Modified: trunk/Source/WebCore/platform/audio/gstreamer/AudioSourceProviderGStreamer.cpp (271289 => 271290)
--- trunk/Source/WebCore/platform/audio/gstreamer/AudioSourceProviderGStreamer.cpp 2021-01-08 10:28:50 UTC (rev 271289)
+++ trunk/Source/WebCore/platform/audio/gstreamer/AudioSourceProviderGStreamer.cpp 2021-01-08 13:16:25 UTC (rev 271290)
@@ -67,10 +67,12 @@
{
auto available = gst_adapter_available(adapter);
if (!available) {
+ GST_TRACE("Adapter empty, silencing bus");
bus->zero();
return;
}
+ GST_TRACE("%zu samples available for channel %d (%zu frames requested)", available, channelNumber, framesToProcess);
size_t bytes = framesToProcess * sizeof(float);
if (available >= bytes) {
gst_adapter_copy(adapter, bus->channel(channelNumber)->mutableData(), 0, bytes);
@@ -150,6 +152,7 @@
void AudioSourceProviderGStreamer::provideInput(AudioBus* bus, size_t framesToProcess)
{
+ GST_TRACE("Fetching buffers from adapters");
auto locker = holdLock(m_adapterMutex);
for (auto& it : m_adapters)
copyGStreamerBuffersToAudioChannel(it.value.get(), bus, it.key - 1, framesToProcess);
@@ -157,6 +160,7 @@
GstFlowReturn AudioSourceProviderGStreamer::handleSample(GstAppSink* sink, bool isPreroll)
{
+ GST_TRACE("Pulling audio sample from the sink");
auto sample = adoptGRef(isPreroll ? gst_app_sink_try_pull_preroll(sink, 0) : gst_app_sink_try_pull_sample(sink, 0));
if (!sample)
return gst_app_sink_is_eos(sink) ? GST_FLOW_EOS : GST_FLOW_ERROR;
@@ -168,10 +172,13 @@
if (!buffer)
return GST_FLOW_ERROR;
+ GST_TRACE("Storing audio sample %" GST_PTR_FORMAT, sample.get());
{
auto locker = holdLock(m_adapterMutex);
GQuark quark = g_quark_from_static_string("channel-id");
int channelId = GPOINTER_TO_INT(g_object_get_qdata(G_OBJECT(sink), quark));
+ GST_DEBUG("Channel ID: %d", channelId);
+
auto result = m_adapters.ensure(channelId, [&] {
return gst_adapter_new();
});
@@ -189,18 +196,10 @@
if (m_client == client)
return;
+ GST_DEBUG("Setting up client %p (previous: %p)", client, m_client);
+ bool previousClientWasValid = m_client;
m_client = client;
-#if ENABLE(MEDIA_STREAM)
- if (m_pipeline)
- gst_element_set_state(m_pipeline.get(), m_client ? GST_STATE_PLAYING : GST_STATE_NULL);
-#endif
-
- // FIXME: This early return should ideally be replaced by a removal of the m_audioSinkBin from
- // its parent pipeline. https://bugs.webkit.org/show_bug.cgi?id=219245
- if (!m_client)
- return;
-
// 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
@@ -207,46 +206,80 @@
// supposedly configured already by application side.
auto volumeElement = adoptGRef(gst_bin_get_by_name(GST_BIN_CAST(m_audioSinkBin.get()), "volume"));
- if (volumeElement)
- g_object_set(volumeElement.get(), "mute", TRUE, nullptr);
+ if (volumeElement) {
+ bool shouldMute = m_client;
+ g_object_set(volumeElement.get(), "mute", shouldMute, 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");
+ auto audioTee = adoptGRef(gst_bin_get_by_name(GST_BIN_CAST(m_audioSinkBin.get()), "audioTee"));
+ if (!m_client || previousClientWasValid) {
+ auto audioQueue = adoptGRef(gst_bin_get_by_name(GST_BIN_CAST(m_audioSinkBin.get()), "queue"));
+ auto audioConvert = adoptGRef(gst_bin_get_by_name(GST_BIN_CAST(m_audioSinkBin.get()), "audioconvert"));
+ auto audioResample = adoptGRef(gst_bin_get_by_name(GST_BIN_CAST(m_audioSinkBin.get()), "audioresample"));
+ auto capsFilter = adoptGRef(gst_bin_get_by_name(GST_BIN_CAST(m_audioSinkBin.get()), "capsfilter"));
+ auto deInterleave = adoptGRef(gst_bin_get_by_name(GST_BIN_CAST(m_audioSinkBin.get()), "deinterleave"));
+ auto queueSinkPad = adoptGRef(gst_element_get_static_pad(audioQueue.get(), "sink"));
+ auto teeSrcPad = adoptGRef(gst_pad_get_peer(queueSinkPad.get()));
- 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);
+ GST_DEBUG("Cleaning up audio deinterleave chain");
+ gst_element_set_locked_state(m_audioSinkBin.get(), true);
- auto caps = adoptGRef(gst_caps_new_simple("audio/x-raw", "rate", G_TYPE_INT, static_cast<int>(gSampleBitRate),
- "format", G_TYPE_STRING, GST_AUDIO_NE(F32), "layout", G_TYPE_STRING, "interleaved", nullptr));
- g_object_set(capsFilter, "caps", caps.get(), nullptr);
+ gst_element_set_state(audioQueue.get(), GST_STATE_NULL);
+ gst_element_set_state(audioConvert.get(), GST_STATE_NULL);
+ gst_element_set_state(audioResample.get(), GST_STATE_NULL);
+ gst_element_set_state(capsFilter.get(), GST_STATE_NULL);
+ gst_element_set_state(deInterleave.get(), GST_STATE_NULL);
+ gst_element_unlink_many(audioTee.get(), audioQueue.get(), audioConvert.get(), audioResample.get(), capsFilter.get(), deInterleave.get(), nullptr);
+ gst_element_set_locked_state(m_audioSinkBin.get(), false);
+ gst_bin_remove_many(GST_BIN_CAST(m_audioSinkBin.get()), audioQueue.get(), audioConvert.get(), audioResample.get(), capsFilter.get(), deInterleave.get(), nullptr);
+ gst_element_release_request_pad(audioTee.get(), teeSrcPad.get());
+ }
- gst_bin_add_many(GST_BIN_CAST(m_audioSinkBin.get()), audioQueue, audioConvert, audioResample, capsFilter, deInterleave, nullptr);
+ if (m_client) {
+ // The audioconvert and audioresample elements are needed to
+ // ensure deinterleave and the sinks downstream receive buffers in
+ // the format specified by the capsfilter.
+ auto* audioQueue = gst_element_factory_make("queue", "queue");
+ auto* audioConvert = gst_element_factory_make("audioconvert", "audioconvert");
+ auto* audioResample = gst_element_factory_make("audioresample", "audioresample");
+ auto* capsFilter = gst_element_factory_make("capsfilter", "capsfilter");
+ auto* deInterleave = gst_element_factory_make("deinterleave", "deinterleave");
- auto audioTee = adoptGRef(gst_bin_get_by_name(GST_BIN(m_audioSinkBin.get()), "audioTee"));
+ GST_DEBUG("Setting up audio deinterleave chain");
+ 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);
- // 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);
+ auto caps = adoptGRef(gst_caps_new_simple("audio/x-raw", "rate", G_TYPE_INT, static_cast<int>(gSampleBitRate),
+ "format", G_TYPE_STRING, GST_AUDIO_NE(F32), "layout", G_TYPE_STRING, "interleaved", nullptr));
+ g_object_set(capsFilter, "caps", caps.get(), nullptr);
- 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);
+ gst_bin_add_many(GST_BIN_CAST(m_audioSinkBin.get()), audioQueue, audioConvert, audioResample, capsFilter, deInterleave, nullptr);
+
+ // 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);
+ }
+
+ m_deinterleaveSourcePads = 0;
+ clearAdapters();
+#if ENABLE(MEDIA_STREAM)
+ if (m_pipeline)
+ gst_element_set_state(m_pipeline.get(), m_client ? GST_STATE_PLAYING : GST_STATE_NULL);
+#endif
}
void AudioSourceProviderGStreamer::handleNewDeinterleavePad(GstPad* pad)
@@ -305,6 +338,9 @@
void AudioSourceProviderGStreamer::handleRemovedDeinterleavePad(GstPad* pad)
{
+ if (GST_PAD_DIRECTION(pad) != GST_PAD_SRC)
+ return;
+
GST_DEBUG("Pad %" GST_PTR_FORMAT " gone", pad);
m_deinterleaveSourcePads--;
@@ -320,7 +356,7 @@
void AudioSourceProviderGStreamer::deinterleavePadsConfigured()
{
- GST_DEBUG("Deinterleave configured, notifying client");
+ GST_DEBUG("Deinterleave configured with %d channels, notifying client", m_deinterleaveSourcePads);
m_notifier->notify(MainThreadNotification::DeinterleavePadsConfigured, [numberOfChannels = m_deinterleaveSourcePads, sampleRate = gSampleBitRate, client = m_client] {
if (client)
client->setFormat(numberOfChannels, sampleRate);
