webkitgtk-2.16.5HEADwebkitgtk-2.16.5master

1 files changed, 162 insertions, 0 deletions

diff --git a/Source/WebCore/platform/mediastream/libwebrtc/LibWebRTCAudioModule.h b/Source/WebCore/platform/mediastream/libwebrtc/LibWebRTCAudioModule.h
new file mode 100644
index 000000000..95988609d
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+++ b/Source/WebCore/platform/mediastream/libwebrtc/LibWebRTCAudioModule.h
@@ -0,0 +1,162 @@
+/*
+ * 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.
+ */
+
+#pragma once
+
+#if USE(LIBWEBRTC)
+
+#include "LibWebRTCMacros.h"
+#include <webrtc/base/messagehandler.h>
+#include <webrtc/base/thread.h>
+#include <webrtc/modules/audio_device/include/audio_device.h>
+
+namespace WebCore {
+
+// LibWebRTCAudioModule is pulling streamed data to ensure audio data is passed to the audio track.
+class LibWebRTCAudioModule final : public webrtc::AudioDeviceModule, private rtc::MessageHandler {
+public:
+    LibWebRTCAudioModule();
+
+private:
+    template<typename U> U shouldNotBeCalled(U value) const
+    {
+        ASSERT_NOT_REACHED();
+        return value;
+    }
+
+    int32_t AddRef() const final { return 1; }
+    int32_t Release() const final { return 1; }
+    void OnMessage(rtc::Message*);
+
+    // webrtc::AudioDeviceModule API
+    int32_t StartPlayout() final;
+    int32_t StopPlayout() final;
+    int32_t RegisterAudioCallback(webrtc::AudioTransport*) final;
+    bool Playing() const final { return m_isPlaying; }
+
+    int64_t TimeUntilNextProcess() final { return std::numeric_limits<int64_t>::max(); }
+    void Process() final { }
+    int32_t ActiveAudioLayer(AudioLayer*) const final { return shouldNotBeCalled(-1); }
+    ErrorCode LastError() const final { return kAdmErrNone; }
+    int32_t RegisterEventObserver(webrtc::AudioDeviceObserver*) final { return 0; }
+    int32_t Init() final { return 0; }
+    int32_t Terminate() final { return 0; }
+    bool Initialized() const final { return true; }
+    int16_t PlayoutDevices() final { return 0; }
+    int16_t RecordingDevices() final { return 0; }
+    int32_t PlayoutDeviceName(uint16_t, char[webrtc::kAdmMaxDeviceNameSize], char[webrtc::kAdmMaxGuidSize]) final { return 0; }
+    int32_t RecordingDeviceName(uint16_t, char[webrtc::kAdmMaxDeviceNameSize], char[webrtc::kAdmMaxGuidSize]) final { return 0; }
+    int32_t SetPlayoutDevice(uint16_t) final { return 0; }
+    int32_t SetPlayoutDevice(WindowsDeviceType) final { return 0; }
+    int32_t SetRecordingDevice(uint16_t) final { return 0; }
+    int32_t SetRecordingDevice(WindowsDeviceType) final { return 0; }
+    int32_t PlayoutIsAvailable(bool*) final { return shouldNotBeCalled(-1); }
+    int32_t InitPlayout() final { return 0; }
+    bool PlayoutIsInitialized() const final { return true; }
+    int32_t RecordingIsAvailable(bool*) final { return shouldNotBeCalled(-1); }
+    int32_t InitRecording() final { return 0; }
+    bool RecordingIsInitialized() const final { return false; }
+    int32_t StartRecording() final { return 0; }
+    int32_t StopRecording() final { return 0;  }
+    bool Recording() const final { return 0;  }
+    int32_t SetAGC(bool) final { return 0;  }
+    bool AGC() const final { return shouldNotBeCalled(0);  }
+    int32_t SetWaveOutVolume(uint16_t, uint16_t) final { return shouldNotBeCalled(-1); }
+    int32_t WaveOutVolume(uint16_t*, uint16_t*) const final { return shouldNotBeCalled(-1); }
+    int32_t InitSpeaker() final { return 0; }
+    bool SpeakerIsInitialized() const final { return false; }
+    int32_t InitMicrophone() final { return 0; }
+    bool MicrophoneIsInitialized() const final { return false; }
+    int32_t SpeakerVolumeIsAvailable(bool*) final { return shouldNotBeCalled(-1); }
+    int32_t SetSpeakerVolume(uint32_t) final { return shouldNotBeCalled(-1); }
+    int32_t SpeakerVolume(uint32_t*) const final { return shouldNotBeCalled(-1); }
+    int32_t MaxSpeakerVolume(uint32_t*) const final { return shouldNotBeCalled(-1); }
+    int32_t MinSpeakerVolume(uint32_t*) const final { return shouldNotBeCalled(-1); }
+    int32_t SpeakerVolumeStepSize(uint16_t*) const final { return shouldNotBeCalled(-1); }
+    int32_t MicrophoneVolumeIsAvailable(bool*) final { return shouldNotBeCalled(-1); }
+    int32_t SetMicrophoneVolume(uint32_t) final { return shouldNotBeCalled(-1); }
+    int32_t MicrophoneVolume(uint32_t*) const final { return shouldNotBeCalled(-1); }
+    int32_t MaxMicrophoneVolume(uint32_t*) const final { return shouldNotBeCalled(-1); }
+    int32_t MinMicrophoneVolume(uint32_t*) const final { return shouldNotBeCalled(-1); }
+    int32_t MicrophoneVolumeStepSize(uint16_t*) const final { return shouldNotBeCalled(-1); }
+    int32_t SpeakerMuteIsAvailable(bool*) final { return shouldNotBeCalled(-1); }
+    int32_t SetSpeakerMute(bool) final { return shouldNotBeCalled(-1); }
+    int32_t SpeakerMute(bool*) const final { return shouldNotBeCalled(-1); }
+    int32_t MicrophoneMuteIsAvailable(bool*) final { return shouldNotBeCalled(-1); }
+    int32_t SetMicrophoneMute(bool) final { return shouldNotBeCalled(-1); }
+    int32_t MicrophoneMute(bool*) const final { return shouldNotBeCalled(-1); }
+    int32_t MicrophoneBoostIsAvailable(bool*) final { return shouldNotBeCalled(-1); }
+    int32_t SetMicrophoneBoost(bool) final { return shouldNotBeCalled(-1); }
+    int32_t MicrophoneBoost(bool*) const final { return shouldNotBeCalled(-1); }
+    int32_t StereoPlayoutIsAvailable(bool* available) const final { *available = false; return 0; }
+    int32_t SetStereoPlayout(bool) final { return 0; }
+    int32_t StereoPlayout(bool*) const final { return shouldNotBeCalled(-1); }
+    int32_t StereoRecordingIsAvailable(bool* available) const final { *available = false; return 0;  }
+    int32_t SetStereoRecording(bool) final { return 0;  }
+    int32_t StereoRecording(bool*) const final { return shouldNotBeCalled(-1); }
+    int32_t SetRecordingChannel(const ChannelType) final { return 0; }
+    int32_t RecordingChannel(ChannelType*) const final { return shouldNotBeCalled(-1); }
+    int32_t SetPlayoutBuffer(const BufferType, uint16_t) final { return shouldNotBeCalled(-1); }
+    int32_t PlayoutBuffer(BufferType*, uint16_t*) const final { return shouldNotBeCalled(-1); }
+    int32_t PlayoutDelay(uint16_t* delay) const final { *delay = 0; return 0; }
+    int32_t RecordingDelay(uint16_t*) const final { return shouldNotBeCalled(-1); }
+    int32_t CPULoad(uint16_t*) const final { return shouldNotBeCalled(-1); }
+    int32_t StartRawOutputFileRecording(const char[webrtc::kAdmMaxFileNameSize]) final { return shouldNotBeCalled(-1); }
+    int32_t StopRawOutputFileRecording() final { return shouldNotBeCalled(-1); }
+    int32_t StartRawInputFileRecording(const char[webrtc::kAdmMaxFileNameSize]) final { return shouldNotBeCalled(-1); }
+    int32_t StopRawInputFileRecording() final { return shouldNotBeCalled(-1); }
+    int32_t SetRecordingSampleRate(const uint32_t) final { return shouldNotBeCalled(-1); }
+    int32_t RecordingSampleRate(uint32_t*) const final { return shouldNotBeCalled(-1); }
+    int32_t SetPlayoutSampleRate(const uint32_t) final { return shouldNotBeCalled(-1); }
+    int32_t PlayoutSampleRate(uint32_t*) const final { return shouldNotBeCalled(-1); }
+    int32_t ResetAudioDevice() final { return shouldNotBeCalled(-1); }
+    int32_t SetLoudspeakerStatus(bool) final { return shouldNotBeCalled(-1); }
+    int32_t GetLoudspeakerStatus(bool*) const final { return shouldNotBeCalled(-1); }
+    bool BuiltInAECIsAvailable() const final { return false; }
+    bool BuiltInAGCIsAvailable() const final { return false;  }
+    bool BuiltInNSIsAvailable() const final { return false;  }
+    int32_t EnableBuiltInAEC(bool) final { return shouldNotBeCalled(-1); }
+    int32_t EnableBuiltInAGC(bool) final { return shouldNotBeCalled(-1); }
+    int32_t EnableBuiltInNS(bool) final { return shouldNotBeCalled(-1); }
+
+#if defined(WEBRTC_IOS)
+    int GetPlayoutAudioParameters(webrtc::AudioParameters*) const final { return shouldNotBeCalled(-1); }
+    int GetRecordAudioParameters(webrtc::AudioParameters*) const final { return shouldNotBeCalled(-1); }
+#endif
+
+private:
+    void StartPlayoutOnAudioThread();
+
+    void PollFromSource();
+
+    std::unique_ptr<rtc::Thread> m_audioTaskRunner;
+
+    bool m_isPlaying = false;
+    webrtc::AudioTransport* m_audioTransport = nullptr;
+};
+
+} // namespace WebCore
+
+#endif // USE(LIBWEBRTC)