Create //webrtc/api:libjingle_peerconnection_api + refactorings.

webrtc/api/test/fakeaudiocapturemodule.h

-/*
- *  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.
-
-#ifndef WEBRTC_API_TEST_FAKEAUDIOCAPTUREMODULE_H_
-#define WEBRTC_API_TEST_FAKEAUDIOCAPTUREMODULE_H_
-
-#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 {
- public:
-  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;
-
- protected:
-  // 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();
-
- private:
-  // 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_;
-};
-
-#endif  // WEBRTC_API_TEST_FAKEAUDIOCAPTUREMODULE_H_