henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 1 | /* |
kjellander | b24317b | 2016-02-10 07:54:43 -0800 | [diff] [blame] | 2 | * Copyright 2012 The WebRTC project authors. All Rights Reserved. |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 3 | * |
kjellander | b24317b | 2016-02-10 07:54:43 -0800 | [diff] [blame] | 4 | * Use of this source code is governed by a BSD-style license |
| 5 | * that can be found in the LICENSE file in the root of the source |
| 6 | * tree. An additional intellectual property rights grant can be found |
| 7 | * in the file PATENTS. All contributing project authors may |
| 8 | * be found in the AUTHORS file in the root of the source tree. |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 9 | */ |
| 10 | |
| 11 | // This class implements an AudioCaptureModule that can be used to detect if |
| 12 | // audio is being received properly if it is fed by another AudioCaptureModule |
| 13 | // in some arbitrary audio pipeline where they are connected. It does not play |
| 14 | // out or record any audio so it does not need access to any hardware and can |
| 15 | // therefore be used in the gtest testing framework. |
| 16 | |
| 17 | // Note P postfix of a function indicates that it should only be called by the |
| 18 | // processing thread. |
| 19 | |
Mirko Bonadei | 92ea95e | 2017-09-15 06:47:31 +0200 | [diff] [blame] | 20 | #ifndef PC_TEST_FAKEAUDIOCAPTUREMODULE_H_ |
| 21 | #define PC_TEST_FAKEAUDIOCAPTUREMODULE_H_ |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 22 | |
kwiberg | d1fe281 | 2016-04-27 06:47:29 -0700 | [diff] [blame] | 23 | #include <memory> |
| 24 | |
Mirko Bonadei | 7120742 | 2017-09-15 13:58:09 +0200 | [diff] [blame] | 25 | #include "common_types.h" // NOLINT(build/include) |
Mirko Bonadei | 92ea95e | 2017-09-15 06:47:31 +0200 | [diff] [blame] | 26 | #include "modules/audio_device/include/audio_device.h" |
| 27 | #include "rtc_base/basictypes.h" |
| 28 | #include "rtc_base/criticalsection.h" |
| 29 | #include "rtc_base/messagehandler.h" |
| 30 | #include "rtc_base/scoped_ref_ptr.h" |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 31 | |
buildbot@webrtc.org | d4e598d | 2014-07-29 17:36:52 +0000 | [diff] [blame] | 32 | namespace rtc { |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 33 | class Thread; |
buildbot@webrtc.org | d4e598d | 2014-07-29 17:36:52 +0000 | [diff] [blame] | 34 | } // namespace rtc |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 35 | |
| 36 | class FakeAudioCaptureModule |
| 37 | : public webrtc::AudioDeviceModule, |
buildbot@webrtc.org | d4e598d | 2014-07-29 17:36:52 +0000 | [diff] [blame] | 38 | public rtc::MessageHandler { |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 39 | public: |
Peter Boström | 0c4e06b | 2015-10-07 12:23:21 +0200 | [diff] [blame] | 40 | typedef uint16_t Sample; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 41 | |
| 42 | // The value for the following constants have been derived by running VoE |
| 43 | // using a real ADM. The constants correspond to 10ms of mono audio at 44kHz. |
Peter Kasting | dce40cf | 2015-08-24 14:52:23 -0700 | [diff] [blame] | 44 | static const size_t kNumberSamples = 440; |
| 45 | static const size_t kNumberBytesPerSample = sizeof(Sample); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 46 | |
| 47 | // Creates a FakeAudioCaptureModule or returns NULL on failure. |
deadbeef | ee8c6d3 | 2015-08-13 14:27:18 -0700 | [diff] [blame] | 48 | static rtc::scoped_refptr<FakeAudioCaptureModule> Create(); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 49 | |
| 50 | // Returns the number of frames that have been successfully pulled by the |
| 51 | // instance. Note that correctly detecting success can only be done if the |
| 52 | // pulled frame was generated/pushed from a FakeAudioCaptureModule. |
| 53 | int frames_received() const; |
| 54 | |
kjellander@webrtc.org | 14665ff | 2015-03-04 12:58:35 +0000 | [diff] [blame] | 55 | int32_t ActiveAudioLayer(AudioLayer* audio_layer) const override; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 56 | |
wu@webrtc.org | 8804a29 | 2013-10-22 23:09:20 +0000 | [diff] [blame] | 57 | // Note: Calling this method from a callback may result in deadlock. |
kjellander@webrtc.org | 14665ff | 2015-03-04 12:58:35 +0000 | [diff] [blame] | 58 | int32_t RegisterAudioCallback( |
| 59 | webrtc::AudioTransport* audio_callback) override; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 60 | |
kjellander@webrtc.org | 14665ff | 2015-03-04 12:58:35 +0000 | [diff] [blame] | 61 | int32_t Init() override; |
| 62 | int32_t Terminate() override; |
| 63 | bool Initialized() const override; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 64 | |
kjellander@webrtc.org | 14665ff | 2015-03-04 12:58:35 +0000 | [diff] [blame] | 65 | int16_t PlayoutDevices() override; |
| 66 | int16_t RecordingDevices() override; |
| 67 | int32_t PlayoutDeviceName(uint16_t index, |
| 68 | char name[webrtc::kAdmMaxDeviceNameSize], |
| 69 | char guid[webrtc::kAdmMaxGuidSize]) override; |
| 70 | int32_t RecordingDeviceName(uint16_t index, |
| 71 | char name[webrtc::kAdmMaxDeviceNameSize], |
| 72 | char guid[webrtc::kAdmMaxGuidSize]) override; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 73 | |
kjellander@webrtc.org | 14665ff | 2015-03-04 12:58:35 +0000 | [diff] [blame] | 74 | int32_t SetPlayoutDevice(uint16_t index) override; |
| 75 | int32_t SetPlayoutDevice(WindowsDeviceType device) override; |
| 76 | int32_t SetRecordingDevice(uint16_t index) override; |
| 77 | int32_t SetRecordingDevice(WindowsDeviceType device) override; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 78 | |
kjellander@webrtc.org | 14665ff | 2015-03-04 12:58:35 +0000 | [diff] [blame] | 79 | int32_t PlayoutIsAvailable(bool* available) override; |
| 80 | int32_t InitPlayout() override; |
| 81 | bool PlayoutIsInitialized() const override; |
| 82 | int32_t RecordingIsAvailable(bool* available) override; |
| 83 | int32_t InitRecording() override; |
| 84 | bool RecordingIsInitialized() const override; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 85 | |
kjellander@webrtc.org | 14665ff | 2015-03-04 12:58:35 +0000 | [diff] [blame] | 86 | int32_t StartPlayout() override; |
| 87 | int32_t StopPlayout() override; |
| 88 | bool Playing() const override; |
| 89 | int32_t StartRecording() override; |
| 90 | int32_t StopRecording() override; |
| 91 | bool Recording() const override; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 92 | |
kjellander@webrtc.org | 14665ff | 2015-03-04 12:58:35 +0000 | [diff] [blame] | 93 | int32_t SetAGC(bool enable) override; |
| 94 | bool AGC() const override; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 95 | |
kjellander@webrtc.org | 14665ff | 2015-03-04 12:58:35 +0000 | [diff] [blame] | 96 | int32_t InitSpeaker() override; |
| 97 | bool SpeakerIsInitialized() const override; |
| 98 | int32_t InitMicrophone() override; |
| 99 | bool MicrophoneIsInitialized() const override; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 100 | |
kjellander@webrtc.org | 14665ff | 2015-03-04 12:58:35 +0000 | [diff] [blame] | 101 | int32_t SpeakerVolumeIsAvailable(bool* available) override; |
| 102 | int32_t SetSpeakerVolume(uint32_t volume) override; |
| 103 | int32_t SpeakerVolume(uint32_t* volume) const override; |
| 104 | int32_t MaxSpeakerVolume(uint32_t* max_volume) const override; |
| 105 | int32_t MinSpeakerVolume(uint32_t* min_volume) const override; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 106 | |
kjellander@webrtc.org | 14665ff | 2015-03-04 12:58:35 +0000 | [diff] [blame] | 107 | int32_t MicrophoneVolumeIsAvailable(bool* available) override; |
| 108 | int32_t SetMicrophoneVolume(uint32_t volume) override; |
| 109 | int32_t MicrophoneVolume(uint32_t* volume) const override; |
| 110 | int32_t MaxMicrophoneVolume(uint32_t* max_volume) const override; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 111 | |
kjellander@webrtc.org | 14665ff | 2015-03-04 12:58:35 +0000 | [diff] [blame] | 112 | int32_t MinMicrophoneVolume(uint32_t* min_volume) const override; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 113 | |
kjellander@webrtc.org | 14665ff | 2015-03-04 12:58:35 +0000 | [diff] [blame] | 114 | int32_t SpeakerMuteIsAvailable(bool* available) override; |
| 115 | int32_t SetSpeakerMute(bool enable) override; |
| 116 | int32_t SpeakerMute(bool* enabled) const override; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 117 | |
kjellander@webrtc.org | 14665ff | 2015-03-04 12:58:35 +0000 | [diff] [blame] | 118 | int32_t MicrophoneMuteIsAvailable(bool* available) override; |
| 119 | int32_t SetMicrophoneMute(bool enable) override; |
| 120 | int32_t MicrophoneMute(bool* enabled) const override; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 121 | |
kjellander@webrtc.org | 14665ff | 2015-03-04 12:58:35 +0000 | [diff] [blame] | 122 | int32_t StereoPlayoutIsAvailable(bool* available) const override; |
| 123 | int32_t SetStereoPlayout(bool enable) override; |
| 124 | int32_t StereoPlayout(bool* enabled) const override; |
| 125 | int32_t StereoRecordingIsAvailable(bool* available) const override; |
| 126 | int32_t SetStereoRecording(bool enable) override; |
| 127 | int32_t StereoRecording(bool* enabled) const override; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 128 | |
kjellander@webrtc.org | 14665ff | 2015-03-04 12:58:35 +0000 | [diff] [blame] | 129 | int32_t PlayoutDelay(uint16_t* delay_ms) const override; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 130 | |
kjellander@webrtc.org | 14665ff | 2015-03-04 12:58:35 +0000 | [diff] [blame] | 131 | int32_t SetRecordingSampleRate(const uint32_t samples_per_sec) override; |
| 132 | int32_t RecordingSampleRate(uint32_t* samples_per_sec) const override; |
| 133 | int32_t SetPlayoutSampleRate(const uint32_t samples_per_sec) override; |
| 134 | int32_t PlayoutSampleRate(uint32_t* samples_per_sec) const override; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 135 | |
kjellander@webrtc.org | 14665ff | 2015-03-04 12:58:35 +0000 | [diff] [blame] | 136 | int32_t SetLoudspeakerStatus(bool enable) override; |
| 137 | int32_t GetLoudspeakerStatus(bool* enabled) const override; |
nisse | ef8b61e | 2016-04-29 06:09:15 -0700 | [diff] [blame] | 138 | bool BuiltInAECIsAvailable() const override { return false; } |
| 139 | int32_t EnableBuiltInAEC(bool enable) override { return -1; } |
| 140 | bool BuiltInAGCIsAvailable() const override { return false; } |
| 141 | int32_t EnableBuiltInAGC(bool enable) override { return -1; } |
| 142 | bool BuiltInNSIsAvailable() const override { return false; } |
| 143 | int32_t EnableBuiltInNS(bool enable) override { return -1; } |
maxmorin | 88e31a3 | 2016-08-16 00:56:09 -0700 | [diff] [blame] | 144 | #if defined(WEBRTC_IOS) |
| 145 | int GetPlayoutAudioParameters( |
| 146 | webrtc::AudioParameters* params) const override { |
| 147 | return -1; |
| 148 | } |
| 149 | int GetRecordAudioParameters(webrtc::AudioParameters* params) const override { |
| 150 | return -1; |
| 151 | } |
| 152 | #endif // WEBRTC_IOS |
| 153 | |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 154 | // End of functions inherited from webrtc::AudioDeviceModule. |
| 155 | |
buildbot@webrtc.org | d4e598d | 2014-07-29 17:36:52 +0000 | [diff] [blame] | 156 | // The following function is inherited from rtc::MessageHandler. |
kjellander@webrtc.org | 14665ff | 2015-03-04 12:58:35 +0000 | [diff] [blame] | 157 | void OnMessage(rtc::Message* msg) override; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 158 | |
| 159 | protected: |
| 160 | // The constructor is protected because the class needs to be created as a |
| 161 | // reference counted object (for memory managment reasons). It could be |
| 162 | // exposed in which case the burden of proper instantiation would be put on |
| 163 | // the creator of a FakeAudioCaptureModule instance. To create an instance of |
| 164 | // this class use the Create(..) API. |
Steve Anton | 36b29d1 | 2017-10-30 09:57:42 -0700 | [diff] [blame] | 165 | FakeAudioCaptureModule(); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 166 | // The destructor is protected because it is reference counted and should not |
| 167 | // be deleted directly. |
| 168 | virtual ~FakeAudioCaptureModule(); |
| 169 | |
| 170 | private: |
| 171 | // Initializes the state of the FakeAudioCaptureModule. This API is called on |
| 172 | // creation by the Create() API. |
| 173 | bool Initialize(); |
| 174 | // SetBuffer() sets all samples in send_buffer_ to |value|. |
| 175 | void SetSendBuffer(int value); |
| 176 | // Resets rec_buffer_. I.e., sets all rec_buffer_ samples to 0. |
| 177 | void ResetRecBuffer(); |
| 178 | // Returns true if rec_buffer_ contains one or more sample greater than or |
| 179 | // equal to |value|. |
| 180 | bool CheckRecBuffer(int value); |
| 181 | |
wu@webrtc.org | 8804a29 | 2013-10-22 23:09:20 +0000 | [diff] [blame] | 182 | // Returns true/false depending on if recording or playback has been |
| 183 | // enabled/started. |
| 184 | bool ShouldStartProcessing(); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 185 | |
wu@webrtc.org | 8804a29 | 2013-10-22 23:09:20 +0000 | [diff] [blame] | 186 | // Starts or stops the pushing and pulling of audio frames. |
| 187 | void UpdateProcessing(bool start); |
| 188 | |
| 189 | // Starts the periodic calling of ProcessFrame() in a thread safe way. |
| 190 | void StartProcessP(); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 191 | // Periodcally called function that ensures that frames are pulled and pushed |
| 192 | // periodically if enabled/started. |
| 193 | void ProcessFrameP(); |
| 194 | // Pulls frames from the registered webrtc::AudioTransport. |
| 195 | void ReceiveFrameP(); |
| 196 | // Pushes frames to the registered webrtc::AudioTransport. |
| 197 | void SendFrameP(); |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 198 | |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 199 | // Callback for playout and recording. |
| 200 | webrtc::AudioTransport* audio_callback_; |
| 201 | |
Steve Anton | 36b29d1 | 2017-10-30 09:57:42 -0700 | [diff] [blame] | 202 | bool recording_; // True when audio is being pushed from the instance. |
| 203 | bool playing_; // True when audio is being pulled by the instance. |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 204 | |
Steve Anton | 36b29d1 | 2017-10-30 09:57:42 -0700 | [diff] [blame] | 205 | bool play_is_initialized_; // True when the instance is ready to pull audio. |
| 206 | bool rec_is_initialized_; // True when the instance is ready to push audio. |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 207 | |
| 208 | // Input to and output from RecordedDataIsAvailable(..) makes it possible to |
| 209 | // modify the current mic level. The implementation does not care about the |
| 210 | // mic level so it just feeds back what it receives. |
| 211 | uint32_t current_mic_level_; |
| 212 | |
| 213 | // next_frame_time_ is updated in a non-drifting manner to indicate the next |
| 214 | // wall clock time the next frame should be generated and received. started_ |
| 215 | // ensures that next_frame_time_ can be initialized properly on first call. |
| 216 | bool started_; |
Honghai Zhang | 82d7862 | 2016-05-06 11:29:15 -0700 | [diff] [blame] | 217 | int64_t next_frame_time_; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 218 | |
kwiberg | d1fe281 | 2016-04-27 06:47:29 -0700 | [diff] [blame] | 219 | std::unique_ptr<rtc::Thread> process_thread_; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 220 | |
| 221 | // Buffer for storing samples received from the webrtc::AudioTransport. |
| 222 | char rec_buffer_[kNumberSamples * kNumberBytesPerSample]; |
| 223 | // Buffer for samples to send to the webrtc::AudioTransport. |
| 224 | char send_buffer_[kNumberSamples * kNumberBytesPerSample]; |
| 225 | |
| 226 | // Counter of frames received that have samples of high enough amplitude to |
| 227 | // indicate that the frames are not faked somewhere in the audio pipeline |
| 228 | // (e.g. by a jitter buffer). |
| 229 | int frames_received_; |
wu@webrtc.org | 8804a29 | 2013-10-22 23:09:20 +0000 | [diff] [blame] | 230 | |
| 231 | // Protects variables that are accessed from process_thread_ and |
| 232 | // the main thread. |
pbos | 5ad935c | 2016-01-25 03:52:44 -0800 | [diff] [blame] | 233 | rtc::CriticalSection crit_; |
wu@webrtc.org | 8804a29 | 2013-10-22 23:09:20 +0000 | [diff] [blame] | 234 | // Protects |audio_callback_| that is accessed from process_thread_ and |
| 235 | // the main thread. |
buildbot@webrtc.org | d4e598d | 2014-07-29 17:36:52 +0000 | [diff] [blame] | 236 | rtc::CriticalSection crit_callback_; |
henrike@webrtc.org | 28e2075 | 2013-07-10 00:45:36 +0000 | [diff] [blame] | 237 | }; |
| 238 | |
Mirko Bonadei | 92ea95e | 2017-09-15 06:47:31 +0200 | [diff] [blame] | 239 | #endif // PC_TEST_FAKEAUDIOCAPTUREMODULE_H_ |