webkitgtk-2.16.5HEADwebkitgtk-2.16.5master

1 files changed, 40 insertions, 48 deletions

diff --git a/Source/WebCore/platform/audio/gstreamer/AudioDestinationGStreamer.cpp b/Source/WebCore/platform/audio/gstreamer/AudioDestinationGStreamer.cpp
index 25ddcb9fa..758389ced 100644
--- a/Source/WebCore/platform/audio/gstreamer/AudioDestinationGStreamer.cpp
+++ b/Source/WebCore/platform/audio/gstreamer/AudioDestinationGStreamer.cpp
@@ -1,5 +1,6 @@
 /*
  *  Copyright (C) 2011, 2012 Igalia S.L
+ *  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
@@ -27,9 +28,9 @@
 #include "GRefPtrGStreamer.h"
 #include "Logging.h"
 #include "WebKitWebAudioSourceGStreamer.h"
+#include <gst/audio/gstaudiobasesink.h>
 #include <gst/gst.h>
-#include <gst/pbutils/pbutils.h>
-#include <wtf/gobject/GUniquePtr.h>
+#include <wtf/glib/GUniquePtr.h>
 
 namespace WebCore {
 
@@ -42,6 +43,12 @@ gboolean messageCallback(GstBus*, GstMessage* message, AudioDestinationGStreamer
     return destination->handleMessage(message);
 }
 
+static void autoAudioSinkChildAddedCallback(GstChildProxy*, GObject* object, gchar*, gpointer)
+{
+    if (GST_IS_AUDIO_BASE_SINK(object))
+        g_object_set(GST_AUDIO_BASE_SINK(object), "buffer-time", static_cast<gint64>(100000), nullptr);
+}
+
 std::unique_ptr<AudioDestination> AudioDestination::create(AudioIOCallback& callback, const String&, unsigned numberOfInputChannels, unsigned numberOfOutputChannels, float sampleRate)
 {
     // FIXME: make use of inputDeviceId as appropriate.
@@ -85,45 +92,17 @@ AudioDestinationGStreamer::AudioDestinationGStreamer(AudioIOCallback& callback,
                                                                             "rate", sampleRate,
                                                                             "bus", m_renderBus.get(),
                                                                             "provider", &m_callback,
-                                                                            "frames", framesToPull, NULL));
-
-    GstElement* wavParser = gst_element_factory_make("wavparse", 0);
-
-    m_wavParserAvailable = wavParser;
-    ASSERT_WITH_MESSAGE(m_wavParserAvailable, "Failed to create GStreamer wavparse element");
-    if (!m_wavParserAvailable)
-        return;
-
-    gst_bin_add_many(GST_BIN(m_pipeline), webkitAudioSrc, wavParser, NULL);
-    gst_element_link_pads_full(webkitAudioSrc, "src", wavParser, "sink", GST_PAD_LINK_CHECK_NOTHING);
-
-    GRefPtr<GstPad> srcPad = adoptGRef(gst_element_get_static_pad(wavParser, "src"));
-    finishBuildingPipelineAfterWavParserPadReady(srcPad.get());
-}
-
-AudioDestinationGStreamer::~AudioDestinationGStreamer()
-{
-    GRefPtr<GstBus> bus = adoptGRef(gst_pipeline_get_bus(GST_PIPELINE(m_pipeline)));
-    ASSERT(bus);
-    g_signal_handlers_disconnect_by_func(bus.get(), reinterpret_cast<gpointer>(messageCallback), this);
-    gst_bus_remove_signal_watch(bus.get());
-
-    gst_element_set_state(m_pipeline, GST_STATE_NULL);
-    gst_object_unref(m_pipeline);
-}
-
-void AudioDestinationGStreamer::finishBuildingPipelineAfterWavParserPadReady(GstPad* pad)
-{
-    ASSERT(m_wavParserAvailable);
+                                                                            "frames", framesToPull, nullptr));
 
-    GRefPtr<GstElement> audioSink = gst_element_factory_make("autoaudiosink", 0);
+    GRefPtr<GstElement> audioSink = gst_element_factory_make("autoaudiosink", nullptr);
     m_audioSinkAvailable = audioSink;
-
     if (!audioSink) {
         LOG_ERROR("Failed to create GStreamer autoaudiosink element");
         return;
     }
 
+    g_signal_connect(audioSink.get(), "child-added", G_CALLBACK(autoAudioSinkChildAddedCallback), nullptr);
+
     // Autoaudiosink does the real sink detection in the GST_STATE_NULL->READY transition
     // so it's best to roll it to READY as soon as possible to ensure the underlying platform
     // audiosink was loaded correctly.
@@ -135,17 +114,25 @@ void AudioDestinationGStreamer::finishBuildingPipelineAfterWavParserPadReady(Gst
         return;
     }
 
-    GstElement* audioConvert = gst_element_factory_make("audioconvert", 0);
-    gst_bin_add_many(GST_BIN(m_pipeline), audioConvert, audioSink.get(), NULL);
+    GstElement* audioConvert = gst_element_factory_make("audioconvert", nullptr);
+    GstElement* audioResample = gst_element_factory_make("audioresample", nullptr);
+    gst_bin_add_many(GST_BIN(m_pipeline), webkitAudioSrc, audioConvert, audioResample, audioSink.get(), nullptr);
 
-    // Link wavparse's src pad to audioconvert sink pad.
-    GRefPtr<GstPad> sinkPad = adoptGRef(gst_element_get_static_pad(audioConvert, "sink"));
-    gst_pad_link_full(pad, sinkPad.get(), GST_PAD_LINK_CHECK_NOTHING);
+    // Link src pads from webkitAudioSrc to audioConvert ! audioResample ! autoaudiosink.
+    gst_element_link_pads_full(webkitAudioSrc, "src", audioConvert, "sink", GST_PAD_LINK_CHECK_NOTHING);
+    gst_element_link_pads_full(audioConvert, "src", audioResample, "sink", GST_PAD_LINK_CHECK_NOTHING);
+    gst_element_link_pads_full(audioResample, "src", audioSink.get(), "sink", GST_PAD_LINK_CHECK_NOTHING);
+}
 
-    // Link audioconvert to audiosink and roll states.
-    gst_element_link_pads_full(audioConvert, "src", audioSink.get(), "sink", GST_PAD_LINK_CHECK_NOTHING);
-    gst_element_sync_state_with_parent(audioConvert);
-    gst_element_sync_state_with_parent(audioSink.leakRef());
+AudioDestinationGStreamer::~AudioDestinationGStreamer()
+{
+    GRefPtr<GstBus> bus = adoptGRef(gst_pipeline_get_bus(GST_PIPELINE(m_pipeline)));
+    ASSERT(bus);
+    g_signal_handlers_disconnect_by_func(bus.get(), reinterpret_cast<gpointer>(messageCallback), this);
+    gst_bus_remove_signal_watch(bus.get());
+
+    gst_element_set_state(m_pipeline, GST_STATE_NULL);
+    gst_object_unref(m_pipeline);
 }
 
 gboolean AudioDestinationGStreamer::handleMessage(GstMessage* message)
@@ -172,18 +159,23 @@ gboolean AudioDestinationGStreamer::handleMessage(GstMessage* message)
 
 void AudioDestinationGStreamer::start()
 {
-    ASSERT(m_wavParserAvailable);
-    if (!m_wavParserAvailable)
+    ASSERT(m_audioSinkAvailable);
+    if (!m_audioSinkAvailable)
         return;
 
-    gst_element_set_state(m_pipeline, GST_STATE_PLAYING);
+    if (gst_element_set_state(m_pipeline, GST_STATE_PLAYING) == GST_STATE_CHANGE_FAILURE) {
+        g_warning("Error: Failed to set pipeline to playing");
+        m_isPlaying = false;
+        return;
+    }
+
     m_isPlaying = true;
 }
 
 void AudioDestinationGStreamer::stop()
 {
-    ASSERT(m_wavParserAvailable && m_audioSinkAvailable);
-    if (!m_wavParserAvailable || !m_audioSinkAvailable)
+    ASSERT(m_audioSinkAvailable);
+    if (!m_audioSinkAvailable)
         return;
 
     gst_element_set_state(m_pipeline, GST_STATE_PAUSED);