Cherry-pick 278306@main (9f7fcac). https://bugs.webkit.org/show_bug.c…

…gi?id=273555

    [GStreamer][WebAudio] Misc improvements in playback pipeline
    https://bugs.webkit.org/show_bug.cgi?id=273555

    Reviewed by Xabier Rodriguez-Calvar.

    A small queue is needed before the sink, by design. In the webaudiosrc element timestamping was
    improved, silent buffers are now flagged as droppable and the render task starts only when going to
    PLAYING state.

    * Source/WebCore/platform/audio/gstreamer/AudioDestinationGStreamer.cpp:
    (WebCore::AudioDestination::create):
    (WebCore::AudioDestinationGStreamer::AudioDestinationGStreamer):
    * Source/WebCore/platform/audio/gstreamer/WebKitWebAudioSourceGStreamer.cpp:
    (webKitWebAudioSrcConstructed):
    (webKitWebAudioSrcRenderAndPushFrames):
    (webKitWebAudioSrcChangeState):

    Canonical link: https://commits.webkit.org/278306@main

Canonical link: https://commits.webkit.org/274313.288@webkitglib/2.44

Source/WebCore/platform/audio/gstreamer/AudioDestinationGStreamer.cpp

@@ -23,7 +23,6 @@
 
 #include "AudioDestinationGStreamer.h"
 
-#include "AudioChannel.h"
 #include "AudioSourceProvider.h"
 #include "AudioUtilities.h"
 #include "GStreamerCommon.h"
@@ -170,12 +169,17 @@ AudioDestinationGStreamer::AudioDestinationGStreamer(AudioIOCallback& callback,
 
     GstElement* audioConvert = makeGStreamerElement("audioconvert", nullptr);
     GstElement* audioResample = makeGStreamerElement("audioresample", nullptr);
-    gst_bin_add_many(GST_BIN_CAST(m_pipeline.get()), m_src.get(), audioConvert, audioResample, audioSink.get(), nullptr);
 
-    // Link src pads from webkitAudioSrc to audioConvert ! audioResample ! autoaudiosink.
+    auto queue = gst_element_factory_make("queue", nullptr);
+    g_object_set(queue, "max-size-buffers", 2, "max-size-bytes", 0, "max-size-time", 0, nullptr);
+
+    gst_bin_add_many(GST_BIN_CAST(m_pipeline.get()), m_src.get(), audioConvert, audioResample, queue, audioSink.get(), nullptr);
+
+    // Link src pads from webkitAudioSrc to audioConvert ! audioResample ! queue ! autoaudiosink.
     gst_element_link_pads_full(m_src.get(), "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", audioSink.get(), "sink", GST_PAD_LINK_CHECK_NOTHING);
+    gst_element_link_pads_full(audioResample, "src", queue, "sink", GST_PAD_LINK_CHECK_NOTHING);
+    gst_element_link_pads_full(queue, "src", audioSink.get(), "sink", GST_PAD_LINK_CHECK_NOTHING);
 }
 
 AudioDestinationGStreamer::~AudioDestinationGStreamer()

Source/WebCore/platform/audio/gstreamer/WebKitWebAudioSourceGStreamer.cpp

@@ -216,7 +216,7 @@ static void webKitWebAudioSrcConstructed(GObject* object)
     priv->source = makeGStreamerElement("appsrc", "webaudioSrc");
 
     // Configure the appsrc for minimal latency.
-    g_object_set(priv->source.get(), "block", TRUE, "blocksize", priv->bufferSize, "format", GST_FORMAT_TIME, nullptr);
+    g_object_set(priv->source.get(), "block", TRUE, "blocksize", priv->bufferSize, "format", GST_FORMAT_TIME, "is-live", TRUE, nullptr);
 
     gst_bin_add(GST_BIN(src), priv->source.get());
     // appsrc's src pad is the only visible pad of our element.
@@ -314,6 +314,9 @@ static void webKitWebAudioSrcRenderAndPushFrames(const GRefPtr<GstElement>& elem
         priv->dispatchCondition.notifyOne();
     });
 
+    if (GST_STATE(element.get()) < GST_STATE_PAUSED)
+        return;
+
     if (!priv->destination)
         return;
 
@@ -323,14 +326,12 @@ static void webKitWebAudioSrcRenderAndPushFrames(const GRefPtr<GstElement>& elem
 
     GstClockTime timestamp = gst_util_uint64_scale(priv->numberOfSamples, GST_SECOND, priv->sampleRate);
     priv->numberOfSamples += priv->framesToPull;
-    GstClockTime duration = gst_util_uint64_scale(priv->numberOfSamples, GST_SECOND, priv->sampleRate) - timestamp;
+    GstClockTime duration = gst_util_uint64_scale(priv->framesToPull, GST_SECOND, priv->sampleRate);
 
     AudioIOPosition outputTimestamp;
-    if (auto clock = adoptGRef(gst_element_get_clock(element.get()))) {
-        auto clockTime = gst_clock_get_time(clock.get());
-        outputTimestamp.position = Seconds::fromNanoseconds(timestamp);
-        outputTimestamp.timestamp = MonotonicTime::fromRawSeconds(static_cast<double>((g_get_monotonic_time() + GST_TIME_AS_USECONDS(clockTime)) / 1000000.0));
-    }
+    outputTimestamp.position = Seconds::fromNanoseconds(timestamp);
+    auto now = static_cast<double>((g_get_monotonic_time() + GST_TIME_AS_USECONDS(timestamp)) / 1000000.0);
+    outputTimestamp.timestamp = MonotonicTime::fromRawSeconds(now);
 
     // FIXME: Add support for local/live audio input.
     if (priv->bus)
@@ -341,11 +342,13 @@ static void webKitWebAudioSrcRenderAndPushFrames(const GRefPtr<GstElement>& elem
         priv->hasRenderedAudibleFrame = true;
     }
 
-    GST_BUFFER_TIMESTAMP(buffer.get()) = outputTimestamp.position.nanoseconds();
+    GST_BUFFER_TIMESTAMP(buffer.get()) = timestamp;
     GST_BUFFER_DURATION(buffer.get()) = duration;
 
-    if (priv->bus->isSilent())
+    if (priv->bus->isSilent()) {
         GST_BUFFER_FLAG_SET(buffer.get(), GST_BUFFER_FLAG_GAP);
+        GST_BUFFER_FLAG_SET(buffer.get(), GST_BUFFER_FLAG_DROPPABLE);
+    }
 
     // Leak the buffer ref, because gst_app_src_push_buffer steals it.
     GstFlowReturn ret = gst_app_src_push_buffer(GST_APP_SRC(priv->source.get()), buffer.leakRef());
@@ -412,6 +415,10 @@ static GstStateChangeReturn webKitWebAudioSrcChangeState(GstElement* element, Gs
         gst_buffer_pool_set_config(priv->pool.get(), config);
         if (!gst_buffer_pool_set_active(priv->pool.get(), TRUE))
             return GST_STATE_CHANGE_FAILURE;
+        webKitWebAudioSrcRenderIteration(src);
+        break;
+    }
+    case GST_STATE_CHANGE_PAUSED_TO_PLAYING: {
         if (!gst_task_start(priv->task.get()))
             return GST_STATE_CHANGE_FAILURE;
         break;