PassRefPtr<AudioBus> AudioFileReader::createBus(float sampleRate, bool mixToMono)
{
m_sampleRate = sampleRate;
m_frontLeftBuffers = gst_buffer_list_new();
m_frontRightBuffers = gst_buffer_list_new();
GRefPtr<GMainContext> context = adoptGRef(g_main_context_new());
g_main_context_push_thread_default(context.get());
m_loop = adoptGRef(g_main_loop_new(context.get(), FALSE));
// Start the pipeline processing just after the loop is started.
GRefPtr<GSource> timeoutSource = adoptGRef(g_timeout_source_new(0));
g_source_attach(timeoutSource.get(), context.get());
g_source_set_callback(timeoutSource.get(), reinterpret_cast<GSourceFunc>(enteredMainLoopCallback), this, 0);
g_main_loop_run(m_loop.get());
g_main_context_pop_thread_default(context.get());
if (m_errorOccurred)
return 0;
unsigned channels = mixToMono ? 1 : 2;
RefPtr<AudioBus> audioBus = AudioBus::create(channels, m_channelSize, true);
audioBus->setSampleRate(m_sampleRate);
copyGstreamerBuffersToAudioChannel(m_frontLeftBuffers, audioBus->channel(0));
if (!mixToMono)
copyGstreamerBuffersToAudioChannel(m_frontRightBuffers, audioBus->channel(1));
return audioBus;
}
PassRefPtr<AudioBus> AudioFileReader::createBus(float sampleRate, bool mixToMono)
{
m_sampleRate = sampleRate;
m_channels = mixToMono ? 1 : 2;
m_frontLeftBuffers = gst_buffer_list_new();
m_frontRightBuffers = gst_buffer_list_new();
GRefPtr<GMainContext> context = adoptGRef(g_main_context_new());
g_main_context_push_thread_default(context.get());
m_loop = adoptGRef(g_main_loop_new(context.get(), FALSE));
// Start the pipeline processing just after the loop is started.
GThreadSafeMainLoopSource source;
source.schedule("[WebKit] AudioFileReader::decodeAudioForBusCreation", std::function<void()>(std::bind(&AudioFileReader::decodeAudioForBusCreation, this)), G_PRIORITY_DEFAULT, nullptr, context.get());
g_main_loop_run(m_loop.get());
g_main_context_pop_thread_default(context.get());
// Set pipeline to GST_STATE_NULL state here already ASAP to
// release any resources that might still be used.
gst_element_set_state(m_pipeline, GST_STATE_NULL);
if (m_errorOccurred)
return 0;
RefPtr<AudioBus> audioBus = AudioBus::create(m_channels, m_channelSize, true);
audioBus->setSampleRate(m_sampleRate);
copyGstreamerBuffersToAudioChannel(m_frontLeftBuffers, audioBus->channel(0));
if (!mixToMono)
copyGstreamerBuffersToAudioChannel(m_frontRightBuffers, audioBus->channel(1));
return audioBus;
}
PassOwnPtr<AudioBus> AudioFileReader::createBus(float sampleRate, bool mixToMono)
{
m_sampleRate = sampleRate;
m_frontLeftBuffers = gst_buffer_list_new();
m_frontRightBuffers = gst_buffer_list_new();
#ifndef GST_API_VERSION_1
m_frontLeftBuffersIterator = gst_buffer_list_iterate(m_frontLeftBuffers);
gst_buffer_list_iterator_add_group(m_frontLeftBuffersIterator);
m_frontRightBuffersIterator = gst_buffer_list_iterate(m_frontRightBuffers);
gst_buffer_list_iterator_add_group(m_frontRightBuffersIterator);
#endif
GRefPtr<GMainContext> context = g_main_context_new();
g_main_context_push_thread_default(context.get());
m_loop = adoptGRef(g_main_loop_new(context.get(), FALSE));
// Start the pipeline processing just after the loop is started.
GSource* timeoutSource = g_timeout_source_new(0);
g_source_attach(timeoutSource, context.get());
g_source_set_callback(timeoutSource, reinterpret_cast<GSourceFunc>(enteredMainLoopCallback), this, 0);
g_main_loop_run(m_loop.get());
g_main_context_pop_thread_default(context.get());
if (m_errorOccurred)
return nullptr;
unsigned channels = mixToMono ? 1 : 2;
OwnPtr<AudioBus> audioBus = adoptPtr(new AudioBus(channels, m_channelSize, true));
audioBus->setSampleRate(m_sampleRate);
copyGstreamerBuffersToAudioChannel(m_frontLeftBuffers, audioBus->channel(0));
if (!mixToMono)
copyGstreamerBuffersToAudioChannel(m_frontRightBuffers, audioBus->channel(1));
return audioBus.release();
}
