webkitgtk-2.16.5HEADwebkitgtk-2.16.5master

1 files changed, 101 insertions, 0 deletions

diff --git a/Source/WebCore/platform/mediastream/libwebrtc/LibWebRTCAudioModule.cpp b/Source/WebCore/platform/mediastream/libwebrtc/LibWebRTCAudioModule.cpp
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+++ b/Source/WebCore/platform/mediastream/libwebrtc/LibWebRTCAudioModule.cpp
@@ -0,0 +1,101 @@
+/*
+ * Copyright (C) 2017 Apple Inc. All rights reserved.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions
+ * are met:
+ * 1. Redistributions of source code must retain the above copyright
+ *    notice, this list of conditions and the following disclaimer.
+ * 2. Redistributions in binary form must reproduce the above copyright
+ *    notice, this list of conditions and the following disclaimer in the
+ *    documentation and/or other materials provided with the distribution.
+ *
+ * THIS SOFTWARE IS PROVIDED BY APPLE INC. AND ITS CONTRIBUTORS ``AS IS''
+ * AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO,
+ * THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR
+ * PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL APPLE INC. OR ITS CONTRIBUTORS
+ * BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+ * CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+ * SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
+ * INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
+ * CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
+ * ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF
+ * THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#include "config.h"
+#include "LibWebRTCAudioModule.h"
+
+#if USE(LIBWEBRTC)
+
+namespace WebCore {
+
+LibWebRTCAudioModule::LibWebRTCAudioModule()
+    : m_audioTaskRunner(rtc::Thread::Create())
+{
+    m_audioTaskRunner->Start();
+}
+
+int32_t LibWebRTCAudioModule::RegisterAudioCallback(webrtc::AudioTransport* audioTransport)
+{
+    m_audioTransport = audioTransport;
+    return 0;
+}
+
+void LibWebRTCAudioModule::OnMessage(rtc::Message* message)
+{
+    ASSERT_UNUSED(message, message->message_id == 1);
+    StartPlayoutOnAudioThread();
+}
+
+int32_t LibWebRTCAudioModule::StartPlayout()
+{
+    if (!m_isPlaying && m_audioTaskRunner) {
+        m_audioTaskRunner->Post(RTC_FROM_HERE, this, 1);
+        m_isPlaying = true;
+    }
+    return 0;
+}
+
+int32_t LibWebRTCAudioModule::StopPlayout()
+{
+    if (m_isPlaying)
+        m_isPlaying = false;
+    return 0;
+}
+
+// libwebrtc uses 10ms frames.
+const unsigned samplingRate = 48000;
+const unsigned frameLengthMs = 10;
+const unsigned samplesPerFrame = samplingRate * frameLengthMs / 1000;
+const unsigned pollSamples = 5;
+const unsigned pollInterval = 5 * frameLengthMs;
+const unsigned channels = 2;
+const unsigned bytesPerSample = 2;
+
+void LibWebRTCAudioModule::StartPlayoutOnAudioThread()
+{
+    while (true) {
+        PollFromSource();
+        m_audioTaskRunner->SleepMs(pollInterval);
+        if (!m_isPlaying)
+            return;
+    }
+}
+
+void LibWebRTCAudioModule::PollFromSource()
+{
+    if (!m_audioTransport)
+        return;
+
+    for (unsigned i = 0; i < pollSamples; i++) {
+        int64_t elapsedTime = -1;
+        int64_t ntpTime = -1;
+        char data[(bytesPerSample * channels * samplesPerFrame)];
+        m_audioTransport->PullRenderData(bytesPerSample * 8, samplingRate, channels, samplesPerFrame, data, &elapsedTime, &ntpTime);
+    }
+}
+
+} // namespace WebCore
+
+#endif // USE(LIBWEBRTC)