/* * 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 #include #include #include 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(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(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 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 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 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 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 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 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(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 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 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 caps = adoptGRef(gst_caps_new_simple("audio/x-raw", "rate", G_TYPE_INT, static_cast(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 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(g_object_get_qdata(G_OBJECT(pad), quark)); GRefPtr queue = adoptGRef(gst_pad_get_parent_element(sinkPad)); GRefPtr queueSrcPad = adoptGRef(gst_element_get_static_pad(queue.get(), "src")); GRefPtr appsinkSinkPad = adoptGRef(gst_pad_get_peer(queueSrcPad.get())); GRefPtr 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 lock(m_adapterMutex); gst_adapter_clear(m_frontLeftAdapter); gst_adapter_clear(m_frontRightAdapter); } } // WebCore #endif // ENABLE(WEB_AUDIO) && ENABLE(VIDEO) && USE(GSTREAMER)