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