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/*
* Copyright (C) 2011, 2012 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"
#if ENABLE(WEB_AUDIO)
#include "AudioDestinationGStreamer.h"
#include "AudioChannel.h"
#include "AudioSourceProvider.h"
#include <wtf/gobject/GOwnPtr.h>
#include "GRefPtrGStreamer.h"
#include "GStreamerVersioning.h"
#include "Logging.h"
#include "WebKitWebAudioSourceGStreamer.h"
#include <gst/gst.h>
#include <gst/pbutils/pbutils.h>
namespace WebCore {
// Size of the AudioBus for playback. The webkitwebaudiosrc element
// needs to handle this number of frames per cycle as well.
const unsigned framesToPull = 128;
gboolean messageCallback(GstBus*, GstMessage* message, AudioDestinationGStreamer* destination)
{
return destination->handleMessage(message);
}
PassOwnPtr<AudioDestination> AudioDestination::create(AudioIOCallback& callback, const String&, unsigned numberOfInputChannels, unsigned numberOfOutputChannels, float sampleRate)
{
// FIXME: make use of inputDeviceId as appropriate.
// FIXME: Add support for local/live audio input.
if (numberOfInputChannels)
LOG(Media, "AudioDestination::create(%u, %u, %f) - unhandled input channels", numberOfInputChannels, numberOfOutputChannels, sampleRate);
// FIXME: Add support for multi-channel (> stereo) output.
if (numberOfOutputChannels != 2)
LOG(Media, "AudioDestination::create(%u, %u, %f) - unhandled output channels", numberOfInputChannels, numberOfOutputChannels, sampleRate);
return adoptPtr(new AudioDestinationGStreamer(callback, sampleRate));
}
float AudioDestination::hardwareSampleRate()
{
return 44100;
}
unsigned long AudioDestination::maxChannelCount()
{
// FIXME: query the default audio hardware device to return the actual number
// of channels of the device. Also see corresponding FIXME in create().
return 0;
}
#ifndef GST_API_VERSION_1
static void onGStreamerWavparsePadAddedCallback(GstElement*, GstPad* pad, AudioDestinationGStreamer* destination)
{
destination->finishBuildingPipelineAfterWavParserPadReady(pad);
}
#endif
AudioDestinationGStreamer::AudioDestinationGStreamer(AudioIOCallback& callback, float sampleRate)
: m_callback(callback)
, m_renderBus(AudioBus::create(2, framesToPull, false))
, m_sampleRate(sampleRate)
, m_isPlaying(false)
{
m_pipeline = gst_pipeline_new("play");
GRefPtr<GstBus> bus = webkitGstPipelineGetBus(GST_PIPELINE(m_pipeline));
ASSERT(bus);
gst_bus_add_signal_watch(bus.get());
g_signal_connect(bus.get(), "message", G_CALLBACK(messageCallback), this);
GstElement* webkitAudioSrc = reinterpret_cast<GstElement*>(g_object_new(WEBKIT_TYPE_WEB_AUDIO_SRC,
"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;
#ifndef GST_API_VERSION_1
g_signal_connect(wavParser, "pad-added", G_CALLBACK(onGStreamerWavparsePadAddedCallback), this);
#endif
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);
#ifdef GST_API_VERSION_1
GRefPtr<GstPad> srcPad = adoptGRef(gst_element_get_static_pad(wavParser, "src"));
finishBuildingPipelineAfterWavParserPadReady(srcPad.get());
#endif
}
AudioDestinationGStreamer::~AudioDestinationGStreamer()
{
GRefPtr<GstBus> bus = webkitGstPipelineGetBus(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);
GRefPtr<GstElement> audioSink = gst_element_factory_make("autoaudiosink", 0);
m_audioSinkAvailable = audioSink;
if (!audioSink) {
LOG_ERROR("Failed to create GStreamer autoaudiosink element");
return;
}
// 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.
GstStateChangeReturn stateChangeReturn = gst_element_set_state(audioSink.get(), GST_STATE_READY);
if (stateChangeReturn == GST_STATE_CHANGE_FAILURE) {
LOG_ERROR("Failed to change autoaudiosink element state");
gst_element_set_state(audioSink.get(), GST_STATE_NULL);
m_audioSinkAvailable = false;
return;
}
GstElement* audioConvert = gst_element_factory_make("audioconvert", 0);
gst_bin_add_many(GST_BIN(m_pipeline), audioConvert, audioSink.get(), NULL);
// 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 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());
}
gboolean AudioDestinationGStreamer::handleMessage(GstMessage* message)
{
GOwnPtr<GError> error;
GOwnPtr<gchar> debug;
switch (GST_MESSAGE_TYPE(message)) {
case GST_MESSAGE_WARNING:
gst_message_parse_warning(message, &error.outPtr(), &debug.outPtr());
g_warning("Warning: %d, %s. Debug output: %s", error->code, error->message, debug.get());
break;
case GST_MESSAGE_ERROR:
gst_message_parse_error(message, &error.outPtr(), &debug.outPtr());
g_warning("Error: %d, %s. Debug output: %s", error->code, error->message, debug.get());
gst_element_set_state(m_pipeline, GST_STATE_NULL);
m_isPlaying = false;
break;
default:
break;
}
return TRUE;
}
void AudioDestinationGStreamer::start()
{
ASSERT(m_wavParserAvailable);
if (!m_wavParserAvailable)
return;
gst_element_set_state(m_pipeline, GST_STATE_PLAYING);
m_isPlaying = true;
}
void AudioDestinationGStreamer::stop()
{
ASSERT(m_wavParserAvailable && m_audioSinkAvailable);
if (!m_wavParserAvailable || !m_audioSinkAvailable)
return;
gst_element_set_state(m_pipeline, GST_STATE_PAUSED);
m_isPlaying = false;
}
} // namespace WebCore
#endif // ENABLE(WEB_AUDIO)
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