webkitgtk-2.16.5HEADwebkitgtk-2.16.5master

1 files changed, 349 insertions, 0 deletions

diff --git a/Source/WebCore/platform/audio/gstreamer/AudioSourceProviderGStreamer.cpp b/Source/WebCore/platform/audio/gstreamer/AudioSourceProviderGStreamer.cpp
new file mode 100644
index 000000000..4d7f4154d
--- /dev/null
+++ b/Source/WebCore/platform/audio/gstreamer/AudioSourceProviderGStreamer.cpp
@@ -0,0 +1,349 @@
+/*
+ *  Copyright (C) 2014 Igalia S.L
+ *
+ *  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 "AudioSourceProviderGStreamer.h"
+
+#if ENABLE(WEB_AUDIO) && ENABLE(VIDEO) && USE(GSTREAMER)
+
+#include "AudioBus.h"
+#include "AudioSourceProviderClient.h"
+#include <gst/app/gstappsink.h>
+#include <gst/audio/audio-info.h>
+#include <gst/base/gstadapter.h>
+#include <wtf/glib/GMutexLocker.h>
+
+
+namespace WebCore {
+
+// For now the provider supports only stereo files at a fixed sample
+// bitrate.
+static const int gNumberOfChannels = 2;
+static const float gSampleBitRate = 44100;
+
+static GstFlowReturn onAppsinkNewBufferCallback(GstAppSink* sink, gpointer userData)
+{
+    return static_cast<AudioSourceProviderGStreamer*>(userData)->handleAudioBuffer(sink);
+}
+
+static void onGStreamerDeinterleavePadAddedCallback(GstElement*, GstPad* pad, AudioSourceProviderGStreamer* provider)
+{
+    provider->handleNewDeinterleavePad(pad);
+}
+
+static void onGStreamerDeinterleaveReadyCallback(GstElement*, AudioSourceProviderGStreamer* provider)
+{
+    provider->deinterleavePadsConfigured();
+}
+
+static void onGStreamerDeinterleavePadRemovedCallback(GstElement*, GstPad* pad, AudioSourceProviderGStreamer* provider)
+{
+    provider->handleRemovedDeinterleavePad(pad);
+}
+
+static GstPadProbeReturn onAppsinkFlushCallback(GstPad*, GstPadProbeInfo* info, gpointer userData)
+{
+    if (GST_PAD_PROBE_INFO_TYPE(info) & (GST_PAD_PROBE_TYPE_EVENT_DOWNSTREAM | GST_PAD_PROBE_TYPE_EVENT_FLUSH)) {
+        GstEvent* event = GST_PAD_PROBE_INFO_EVENT(info);
+        if (GST_EVENT_TYPE(event) == GST_EVENT_FLUSH_STOP) {
+            AudioSourceProviderGStreamer* provider = reinterpret_cast<AudioSourceProviderGStreamer*>(userData);
+            provider->clearAdapters();
+        }
+    }
+    return GST_PAD_PROBE_OK;
+}
+
+static void copyGStreamerBuffersToAudioChannel(GstAdapter* adapter, AudioBus* bus , int channelNumber, size_t framesToProcess)
+{
+    if (!gst_adapter_available(adapter)) {
+        bus->zero();
+        return;
+    }
+
+    size_t bytes = framesToProcess * sizeof(float);
+    if (gst_adapter_available(adapter) >= bytes) {
+        gst_adapter_copy(adapter, bus->channel(channelNumber)->mutableData(), 0, bytes);
+        gst_adapter_flush(adapter, bytes);
+    }
+}
+
+AudioSourceProviderGStreamer::AudioSourceProviderGStreamer()
+    : m_client(nullptr)
+    , m_deinterleaveSourcePads(0)
+    , m_deinterleavePadAddedHandlerId(0)
+    , m_deinterleaveNoMorePadsHandlerId(0)
+    , m_deinterleavePadRemovedHandlerId(0)
+{
+    g_mutex_init(&m_adapterMutex);
+    m_frontLeftAdapter = gst_adapter_new();
+    m_frontRightAdapter = gst_adapter_new();
+}
+
+AudioSourceProviderGStreamer::~AudioSourceProviderGStreamer()
+{
+    GRefPtr<GstElement> deinterleave = adoptGRef(gst_bin_get_by_name(GST_BIN(m_audioSinkBin.get()), "deinterleave"));
+    if (deinterleave) {
+        g_signal_handler_disconnect(deinterleave.get(), m_deinterleavePadAddedHandlerId);
+        g_signal_handler_disconnect(deinterleave.get(), m_deinterleaveNoMorePadsHandlerId);
+        g_signal_handler_disconnect(deinterleave.get(), m_deinterleavePadRemovedHandlerId);
+    }
+
+    g_object_unref(m_frontLeftAdapter);
+    g_object_unref(m_frontRightAdapter);
+    g_mutex_clear(&m_adapterMutex);
+}
+
+void AudioSourceProviderGStreamer::configureAudioBin(GstElement* audioBin, GstElement* teePredecessor)
+{
+    m_audioSinkBin = audioBin;
+
+    GstElement* audioTee = gst_element_factory_make("tee", "audioTee");
+    GstElement* audioQueue = gst_element_factory_make("queue", nullptr);
+    GstElement* audioConvert = gst_element_factory_make("audioconvert", nullptr);
+    GstElement* audioConvert2 = gst_element_factory_make("audioconvert", nullptr);
+    GstElement* audioResample = gst_element_factory_make("audioresample", nullptr);
+    GstElement* audioResample2 = gst_element_factory_make("audioresample", nullptr);
+    GstElement* volumeElement = gst_element_factory_make("volume", "volume");
+    GstElement* audioSink = gst_element_factory_make("autoaudiosink", nullptr);
+
+    gst_bin_add_many(GST_BIN(m_audioSinkBin.get()), audioTee, audioQueue, audioConvert, audioResample, volumeElement, audioConvert2, audioResample2, audioSink, nullptr);
+
+    // In cases where the audio-sink needs elements before tee (such
+    // as scaletempo) they need to be linked to tee which in this case
+    // doesn't need a ghost pad. It is assumed that the teePredecessor
+    // chain already configured a ghost pad.
+    if (teePredecessor)
+        gst_element_link_pads_full(teePredecessor, "src", audioTee, "sink", GST_PAD_LINK_CHECK_NOTHING);
+    else {
+        // Add a ghostpad to the bin so it can proxy to tee.
+        GRefPtr<GstPad> audioTeeSinkPad = adoptGRef(gst_element_get_static_pad(audioTee, "sink"));
+        gst_element_add_pad(m_audioSinkBin.get(), gst_ghost_pad_new("sink", audioTeeSinkPad.get()));
+    }
+
+    // Link a new src pad from tee to queue ! audioconvert !
+    // audioresample ! volume ! audioconvert ! audioresample !
+    // autoaudiosink. The audioresample and audioconvert are needed to
+    // ensure the audio sink receives buffers in the correct format.
+    gst_element_link_pads_full(audioTee, "src_%u", audioQueue, "sink", GST_PAD_LINK_CHECK_NOTHING);
+    gst_element_link_pads_full(audioQueue, "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", volumeElement, "sink", GST_PAD_LINK_CHECK_NOTHING);
+    gst_element_link_pads_full(volumeElement, "src", audioConvert2, "sink", GST_PAD_LINK_CHECK_NOTHING);
+    gst_element_link_pads_full(audioConvert2, "src", audioResample2, "sink", GST_PAD_LINK_CHECK_NOTHING);
+    gst_element_link_pads_full(audioResample2, "src", audioSink, "sink", GST_PAD_LINK_CHECK_NOTHING);
+}
+
+void AudioSourceProviderGStreamer::provideInput(AudioBus* bus, size_t framesToProcess)
+{
+    WTF::GMutexLocker<GMutex> lock(m_adapterMutex);
+    copyGStreamerBuffersToAudioChannel(m_frontLeftAdapter, bus, 0, framesToProcess);
+    copyGStreamerBuffersToAudioChannel(m_frontRightAdapter, bus, 1, framesToProcess);
+}
+
+GstFlowReturn AudioSourceProviderGStreamer::handleAudioBuffer(GstAppSink* sink)
+{
+    if (!m_client)
+        return GST_FLOW_OK;
+
+    // Pull a buffer from appsink and store it the appropriate buffer
+    // list for the audio channel it represents.
+    GRefPtr<GstSample> sample = adoptGRef(gst_app_sink_pull_sample(sink));
+    if (!sample)
+        return gst_app_sink_is_eos(sink) ? GST_FLOW_EOS : GST_FLOW_ERROR;
+
+    GstBuffer* buffer = gst_sample_get_buffer(sample.get());
+    if (!buffer)
+        return GST_FLOW_ERROR;
+
+    GstCaps* caps = gst_sample_get_caps(sample.get());
+    if (!caps)
+        return GST_FLOW_ERROR;
+
+    GstAudioInfo info;
+    gst_audio_info_from_caps(&info, caps);
+
+    WTF::GMutexLocker<GMutex> lock(m_adapterMutex);
+
+    // Check the first audio channel. The buffer is supposed to store
+    // data of a single channel anyway.
+    switch (GST_AUDIO_INFO_POSITION(&info, 0)) {
+    case GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT:
+    case GST_AUDIO_CHANNEL_POSITION_MONO:
+        gst_adapter_push(m_frontLeftAdapter, gst_buffer_ref(buffer));
+        break;
+    case GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT:
+        gst_adapter_push(m_frontRightAdapter, gst_buffer_ref(buffer));
+        break;
+    default:
+        break;
+    }
+
+    return GST_FLOW_OK;
+}
+
+void AudioSourceProviderGStreamer::setClient(AudioSourceProviderClient* client)
+{
+    ASSERT(client);
+    m_client = client;
+
+    // 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);
+
+    // The audioconvert and audioresample elements are needed to
+    // ensure deinterleave and the sinks downstream receive buffers in
+    // the format specified by the capsfilter.
+    GstElement* audioQueue = gst_element_factory_make("queue", nullptr);
+    GstElement* audioConvert  = gst_element_factory_make("audioconvert", nullptr);
+    GstElement* audioResample = gst_element_factory_make("audioresample", nullptr);
+    GstElement* capsFilter = gst_element_factory_make("capsfilter", nullptr);
+    GstElement* deInterleave = gst_element_factory_make("deinterleave", "deinterleave");
+
+    g_object_set(deInterleave, "keep-positions", TRUE, nullptr);
+    m_deinterleavePadAddedHandlerId = g_signal_connect(deInterleave, "pad-added", G_CALLBACK(onGStreamerDeinterleavePadAddedCallback), this);
+    m_deinterleaveNoMorePadsHandlerId = g_signal_connect(deInterleave, "no-more-pads", G_CALLBACK(onGStreamerDeinterleaveReadyCallback), this);
+    m_deinterleavePadRemovedHandlerId = g_signal_connect(deInterleave, "pad-removed", G_CALLBACK(onGStreamerDeinterleavePadRemovedCallback), this);
+
+    GstCaps* caps = gst_caps_new_simple("audio/x-raw", "rate", G_TYPE_INT, static_cast<int>(gSampleBitRate),
+        "channels", G_TYPE_INT, gNumberOfChannels,
+        "format", G_TYPE_STRING, GST_AUDIO_NE(F32),
+        "layout", G_TYPE_STRING, "interleaved", nullptr);
+
+    g_object_set(capsFilter, "caps", caps, nullptr);
+    gst_caps_unref(caps);
+
+    gst_bin_add_many(GST_BIN(m_audioSinkBin.get()), audioQueue, audioConvert, audioResample, capsFilter, deInterleave, nullptr);
+
+    GRefPtr<GstElement> audioTee = adoptGRef(gst_bin_get_by_name(GST_BIN(m_audioSinkBin.get()), "audioTee"));
+
+    // Link a new src pad from tee to queue ! audioconvert !
+    // audioresample ! capsfilter ! deinterleave. Later
+    // on each deinterleaved planar audio channel will be routed to an
+    // appsink for data extraction and processing.
+    gst_element_link_pads_full(audioTee.get(), "src_%u", audioQueue, "sink", GST_PAD_LINK_CHECK_NOTHING);
+    gst_element_link_pads_full(audioQueue, "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", capsFilter, "sink", GST_PAD_LINK_CHECK_NOTHING);
+    gst_element_link_pads_full(capsFilter, "src", deInterleave, "sink", GST_PAD_LINK_CHECK_NOTHING);
+
+    gst_element_sync_state_with_parent(audioQueue);
+    gst_element_sync_state_with_parent(audioConvert);
+    gst_element_sync_state_with_parent(audioResample);
+    gst_element_sync_state_with_parent(capsFilter);
+    gst_element_sync_state_with_parent(deInterleave);
+}
+
+void AudioSourceProviderGStreamer::handleNewDeinterleavePad(GstPad* pad)
+{
+    m_deinterleaveSourcePads++;
+
+    if (m_deinterleaveSourcePads > 2) {
+        g_warning("The AudioSourceProvider supports only mono and stereo audio. Silencing out this new channel.");
+        GstElement* queue = gst_element_factory_make("queue", nullptr);
+        GstElement* sink = gst_element_factory_make("fakesink", nullptr);
+        g_object_set(sink, "async", FALSE, nullptr);
+        gst_bin_add_many(GST_BIN(m_audioSinkBin.get()), queue, sink, nullptr);
+
+        GRefPtr<GstPad> sinkPad = adoptGRef(gst_element_get_static_pad(queue, "sink"));
+        gst_pad_link_full(pad, sinkPad.get(), GST_PAD_LINK_CHECK_NOTHING);
+
+        GQuark quark = g_quark_from_static_string("peer");
+        g_object_set_qdata(G_OBJECT(pad), quark, sinkPad.get());
+        gst_element_link_pads_full(queue, "src", sink, "sink", GST_PAD_LINK_CHECK_NOTHING);
+        gst_element_sync_state_with_parent(queue);
+        gst_element_sync_state_with_parent(sink);
+        return;
+    }
+
+    // A new pad for a planar channel was added in deinterleave. Plug
+    // in an appsink so we can pull the data from each
+    // channel. Pipeline looks like:
+    // ... deinterleave ! queue ! appsink.
+    GstElement* queue = gst_element_factory_make("queue", nullptr);
+    GstElement* sink = gst_element_factory_make("appsink", nullptr);
+
+    GstAppSinkCallbacks callbacks;
+    callbacks.eos = nullptr;
+    callbacks.new_preroll = nullptr;
+    callbacks.new_sample = onAppsinkNewBufferCallback;
+    gst_app_sink_set_callbacks(GST_APP_SINK(sink), &callbacks, this, nullptr);
+
+    g_object_set(sink, "async", FALSE, nullptr);
+
+    GRefPtr<GstCaps> caps = adoptGRef(gst_caps_new_simple("audio/x-raw", "rate", G_TYPE_INT, static_cast<int>(gSampleBitRate),
+        "channels", G_TYPE_INT, 1,
+        "format", G_TYPE_STRING, GST_AUDIO_NE(F32),
+        "layout", G_TYPE_STRING, "interleaved", nullptr));
+
+    gst_app_sink_set_caps(GST_APP_SINK(sink), caps.get());
+
+    gst_bin_add_many(GST_BIN(m_audioSinkBin.get()), queue, sink, nullptr);
+
+    GRefPtr<GstPad> sinkPad = adoptGRef(gst_element_get_static_pad(queue, "sink"));
+    gst_pad_link_full(pad, sinkPad.get(), GST_PAD_LINK_CHECK_NOTHING);
+
+    GQuark quark = g_quark_from_static_string("peer");
+    g_object_set_qdata(G_OBJECT(pad), quark, sinkPad.get());
+
+    gst_element_link_pads_full(queue, "src", sink, "sink", GST_PAD_LINK_CHECK_NOTHING);
+
+    sinkPad = adoptGRef(gst_element_get_static_pad(sink, "sink"));
+    gst_pad_add_probe(sinkPad.get(), GST_PAD_PROBE_TYPE_EVENT_FLUSH, onAppsinkFlushCallback, this, nullptr);
+
+    gst_element_sync_state_with_parent(queue);
+    gst_element_sync_state_with_parent(sink);
+}
+
+void AudioSourceProviderGStreamer::handleRemovedDeinterleavePad(GstPad* pad)
+{
+    m_deinterleaveSourcePads--;
+
+    // Remove the queue ! appsink chain downstream of deinterleave.
+    GQuark quark = g_quark_from_static_string("peer");
+    GstPad* sinkPad = reinterpret_cast<GstPad*>(g_object_get_qdata(G_OBJECT(pad), quark));
+    GRefPtr<GstElement> queue = adoptGRef(gst_pad_get_parent_element(sinkPad));
+    GRefPtr<GstPad> queueSrcPad = adoptGRef(gst_element_get_static_pad(queue.get(), "src"));
+    GRefPtr<GstPad> appsinkSinkPad = adoptGRef(gst_pad_get_peer(queueSrcPad.get()));
+    GRefPtr<GstElement> sink = adoptGRef(gst_pad_get_parent_element(appsinkSinkPad.get()));
+    gst_element_set_state(sink.get(), GST_STATE_NULL);
+    gst_element_set_state(queue.get(), GST_STATE_NULL);
+    gst_element_unlink(queue.get(), sink.get());
+    gst_bin_remove_many(GST_BIN(m_audioSinkBin.get()), queue.get(), sink.get(), nullptr);
+}
+
+void AudioSourceProviderGStreamer::deinterleavePadsConfigured()
+{
+    ASSERT(m_client);
+    ASSERT(m_deinterleaveSourcePads == gNumberOfChannels);
+
+    m_client->setFormat(m_deinterleaveSourcePads, gSampleBitRate);
+}
+
+void AudioSourceProviderGStreamer::clearAdapters()
+{
+    WTF::GMutexLocker<GMutex> lock(m_adapterMutex);
+    gst_adapter_clear(m_frontLeftAdapter);
+    gst_adapter_clear(m_frontRightAdapter);
+}
+
+} // WebCore
+
+#endif // ENABLE(WEB_AUDIO) && ENABLE(VIDEO) && USE(GSTREAMER)