272 lines
11 KiB
C
272 lines
11 KiB
C
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/*
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* Copyright 2012 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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// This class implements an AudioCaptureModule that can be used to detect if
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// audio is being received properly if it is fed by another AudioCaptureModule
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// in some arbitrary audio pipeline where they are connected. It does not play
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// out or record any audio so it does not need access to any hardware and can
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// therefore be used in the gtest testing framework.
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// Note P postfix of a function indicates that it should only be called by the
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// processing thread.
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#ifndef WEBRTC_API_TEST_FAKEAUDIOCAPTUREMODULE_H_
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#define WEBRTC_API_TEST_FAKEAUDIOCAPTUREMODULE_H_
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#include <memory>
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#include "webrtc/base/basictypes.h"
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#include "webrtc/base/criticalsection.h"
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#include "webrtc/base/messagehandler.h"
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#include "webrtc/base/scoped_ref_ptr.h"
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#include "webrtc/common_types.h"
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#include "webrtc/modules/audio_device/include/audio_device.h"
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namespace rtc {
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class Thread;
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} // namespace rtc
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class FakeAudioCaptureModule
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: public webrtc::AudioDeviceModule,
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public rtc::MessageHandler {
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public:
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typedef uint16_t Sample;
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// The value for the following constants have been derived by running VoE
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// using a real ADM. The constants correspond to 10ms of mono audio at 44kHz.
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static const size_t kNumberSamples = 440;
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static const size_t kNumberBytesPerSample = sizeof(Sample);
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// Creates a FakeAudioCaptureModule or returns NULL on failure.
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static rtc::scoped_refptr<FakeAudioCaptureModule> Create();
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// Returns the number of frames that have been successfully pulled by the
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// instance. Note that correctly detecting success can only be done if the
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// pulled frame was generated/pushed from a FakeAudioCaptureModule.
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int frames_received() const;
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// Following functions are inherited from webrtc::AudioDeviceModule.
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// Only functions called by PeerConnection are implemented, the rest do
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// nothing and return success. If a function is not expected to be called by
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// PeerConnection an assertion is triggered if it is in fact called.
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int64_t TimeUntilNextProcess() override;
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void Process() override;
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int32_t ActiveAudioLayer(AudioLayer* audio_layer) const override;
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ErrorCode LastError() const override;
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int32_t RegisterEventObserver(
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webrtc::AudioDeviceObserver* event_callback) override;
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// Note: Calling this method from a callback may result in deadlock.
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int32_t RegisterAudioCallback(
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webrtc::AudioTransport* audio_callback) override;
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int32_t Init() override;
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int32_t Terminate() override;
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bool Initialized() const override;
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int16_t PlayoutDevices() override;
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int16_t RecordingDevices() override;
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int32_t PlayoutDeviceName(uint16_t index,
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char name[webrtc::kAdmMaxDeviceNameSize],
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char guid[webrtc::kAdmMaxGuidSize]) override;
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int32_t RecordingDeviceName(uint16_t index,
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char name[webrtc::kAdmMaxDeviceNameSize],
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char guid[webrtc::kAdmMaxGuidSize]) override;
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int32_t SetPlayoutDevice(uint16_t index) override;
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int32_t SetPlayoutDevice(WindowsDeviceType device) override;
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int32_t SetRecordingDevice(uint16_t index) override;
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int32_t SetRecordingDevice(WindowsDeviceType device) override;
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int32_t PlayoutIsAvailable(bool* available) override;
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int32_t InitPlayout() override;
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bool PlayoutIsInitialized() const override;
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int32_t RecordingIsAvailable(bool* available) override;
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int32_t InitRecording() override;
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bool RecordingIsInitialized() const override;
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int32_t StartPlayout() override;
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int32_t StopPlayout() override;
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bool Playing() const override;
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int32_t StartRecording() override;
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int32_t StopRecording() override;
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bool Recording() const override;
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int32_t SetAGC(bool enable) override;
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bool AGC() const override;
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int32_t SetWaveOutVolume(uint16_t volume_left,
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uint16_t volume_right) override;
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int32_t WaveOutVolume(uint16_t* volume_left,
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uint16_t* volume_right) const override;
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int32_t InitSpeaker() override;
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bool SpeakerIsInitialized() const override;
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int32_t InitMicrophone() override;
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bool MicrophoneIsInitialized() const override;
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int32_t SpeakerVolumeIsAvailable(bool* available) override;
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int32_t SetSpeakerVolume(uint32_t volume) override;
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int32_t SpeakerVolume(uint32_t* volume) const override;
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int32_t MaxSpeakerVolume(uint32_t* max_volume) const override;
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int32_t MinSpeakerVolume(uint32_t* min_volume) const override;
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int32_t SpeakerVolumeStepSize(uint16_t* step_size) const override;
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int32_t MicrophoneVolumeIsAvailable(bool* available) override;
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int32_t SetMicrophoneVolume(uint32_t volume) override;
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int32_t MicrophoneVolume(uint32_t* volume) const override;
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int32_t MaxMicrophoneVolume(uint32_t* max_volume) const override;
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int32_t MinMicrophoneVolume(uint32_t* min_volume) const override;
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int32_t MicrophoneVolumeStepSize(uint16_t* step_size) const override;
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int32_t SpeakerMuteIsAvailable(bool* available) override;
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int32_t SetSpeakerMute(bool enable) override;
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int32_t SpeakerMute(bool* enabled) const override;
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int32_t MicrophoneMuteIsAvailable(bool* available) override;
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int32_t SetMicrophoneMute(bool enable) override;
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int32_t MicrophoneMute(bool* enabled) const override;
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int32_t MicrophoneBoostIsAvailable(bool* available) override;
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int32_t SetMicrophoneBoost(bool enable) override;
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int32_t MicrophoneBoost(bool* enabled) const override;
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int32_t StereoPlayoutIsAvailable(bool* available) const override;
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int32_t SetStereoPlayout(bool enable) override;
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int32_t StereoPlayout(bool* enabled) const override;
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int32_t StereoRecordingIsAvailable(bool* available) const override;
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int32_t SetStereoRecording(bool enable) override;
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int32_t StereoRecording(bool* enabled) const override;
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int32_t SetRecordingChannel(const ChannelType channel) override;
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int32_t RecordingChannel(ChannelType* channel) const override;
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int32_t SetPlayoutBuffer(const BufferType type,
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uint16_t size_ms = 0) override;
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int32_t PlayoutBuffer(BufferType* type, uint16_t* size_ms) const override;
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int32_t PlayoutDelay(uint16_t* delay_ms) const override;
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int32_t RecordingDelay(uint16_t* delay_ms) const override;
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int32_t CPULoad(uint16_t* load) const override;
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int32_t StartRawOutputFileRecording(
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const char pcm_file_name_utf8[webrtc::kAdmMaxFileNameSize]) override;
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int32_t StopRawOutputFileRecording() override;
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int32_t StartRawInputFileRecording(
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const char pcm_file_name_utf8[webrtc::kAdmMaxFileNameSize]) override;
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int32_t StopRawInputFileRecording() override;
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int32_t SetRecordingSampleRate(const uint32_t samples_per_sec) override;
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int32_t RecordingSampleRate(uint32_t* samples_per_sec) const override;
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int32_t SetPlayoutSampleRate(const uint32_t samples_per_sec) override;
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int32_t PlayoutSampleRate(uint32_t* samples_per_sec) const override;
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int32_t ResetAudioDevice() override;
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int32_t SetLoudspeakerStatus(bool enable) override;
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int32_t GetLoudspeakerStatus(bool* enabled) const override;
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bool BuiltInAECIsAvailable() const override { return false; }
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int32_t EnableBuiltInAEC(bool enable) override { return -1; }
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bool BuiltInAGCIsAvailable() const override { return false; }
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int32_t EnableBuiltInAGC(bool enable) override { return -1; }
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bool BuiltInNSIsAvailable() const override { return false; }
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int32_t EnableBuiltInNS(bool enable) override { return -1; }
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// End of functions inherited from webrtc::AudioDeviceModule.
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// The following function is inherited from rtc::MessageHandler.
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void OnMessage(rtc::Message* msg) override;
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protected:
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// The constructor is protected because the class needs to be created as a
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// reference counted object (for memory managment reasons). It could be
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// exposed in which case the burden of proper instantiation would be put on
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// the creator of a FakeAudioCaptureModule instance. To create an instance of
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// this class use the Create(..) API.
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explicit FakeAudioCaptureModule();
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// The destructor is protected because it is reference counted and should not
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// be deleted directly.
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virtual ~FakeAudioCaptureModule();
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private:
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// Initializes the state of the FakeAudioCaptureModule. This API is called on
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// creation by the Create() API.
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bool Initialize();
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// SetBuffer() sets all samples in send_buffer_ to |value|.
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void SetSendBuffer(int value);
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// Resets rec_buffer_. I.e., sets all rec_buffer_ samples to 0.
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void ResetRecBuffer();
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// Returns true if rec_buffer_ contains one or more sample greater than or
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// equal to |value|.
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bool CheckRecBuffer(int value);
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// Returns true/false depending on if recording or playback has been
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// enabled/started.
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bool ShouldStartProcessing();
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// Starts or stops the pushing and pulling of audio frames.
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void UpdateProcessing(bool start);
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// Starts the periodic calling of ProcessFrame() in a thread safe way.
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void StartProcessP();
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// Periodcally called function that ensures that frames are pulled and pushed
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// periodically if enabled/started.
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void ProcessFrameP();
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// Pulls frames from the registered webrtc::AudioTransport.
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void ReceiveFrameP();
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// Pushes frames to the registered webrtc::AudioTransport.
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void SendFrameP();
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// The time in milliseconds when Process() was last called or 0 if no call
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// has been made.
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int64_t last_process_time_ms_;
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// Callback for playout and recording.
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webrtc::AudioTransport* audio_callback_;
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bool recording_; // True when audio is being pushed from the instance.
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bool playing_; // True when audio is being pulled by the instance.
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bool play_is_initialized_; // True when the instance is ready to pull audio.
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bool rec_is_initialized_; // True when the instance is ready to push audio.
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// Input to and output from RecordedDataIsAvailable(..) makes it possible to
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// modify the current mic level. The implementation does not care about the
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// mic level so it just feeds back what it receives.
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uint32_t current_mic_level_;
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// next_frame_time_ is updated in a non-drifting manner to indicate the next
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// wall clock time the next frame should be generated and received. started_
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// ensures that next_frame_time_ can be initialized properly on first call.
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bool started_;
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int64_t next_frame_time_;
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std::unique_ptr<rtc::Thread> process_thread_;
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// Buffer for storing samples received from the webrtc::AudioTransport.
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char rec_buffer_[kNumberSamples * kNumberBytesPerSample];
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// Buffer for samples to send to the webrtc::AudioTransport.
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char send_buffer_[kNumberSamples * kNumberBytesPerSample];
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// Counter of frames received that have samples of high enough amplitude to
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// indicate that the frames are not faked somewhere in the audio pipeline
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// (e.g. by a jitter buffer).
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int frames_received_;
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// Protects variables that are accessed from process_thread_ and
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// the main thread.
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rtc::CriticalSection crit_;
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// Protects |audio_callback_| that is accessed from process_thread_ and
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// the main thread.
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rtc::CriticalSection crit_callback_;
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};
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#endif // WEBRTC_API_TEST_FAKEAUDIOCAPTUREMODULE_H_
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