandlerId);
}
+ if (m_pipeline)
+ gst_element_set_state(m_pipeline.get(), GST_STATE_NULL);
+
g_object_unref(m_frontLeftAdapter);
g_object_unref(m_frontRightAdapter);
}
@@ -205,13 +237,18 @@
ASSERT(client);
m_client = client;
+ if (m_pipeline)
+ gst_element_set_state(m_pipeline.get(), GST_STATE_PLAYING);
+
// The volume element is used to mute audio playback towards the
// autoaudiosink. This is needed to avoid double playback of audio
// from our audio sink and from the WebAudio AudioDestination node
// supposedly configured already by application side.
GRefPtr<GstElement> volumeElement = adoptGRef(gst_bin_get_by_name(GST_BIN(m_audioSinkBin.get()), "volume"));
- g_object_set(volumeElement.get(), "mute", TRUE, nullptr);
+ if (volumeElement)
+ g_object_set(volumeElement.get(), "mute", TRUE, nullptr);
+
// The audioconvert and audioresample elements are needed to
// ensure deinterleave and the sinks downstream receive buffers in
// the format specified by the capsfilter.
Modified: trunk/Source/WebCore/platform/audio/gstreamer/AudioSourceProviderGStreamer.h (238950 => 238951)
--- trunk/Source/WebCore/platform/audio/gstreamer/AudioSourceProviderGStreamer.h 2018-12-07 06:18:33 UTC (rev 238950)
+++ trunk/Source/WebCore/platform/audio/gstreamer/AudioSourceProviderGStreamer.h 2018-12-07 10:48:56 UTC (rev 238951)
@@ -28,14 +28,31 @@
#include <wtf/Forward.h>
#include <wtf/Noncopyable.h>
+#if ENABLE(MEDIA_STREAM) && USE(LIBWEBRTC)
+#include "GStreamerAudioStreamDescription.h"
+#include "MediaStreamTrackPrivate.h"
+#include "WebAudioSourceProvider.h"
+#endif
+
typedef struct _GstAdapter GstAdapter;
typedef struct _GstAppSink GstAppSink;
namespace WebCore {
+#if ENABLE(MEDIA_STREAM) && USE(LIBWEBRTC)
+class AudioSourceProviderGStreamer final : public WebAudioSourceProvider {
+public:
+ static Ref<AudioSourceProviderGStreamer> create(MediaStreamTrackPrivate& source)
+ {
+ return adoptRef(*new AudioSourceProviderGStreamer(source));
+ }
+ AudioSourceProviderGStreamer(MediaStreamTrackPrivate&);
+#else
class AudioSourceProviderGStreamer : public AudioSourceProvider {
WTF_MAKE_NONCOPYABLE(AudioSourceProviderGStreamer);
public:
+#endif
+
AudioSourceProviderGStreamer();
~AudioSourceProviderGStreamer();
@@ -54,6 +71,7 @@
void clearAdapters();
private:
+ GRefPtr<GstElement> m_pipeline;
enum MainThreadNotification {
DeinterleavePadsConfigured = 1 << 0,
};
Modified: trunk/Source/WebCore/platform/mediastream/MediaStreamTrackPrivate.cpp (238950 => 238951)
--- trunk/Source/WebCore/platform/mediastream/MediaStreamTrackPrivate.cpp 2018-12-07 06:18:33 UTC (rev 238950)
+++ trunk/Source/WebCore/platform/mediastream/MediaStreamTrackPrivate.cpp 2018-12-07 10:48:56 UTC (rev 238951)
@@ -35,6 +35,8 @@
#if PLATFORM(COCOA)
#include "WebAudioSourceProviderAVFObjC.h"
+#elif ENABLE(WEB_AUDIO) && ENABLE(MEDIA_STREAM) && USE(LIBWEBRTC) && USE(GSTREAMER)
+#include "AudioSourceProviderGStreamer.h"
#else
#include "WebAudioSourceProvider.h"
#endif
@@ -178,6 +180,9 @@
#if PLATFORM(COCOA)
if (!m_audioSourceProvider)
m_audioSourceProvider = WebAudioSourceProviderAVFObjC::create(*this);
+#elif USE(LIBWEBRTC) && USE(GSTREAMER)
+ if (!m_audioSourceProvider)
+ m_audioSourceProvider = AudioSourceProviderGStreamer::create(*this);
#endif
return m_audioSourceProvider.get();
}
Modified: trunk/Source/WebCore/platform/mediastream/gstreamer/GStreamerAudioCapturer.cpp (238950 => 238951)
--- trunk/Source/WebCore/platform/mediastream/gstreamer/GStreamerAudioCapturer.cpp 2018-12-07 06:18:33 UTC (rev 238950)
+++ trunk/Source/WebCore/platform/mediastream/gstreamer/GStreamerAudioCapturer.cpp 2018-12-07 10:48:56 UTC (rev 238951)
@@ -36,20 +36,10 @@
}
GStreamerAudioCapturer::GStreamerAudioCapturer()
- : GStreamerCapturer("audiotestsrc", adoptGRef(gst_caps_new_simple("audio/x-raw", "rate", G_TYPE_INT, LibWebRTCAudioFormat::sampleRate, nullptr)))
+ : GStreamerCapturer("appsrc", adoptGRef(gst_caps_new_simple("audio/x-raw", "rate", G_TYPE_INT, LibWebRTCAudioFormat::sampleRate, nullptr)))
{
}
-GstElement* GStreamerAudioCapturer::createSource()
-{
- GstElement* source = GStreamerCapturer::createSource();
-
- if (!m_device)
- gst_util_set_object_arg(G_OBJECT(m_src.get()), "wave", "ticks");
-
- return source;
-}
-
GstElement* GStreamerAudioCapturer::createConverter()
{
auto converter = gst_parse_bin_from_description("audioconvert ! audioresample", TRUE, nullptr);
@@ -62,14 +52,15 @@
bool GStreamerAudioCapturer::setSampleRate(int sampleRate)
{
- if (sampleRate > 0) {
- GST_INFO_OBJECT(m_pipeline.get(), "Setting SampleRate %d", sampleRate);
- m_caps = adoptGRef(gst_caps_new_simple("audio/x-raw", "rate", G_TYPE_INT, sampleRate, nullptr));
- } else {
+ if (sampleRate <= 0) {
GST_INFO_OBJECT(m_pipeline.get(), "Not forcing sample rate");
- m_caps = adoptGRef(gst_caps_new_empty_simple("audio/x-raw"));
+
+ return false;
}
+ GST_INFO_OBJECT(m_pipeline.get(), "Setting SampleRate %d", sampleRate);
+ m_caps = adoptGRef(gst_caps_new_simple("audio/x-raw", "rate", G_TYPE_INT, sampleRate, nullptr));
+
if (!m_capsfilter.get())
return false;
Modified: trunk/Source/WebCore/platform/mediastream/gstreamer/GStreamerAudioCapturer.h (238950 => 238951)
--- trunk/Source/WebCore/platform/mediastream/gstreamer/GStreamerAudioCapturer.h 2018-12-07 06:18:33 UTC (rev 238950)
+++ trunk/Source/WebCore/platform/mediastream/gstreamer/GStreamerAudioCapturer.h 2018-12-07 10:48:56 UTC (rev 238951)
@@ -32,7 +32,6 @@
GStreamerAudioCapturer(GStreamerCaptureDevice);
GStreamerAudioCapturer();
- GstElement* createSource() final;
GstElement* createConverter() final;
const char* name() final { return "Audio"; }
Modified: trunk/Source/WebCore/platform/mediastream/gstreamer/GStreamerAudioStreamDescription.h (238950 => 238951)
--- trunk/Source/WebCore/platform/mediastream/gstreamer/GStreamerAudioStreamDescription.h 2018-12-07 06:18:33 UTC (rev 238950)
+++ trunk/Source/WebCore/platform/mediastream/gstreamer/GStreamerAudioStreamDescription.h 2018-12-07 10:48:56 UTC (rev 238951)
@@ -82,6 +82,7 @@
bool isSignedInteger() const final { return GST_AUDIO_INFO_IS_INTEGER(&m_info); }
bool isNativeEndian() const final { return GST_AUDIO_INFO_ENDIANNESS(&m_info) == G_BYTE_ORDER; }
bool isFloat() const final { return GST_AUDIO_INFO_IS_FLOAT(&m_info); }
+ int bytesPerFrame() { return GST_AUDIO_INFO_BPF(&m_info); }
uint32_t numberOfInterleavedChannels() const final { return isInterleaved() ? GST_AUDIO_INFO_CHANNELS(&m_info) : TRUE; }
uint32_t numberOfChannelStreams() const final { return GST_AUDIO_INFO_CHANNELS(&m_info); }
Modified: trunk/Source/WebCore/platform/mediastream/gstreamer/GStreamerMediaStreamSource.cpp (238950 => 238951)
--- trunk/Source/WebCore/platform/mediastream/gstreamer/GStreamerMediaStreamSource.cpp 2018-12-07 06:18:33 UTC (rev 238950)
+++ trunk/Source/WebCore/platform/mediastream/gstreamer/GStreamerMediaStreamSource.cpp 2018-12-07 10:48:56 UTC (rev 238951)
@@ -41,7 +41,6 @@
static void webkitMediaStreamSrcPushVideoSample(WebKitMediaStreamSrc* self, GstSample* gstsample);
static void webkitMediaStreamSrcPushAudioSample(WebKitMediaStreamSrc* self, GstSample* gstsample);
static void webkitMediaStreamSrcTrackEnded(WebKitMediaStreamSrc* self, MediaStreamTrackPrivate&);
-static void webkitMediaStreamSrcAddTrack(WebKitMediaStreamSrc* self, MediaStreamTrackPrivate*);
static void webkitMediaStreamSrcRemoveTrackByType(WebKitMediaStreamSrc* self, RealtimeMediaSource::Type trackType);
static GstStaticPadTemplate videoSrcTemplate = GST_STATIC_PAD_TEMPLATE("video_src",
@@ -156,7 +155,7 @@
void didAddTrack(MediaStreamTrackPrivate& track) final
{
- webkitMediaStreamSrcAddTrack(m_mediaStreamSrc.get(), &track);
+ webkitMediaStreamSrcAddTrack(m_mediaStreamSrc.get(), &track, false);
}
void didRemoveTrack(MediaStreamTrackPrivate& track) final
@@ -182,8 +181,8 @@
std::unique_ptr<WebKitMediaStreamTrackObserver> mediaStreamTrackObserver;
std::unique_ptr<WebKitMediaStreamObserver> mediaStreamObserver;
volatile gint npads;
- gulong probeid;
RefPtr<MediaStreamPrivate> stream;
+ RefPtr<MediaStreamTrackPrivate> track;
GstFlowCombiner* flowCombiner;
GRefPtr<GstStreamCollection> streamCollection;
@@ -314,8 +313,11 @@
if (transition == GST_STATE_CHANGE_PAUSED_TO_READY) {
GST_OBJECT_LOCK(self);
- for (auto& track : self->stream->tracks())
- track->removeObserver(*self->mediaStreamTrackObserver.get());
+ if (self->stream) {
+ for (auto& track : self->stream->tracks())
+ track->removeObserver(*self->mediaStreamTrackObserver.get());
+ } else if (self->track)
+ self->track->removeObserver(*self->mediaStreamTrackObserver.get());
GST_OBJECT_UNLOCK(self);
}
@@ -435,22 +437,26 @@
static gboolean webkitMediaStreamSrcSetupSrc(WebKitMediaStreamSrc* self,
MediaStreamTrackPrivate* track, GstElement* element,
- GstStaticPadTemplate* pad_template, gboolean observe_track)
+ GstStaticPadTemplate* pad_template, gboolean observe_track,
+ bool onlyTrack)
{
auto pad = adoptGRef(gst_element_get_static_pad(element, "src"));
gst_bin_add(GST_BIN(self), element);
- ProbeData* data = "" ProbeData;
- data->self = WEBKIT_MEDIA_STREAM_SRC(self);
- data->pad_template = pad_template;
- data->track = track;
+ if (!onlyTrack) {
+ ProbeData* data = "" ProbeData;
+ data->self = WEBKIT_MEDIA_STREAM_SRC(self);
+ data->pad_template = pad_template;
+ data->track = track;
- self->probeid = gst_pad_add_probe(pad.get(), (GstPadProbeType)GST_PAD_PROBE_TYPE_EVENT_DOWNSTREAM,
- (GstPadProbeCallback)webkitMediaStreamSrcPadProbeCb, data,
- [](gpointer data) {
- delete (ProbeData*)data;
- });
+ gst_pad_add_probe(pad.get(), (GstPadProbeType)GST_PAD_PROBE_TYPE_EVENT_DOWNSTREAM,
+ (GstPadProbeCallback)webkitMediaStreamSrcPadProbeCb, data,
+ [](gpointer data) {
+ delete (ProbeData*)data;
+ });
+ } else
+ webkitMediaStreamSrcAddPad(self, pad.get(), pad_template);
if (observe_track)
track->addObserver(*self->mediaStreamTrackObserver.get());
@@ -461,12 +467,12 @@
static gboolean webkitMediaStreamSrcSetupAppSrc(WebKitMediaStreamSrc* self,
MediaStreamTrackPrivate* track, GstElement** element,
- GstStaticPadTemplate* pad_template)
+ GstStaticPadTemplate* pad_template, bool onlyTrack)
{
*element = gst_element_factory_make("appsrc", nullptr);
g_object_set(*element, "is-live", true, "format", GST_FORMAT_TIME, nullptr);
- return webkitMediaStreamSrcSetupSrc(self, track, *element, pad_template, TRUE);
+ return webkitMediaStreamSrcSetupSrc(self, track, *element, pad_template, TRUE, onlyTrack);
}
static void webkitMediaStreamSrcPostStreamCollection(WebKitMediaStreamSrc* self, MediaStreamPrivate* stream)
@@ -484,14 +490,20 @@
gst_message_new_stream_collection(GST_OBJECT(self), self->streamCollection.get()));
}
-static void webkitMediaStreamSrcAddTrack(WebKitMediaStreamSrc* self, MediaStreamTrackPrivate* track)
+bool webkitMediaStreamSrcAddTrack(WebKitMediaStreamSrc* self, MediaStreamTrackPrivate* track, bool onlyTrack)
{
+ bool res = false;
if (track->type() == RealtimeMediaSource::Type::Audio)
- webkitMediaStreamSrcSetupAppSrc(self, track, &self->audioSrc, &audioSrcTemplate);
+ res = webkitMediaStreamSrcSetupAppSrc(self, track, &self->audioSrc, &audioSrcTemplate, onlyTrack);
else if (track->type() == RealtimeMediaSource::Type::Video)
- webkitMediaStreamSrcSetupAppSrc(self, track, &self->videoSrc, &videoSrcTemplate);
+ res = webkitMediaStreamSrcSetupAppSrc(self, track, &self->videoSrc, &videoSrcTemplate, onlyTrack);
else
GST_INFO("Unsupported track type: %d", static_cast<int>(track->type()));
+
+ if (onlyTrack && res)
+ self->track = track;
+
+ return false;
}
static void webkitMediaStreamSrcRemoveTrackByType(WebKitMediaStreamSrc* self, RealtimeMediaSource::Type trackType)
@@ -524,7 +536,7 @@
self->stream = stream;
self->stream->addObserver(*self->mediaStreamObserver.get());
for (auto& track : stream->tracks())
- webkitMediaStreamSrcAddTrack(self, track.get());
+ webkitMediaStreamSrcAddTrack(self, track.get(), false);
return TRUE;
}
@@ -593,6 +605,11 @@
gst_pad_push_event(pad.get(), gst_event_new_eos());
}
+GstElement* webkitMediaStreamSrcNew(void)
+{
+ return GST_ELEMENT(g_object_new(webkit_media_stream_src_get_type(), nullptr));
+}
+
} // WebCore
#endif // GST_CHECK_VERSION(1, 10, 0)
#endif // ENABLE(VIDEO) && ENABLE(MEDIA_STREAM) && USE(LIBWEBRTC)
Modified: trunk/Source/WebCore/platform/mediastream/gstreamer/GStreamerMediaStreamSource.h (238950 => 238951)
--- trunk/Source/WebCore/platform/mediastream/gstreamer/GStreamerMediaStreamSource.h 2018-12-07 06:18:33 UTC (rev 238950)
+++ trunk/Source/WebCore/platform/mediastream/gstreamer/GStreamerMediaStreamSource.h 2018-12-07 10:48:56 UTC (rev 238951)
@@ -41,6 +41,8 @@
#define WEBKIT_TYPE_MEDIA_STREAM_SRC (webkit_media_stream_src_get_type())
GType webkit_media_stream_src_get_type(void) G_GNUC_CONST;
bool webkitMediaStreamSrcSetStream(WebKitMediaStreamSrc*, MediaStreamPrivate*);
+bool webkitMediaStreamSrcAddTrack(WebKitMediaStreamSrc*, MediaStreamTrackPrivate*, bool onlyTrack);
+GstElement * webkitMediaStreamSrcNew(void);
} // WebCore
#endif // ENABLE(VIDEO) && ENABLE(MEDIA_STREAM) && USE(LIBWEBRTC)
Modified: trunk/Source/WebCore/platform/mediastream/gstreamer/MockGStreamerAudioCaptureSource.cpp (238950 => 238951)
--- trunk/Source/WebCore/platform/mediastream/gstreamer/MockGStreamerAudioCaptureSource.cpp 2018-12-07 06:18:33 UTC (rev 238950)
+++ trunk/Source/WebCore/platform/mediastream/gstreamer/MockGStreamerAudioCaptureSource.cpp 2018-12-07 10:48:56 UTC (rev 238951)
@@ -24,16 +24,110 @@
#if ENABLE(MEDIA_STREAM) && USE(LIBWEBRTC) && USE(GSTREAMER)
#include "MockGStreamerAudioCaptureSource.h"
+#include "GStreamerAudioStreamDescription.h"
#include "MockRealtimeAudioSource.h"
+#include <gst/app/gstappsrc.h>
+
namespace WebCore {
+static const double s_Tau = 2 * M_PI;
+static const double s_BipBopDuration = 0.07;
+static const double s_BipBopVolume = 0.5;
+static const double s_BipFrequency = 1500;
+static const double s_BopFrequency = 500;
+static const double s_HumFrequency = 150;
+static const double s_HumVolume = 0.1;
+
class WrappedMockRealtimeAudioSource : public MockRealtimeAudioSource {
public:
WrappedMockRealtimeAudioSource(String&& deviceID, String&& name, String&& hashSalt)
: MockRealtimeAudioSource(WTFMove(deviceID), WTFMove(name), WTFMove(hashSalt))
+ , m_src(nullptr)
{
}
+
+ void start(GRefPtr<GstElement> src)
+ {
+ m_src = src;
+ if (m_streamFormat)
+ gst_app_src_set_caps(GST_APP_SRC(m_src.get()), m_streamFormat->caps());
+ MockRealtimeAudioSource::start();
+ }
+
+ void addHum(float amplitude, float frequency, float sampleRate, uint64_t start, float *p, uint64_t count)
+ {
+ float humPeriod = sampleRate / frequency;
+ for (uint64_t i = start, end = start + count; i < end; ++i) {
+ float a = amplitude * sin(i * s_Tau / humPeriod);
+ a += *p;
+ *p++ = a;
+ }
+ }
+
+ void render(Seconds delta)
+ {
+ ASSERT(m_src);
+
+ uint32_t totalFrameCount = GST_ROUND_UP_16(static_cast<size_t>(delta.seconds() * sampleRate()));
+ uint32_t frameCount = std::min(totalFrameCount, m_maximiumFrameCount);
+ while (frameCount) {
+ uint32_t bipBopStart = m_samplesRendered % m_bipBopBuffer.size();
+ uint32_t bipBopRemain = m_bipBopBuffer.size() - bipBopStart;
+ uint32_t bipBopCount = std::min(frameCount, bipBopRemain);
+
+ GstBuffer* buffer = gst_buffer_new_allocate(nullptr, bipBopCount * m_streamFormat->bytesPerFrame(), nullptr);
+ {
+ GstMappedBuffer map(buffer, GST_MAP_WRITE);
+
+ if (!muted()) {
+ memcpy(map.data(), &m_bipBopBuffer[bipBopStart], sizeof(float) * bipBopCount);
+ addHum(s_HumVolume, s_HumFrequency, sampleRate(), m_samplesRendered, (float*)map.data(), bipBopCount);
+ } else
+ memset(map.data(), 0, sizeof(float) * bipBopCount);
+ }
+
+ gst_app_src_push_buffer(GST_APP_SRC(m_src.get()), buffer);
+ m_samplesRendered += bipBopCount;
+ totalFrameCount -= bipBopCount;
+ frameCount = std::min(totalFrameCount, m_maximiumFrameCount);
+ }
+ }
+
+ void settingsDidChange(OptionSet<RealtimeMediaSourceSettings::Flag> settings)
+ {
+ if (settings.contains(RealtimeMediaSourceSettings::Flag::SampleRate)) {
+ GstAudioInfo info;
+ auto rate = sampleRate();
+ size_t sampleCount = 2 * rate;
+
+ m_maximiumFrameCount = WTF::roundUpToPowerOfTwo(renderInterval().seconds() * sampleRate());
+ gst_audio_info_set_format(&info, GST_AUDIO_FORMAT_F32LE, rate, 1, nullptr);
+ m_streamFormat = GStreamerAudioStreamDescription(info);
+
+ if (m_src)
+ gst_app_src_set_caps(GST_APP_SRC(m_src.get()), m_streamFormat->caps());
+
+ m_bipBopBuffer.grow(sampleCount);
+ m_bipBopBuffer.fill(0);
+
+ size_t bipBopSampleCount = ceil(s_BipBopDuration * rate);
+ size_t bipStart = 0;
+ size_t bopStart = rate;
+
+ addHum(s_BipBopVolume, s_BipFrequency, rate, 0, static_cast<float*>(m_bipBopBuffer.data() + bipStart), bipBopSampleCount);
+ addHum(s_BipBopVolume, s_BopFrequency, rate, 0, static_cast<float*>(m_bipBopBuffer.data() + bopStart), bipBopSampleCount);
+ }
+
+ MockRealtimeAudioSource::settingsDidChange(settings);
+ }
+
+ GRefPtr<GstElement> m_src;
+ std::optional<GStreamerAudioStreamDescription> m_streamFormat;
+ Vector<float> m_bipBopBuffer;
+ uint32_t m_maximiumFrameCount;
+ uint64_t m_samplesEmitted { 0 };
+ uint64_t m_samplesRendered { 0 };
};
CaptureSourceOrError MockRealtimeAudioSource::create(String&& deviceID,
@@ -80,7 +174,7 @@
void MockGStreamerAudioCaptureSource::startProducingData()
{
GStreamerAudioCaptureSource::startProducingData();
- m_wrappedSource->start();
+ static_cast<WrappedMockRealtimeAudioSource*>(m_wrappedSource.get())->start(capturer()->source());
}
const RealtimeMediaSourceSettings& MockGStreamerAudioCaptureSource::settings()
Modified: trunk/Source/WebCore/platform/mediastream/gstreamer/RealtimeOutgoingAudioSourceLibWebRTC.cpp (238950 => 238951)
--- trunk/Source/WebCore/platform/mediastream/gstreamer/RealtimeOutgoingAudioSourceLibWebRTC.cpp 2018-12-07 06:18:33 UTC (rev 238950)
+++ trunk/Source/WebCore/platform/mediastream/gstreamer/RealtimeOutgoingAudioSourceLibWebRTC.cpp 2018-12-07 10:48:56 UTC (rev 238951)
@@ -108,33 +108,26 @@
size_t inChunkSampleCount = gst_audio_converter_get_in_frames(m_sampleConverter.get(), outChunkSampleCount);
size_t inBufferSize = inChunkSampleCount * m_inputStreamDescription->getInfo()->bpf;
- auto available = gst_adapter_available(m_adapter.get());
- if (inBufferSize > available) {
- GST_DEBUG("Not enough data: wanted: %ld > %ld available",
- inBufferSize, available);
+ while (gst_adapter_available(m_adapter.get()) > inBufferSize) {
+ auto inBuffer = adoptGRef(gst_adapter_take_buffer(m_adapter.get(), inBufferSize));
+ m_audioBuffer.grow(outBufferSize);
+ if (isSilenced())
+ gst_audio_format_fill_silence(m_outputStreamDescription->getInfo()->finfo, m_audioBuffer.data(), outBufferSize);
+ else {
+ GstMappedBuffer inMap(inBuffer.get(), GST_MAP_READ);
- return;
- }
+ gpointer in[1] = { inMap.data() };
+ gpointer out[1] = { m_audioBuffer.data() };
+ if (!gst_audio_converter_samples(m_sampleConverter.get(), static_cast<GstAudioConverterFlags>(0), in, inChunkSampleCount, out, outChunkSampleCount)) {
+ GST_ERROR("Could not convert samples.");
- auto inBuffer = adoptGRef(gst_adapter_take_buffer(m_adapter.get(), inBufferSize));
- auto outBuffer = adoptGRef(gst_buffer_new_allocate(nullptr, outBufferSize, 0));
- GstMappedBuffer outMap(outBuffer.get(), GST_MAP_WRITE);
- if (isSilenced())
- gst_audio_format_fill_silence(m_outputStreamDescription->getInfo()->finfo, outMap.data(), outMap.size());
- else {
- GstMappedBuffer inMap(inBuffer.get(), GST_MAP_READ);
+ return;
+ }
+ }
- gpointer in[1] = { inMap.data() };
- gpointer out[1] = { outMap.data() };
- if (!gst_audio_converter_samples(m_sampleConverter.get(), static_cast<GstAudioConverterFlags>(0), in, inChunkSampleCount, out, outChunkSampleCount)) {
- GST_ERROR("Could not convert samples.");
-
- return;
- }
+ sendAudioFrames(m_audioBuffer.data(), LibWebRTCAudioFormat::sampleSize, static_cast<int>(m_outputStreamDescription->sampleRate()),
+ static_cast<int>(m_outputStreamDescription->numberOfChannels()), outChunkSampleCount);
}
-
- sendAudioFrames(outMap.data(), LibWebRTCAudioFormat::sampleSize, static_cast<int>(m_outputStreamDescription->sampleRate()),
- static_cast<int>(m_outputStreamDescription->numberOfChannels()), outChunkSampleCount);
}
bool RealtimeOutgoingAudioSourceLibWebRTC::isReachingBufferedAudioDataHighLimit()
Modified: trunk/Source/WebCore/platform/mediastream/gstreamer/RealtimeOutgoingAudioSourceLibWebRTC.h (238950 => 238951)
--- trunk/Source/WebCore/platform/mediastream/gstreamer/RealtimeOutgoingAudioSourceLibWebRTC.h 2018-12-07 06:18:33 UTC (rev 238950)
+++ trunk/Source/WebCore/platform/mediastream/gstreamer/RealtimeOutgoingAudioSourceLibWebRTC.h 2018-12-07 10:48:56 UTC (rev 238951)
@@ -54,6 +54,7 @@
Lock m_adapterMutex;
GRefPtr<GstAdapter> m_adapter;
+ Vector<uint8_t> m_audioBuffer;
};
} // namespace WebCore
