Attachment 349662 Details for Bug 189584 – Patch

[patch] Patch

bug-189584-20180913160042.patch (text/plain), 4.16 KB, created by Philippe Normand on 2018-09-13 07:00:43 PDT

(hide)

Description:

Filename:

MIME Type:

Creator: Philippe Normand

Created: 2018-09-13 07:00:43 PDT

Size: 4.16 KB

patch

obsolete

>Subversion Revision: 235971
>diff --git a/Source/WebCore/ChangeLog b/Source/WebCore/ChangeLog
>index 88320c4bcbd18747d145ec17242163c27b11a2b3..eaae5187d0a225af53f72ecdc28b9758ddaedcbb 100644
>--- a/Source/WebCore/ChangeLog
>+++ b/Source/WebCore/ChangeLog
>@@ -1,3 +1,15 @@
>+2018-09-13  Philippe Normand  <pnormand@igalia.com>
>+
>+        [GStreamer] WebAudio idle/silence playback improvements
>+        https://bugs.webkit.org/show_bug.cgi?id=189584
>+
>+        Reviewed by NOBODY (OOPS!).
>+
>+        Initial patch by: Arunakiran Chennubhatla <arunakiran.chennubhatla@valuelabs.com>
>+
>+        * platform/audio/gstreamer/WebKitWebAudioSourceGStreamer.cpp:
>+        (webKitWebAudioSrcLoop): Allow idle sleep when the audio bus is silent.
>+
> 2018-09-13  Xabier Rodriguez Calvar  <calvaris@igalia.com>
> 
>         [GStreamer][EME] decrypt-key-needed message renamed to drm-cdm-instance-needed
>diff --git a/Source/WebCore/platform/audio/gstreamer/WebKitWebAudioSourceGStreamer.cpp b/Source/WebCore/platform/audio/gstreamer/WebKitWebAudioSourceGStreamer.cpp
>index 55cdb42cc6edd0158fc91645d40998b6682f95fc..dbbf39d39300e43577d633d02b77a5864158ba81 100644
>--- a/Source/WebCore/platform/audio/gstreamer/WebKitWebAudioSourceGStreamer.cpp
>+++ b/Source/WebCore/platform/audio/gstreamer/WebKitWebAudioSourceGStreamer.cpp
>@@ -29,6 +29,8 @@
> #include <gst/app/gstappsrc.h>
> #include <gst/audio/audio-info.h>
> #include <gst/pbutils/missing-plugins.h>
>+#include <wtf/Seconds.h>
>+#include <wtf/WallTime.h>
> #include <wtf/glib/GUniquePtr.h>
> 
> using namespace WebCore;
>@@ -49,6 +51,7 @@ struct _WebKitWebAudioSrcClass {
> #define WEBKIT_WEB_AUDIO_SRC_GET_PRIVATE(obj) (G_TYPE_INSTANCE_GET_PRIVATE((obj), WEBKIT_TYPE_WEBAUDIO_SRC, WebKitWebAudioSourcePrivate))
> struct _WebKitWebAudioSourcePrivate {
>     gfloat sampleRate;
>+    Seconds silenceStartTime;
>     AudioBus* bus;
>     AudioIOCallback* provider;
>     guint framesToPull;
>@@ -352,6 +355,17 @@ static void webKitWebAudioSrcLoop(WebKitWebAudioSrc* src)
>     // FIXME: Add support for local/live audio input.
>     priv->provider->render(nullptr, priv->bus, priv->framesToPull);
> 
>+    if (priv->bus->isSilent()) {
>+        priv->numberOfSamples -= priv->framesToPull;
>+        if (!priv->silenceStartTime.value())
>+            priv->silenceStartTime = WTF::WallTime::now().secondsSinceEpoch();
>+    } else if (priv->silenceStartTime) {
>+        auto delta = WTF::WallTime::now().secondsSinceEpoch() - priv->silenceStartTime;
>+        priv->numberOfSamples += delta.seconds() * priv->sampleRate;
>+        priv->silenceStartTime = WTF::Seconds::nan();
>+    }
>+
>+    GstFlowReturn ret = GST_FLOW_OK;
>     ASSERT(channelBufferList.size() == priv->sources.size());
>     bool failed = false;
>     for (unsigned i = 0; i < priv->sources.size(); ++i) {
>@@ -366,15 +380,20 @@ static void webKitWebAudioSrcLoop(WebKitWebAudioSrc* src)
>             continue;
> 
>         auto& appsrc = priv->sources[i];
>-        // Leak the buffer ref, because gst_app_src_push_buffer steals it.
>-        GstFlowReturn ret = gst_app_src_push_buffer(GST_APP_SRC(appsrc.get()), buffer.leakRef());
>-        if (ret != GST_FLOW_OK) {
>-            // FLUSHING and EOS are not errors.
>-            if (ret < GST_FLOW_EOS || ret == GST_FLOW_NOT_LINKED)
>-                GST_ELEMENT_ERROR(src, CORE, PAD, ("Internal WebAudioSrc error"), ("Failed to push buffer on %s flow: %s", GST_OBJECT_NAME(appsrc.get()), gst_flow_get_name(ret)));
>-            gst_task_stop(src->priv->task.get());
>-            failed = true;
>-        }
>+
>+        if ((ret == GST_FLOW_OK) && (!priv->bus->isSilent())) {
>+
>+            // Leak the buffer ref, because gst_app_src_push_buffer steals it.
>+            ret = gst_app_src_push_buffer(GST_APP_SRC(appsrc.get()), buffer.leakRef());
>+            if (ret != GST_FLOW_OK) {
>+                // FLUSHING and EOS are not errors.
>+                if (ret < GST_FLOW_EOS || ret == GST_FLOW_NOT_LINKED)
>+                    GST_ELEMENT_ERROR(src, CORE, PAD, ("Internal WebAudioSrc error"), ("Failed to push buffer on %s flow: %s", GST_OBJECT_NAME(appsrc.get()), gst_flow_get_name(ret)));
>+                gst_task_stop(src->priv->task.get());
>+                failed = true;
>+            }
>+        } else
>+            g_usleep(1000);
>     }
> }
>

Flags:

ews-watchlist: commit-queue-

Actions: View | Formatted Diff | Diff

Attachments on bug 189584: 349662 | 349674