Changeset 238606 in webkit

Timestamp:

Nov 28, 2018 4:41:51 AM (5 years ago)

Author:

commit-queue@webkit.org

Message:

[WebRTC][GStreamer] Use a GUniquePtr to hold the GstAudioConverter in our OutgoingAudioSource
​https://bugs.webkit.org/show_bug.cgi?id=192027

Patch by Thibault Saunier <​tsaunier@igalia.com> on 2018-11-28
Reviewed by Xabier Rodriguez-Calvar.

Cleaning up a bit the code.

It is a minor refactoring, no new test required.

• platform/graphics/gstreamer/GUniquePtrGStreamer.h:
• platform/mediastream/gstreamer/RealtimeOutgoingAudioSourceLibWebRTC.cpp:

(WebCore::RealtimeOutgoingAudioSourceLibWebRTC::~RealtimeOutgoingAudioSourceLibWebRTC):
(WebCore::RealtimeOutgoingAudioSourceLibWebRTC::audioSamplesAvailable):
(WebCore::RealtimeOutgoingAudioSourceLibWebRTC::pullAudioData):

• platform/mediastream/gstreamer/RealtimeOutgoingAudioSourceLibWebRTC.h:

Location:

trunk/Source/WebCore

Files:

• ChangeLog (modified) (1 diff)
• platform/graphics/gstreamer/GUniquePtrGStreamer.h (modified) (2 diffs)
• platform/mediastream/gstreamer/RealtimeOutgoingAudioSourceLibWebRTC.cpp (modified) (5 diffs)
• platform/mediastream/gstreamer/RealtimeOutgoingAudioSourceLibWebRTC.h (modified) (1 diff)

trunk/Source/WebCore/ChangeLog

@@ +1 @@
+2018-11-28  Thibault Saunier  <tsaunier@igalia.com>
+
+        [WebRTC][GStreamer] Use a GUniquePtr to hold the GstAudioConverter in our OutgoingAudioSource
+        https://bugs.webkit.org/show_bug.cgi?id=192027
+
+        Reviewed by Xabier Rodriguez-Calvar.
+
+        Cleaning up a bit the code.
+
+        It is a minor refactoring, no new test required.
+
+        * platform/graphics/gstreamer/GUniquePtrGStreamer.h:
+        * platform/mediastream/gstreamer/RealtimeOutgoingAudioSourceLibWebRTC.cpp:
+        (WebCore::RealtimeOutgoingAudioSourceLibWebRTC::~RealtimeOutgoingAudioSourceLibWebRTC):
+        (WebCore::RealtimeOutgoingAudioSourceLibWebRTC::audioSamplesAvailable):
+        (WebCore::RealtimeOutgoingAudioSourceLibWebRTC::pullAudioData):
+        * platform/mediastream/gstreamer/RealtimeOutgoingAudioSourceLibWebRTC.h:
+
 2018-11-28  Thibault Saunier  <tsaunier@igalia.com>
 

trunk/Source/WebCore/platform/graphics/gstreamer/GUniquePtrGStreamer.h

@@ -22 +22 @@
 #if USE(GSTREAMER)
 
+#include <gst/audio/audio.h>
 #include <gst/base/gstbytereader.h>
 #include <gst/base/gstflowcombiner.h>
@@ -39 +40 @@
 WTF_DEFINE_GPTR_DELETER(GstByteReader, gst_byte_reader_free)
 WTF_DEFINE_GPTR_DELETER(GstVideoConverter, gst_video_converter_free)
+WTF_DEFINE_GPTR_DELETER(GstAudioConverter, gst_audio_converter_free)
 
 }

trunk/Source/WebCore/platform/mediastream/gstreamer/RealtimeOutgoingAudioSourceLibWebRTC.cpp

@@ -40 +40 @@
 {
     unobserveSource();
-    if (m_sampleConverter)
-        g_clear_pointer(&m_sampleConverter, gst_audio_converter_free);
+    m_sampleConverter = nullptr;
 }
 
@@ -75 +74 @@
     if (m_sampleConverter && !gst_audio_info_is_equal(m_inputStreamDescription->getInfo(), desc.getInfo())) {
         GST_ERROR_OBJECT(this, "FIXME - Audio format renegotiation is not possible yet!");
-        g_clear_pointer(&m_sampleConverter, gst_audio_converter_free);
+        m_sampleConverter = nullptr;
     }
 
@@ -81 +80 @@
         m_inputStreamDescription = std::unique_ptr<GStreamerAudioStreamDescription>(new GStreamerAudioStreamDescription(desc.getInfo()));
         m_outputStreamDescription = libwebrtcAudioFormat(LibWebRTCAudioFormat::sampleRate, streamDescription.numberOfChannels());
-
-        m_sampleConverter = gst_audio_converter_new(GST_AUDIO_CONVERTER_FLAG_IN_WRITABLE,
+        m_sampleConverter.reset(gst_audio_converter_new(GST_AUDIO_CONVERTER_FLAG_IN_WRITABLE,
             m_inputStreamDescription->getInfo(),
             m_outputStreamDescription->getInfo(),
-            nullptr);
+            nullptr));
     }
 
@@ -108 +106 @@
 
     LockHolder locker(m_adapterMutex);
-    size_t inChunkSampleCount = gst_audio_converter_get_in_frames(m_sampleConverter, outChunkSampleCount);
+    size_t inChunkSampleCount = gst_audio_converter_get_in_frames(m_sampleConverter.get(), outChunkSampleCount);
     size_t inBufferSize = inChunkSampleCount * m_inputStreamDescription->getInfo()->bpf;
 
@@ -129 +127 @@
         gpointer in[1] = { inMap.data() };
         gpointer out[1] = { outMap.data() };
-        if (!gst_audio_converter_samples(m_sampleConverter, static_cast<GstAudioConverterFlags>(0), in, inChunkSampleCount, out, outChunkSampleCount)) {
+        if (!gst_audio_converter_samples(m_sampleConverter.get(), static_cast<GstAudioConverterFlags>(0), in, inChunkSampleCount, out, outChunkSampleCount)) {
             GST_ERROR("Could not convert samples.");
 

trunk/Source/WebCore/platform/mediastream/gstreamer/RealtimeOutgoingAudioSourceLibWebRTC.h

@@ -49 +49 @@
     void pullAudioData() final;
 
-    GstAudioConverter* m_sampleConverter;
+    GUniquePtr<GstAudioConverter> m_sampleConverter;
     std::unique_ptr<GStreamerAudioStreamDescription> m_inputStreamDescription;
     std::unique_ptr<GStreamerAudioStreamDescription> m_outputStreamDescription;