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| 1 /* | |
| 2 * libjingle | |
| 3 * Copyright 2012 Google Inc. | |
| 4 * | |
| 5 * Redistribution and use in source and binary forms, with or without | |
| 6 * modification, are permitted provided that the following conditions are met: | |
| 7 * | |
| 8 * 1. Redistributions of source code must retain the above copyright notice, | |
| 9 * this list of conditions and the following disclaimer. | |
| 10 * 2. Redistributions in binary form must reproduce the above copyright notice, | |
| 11 * this list of conditions and the following disclaimer in the documentation | |
| 12 * and/or other materials provided with the distribution. | |
| 13 * 3. The name of the author may not be used to endorse or promote products | |
| 14 * derived from this software without specific prior written permission. | |
| 15 * | |
| 16 * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED | |
| 17 * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF | |
| 18 * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO | |
| 19 * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, | |
| 20 * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, | |
| 21 * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; | |
| 22 * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, | |
| 23 * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR | |
| 24 * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF | |
| 25 * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. | |
| 26 */ | |
| 27 | |
| 28 // This class implements an AudioCaptureModule that can be used to detect if | |
| 29 // audio is being received properly if it is fed by another AudioCaptureModule | |
| 30 // in some arbitrary audio pipeline where they are connected. It does not play | |
| 31 // out or record any audio so it does not need access to any hardware and can | |
| 32 // therefore be used in the gtest testing framework. | |
| 33 | |
| 34 // Note P postfix of a function indicates that it should only be called by the | |
| 35 // processing thread. | |
| 36 | |
| 37 #ifndef TALK_APP_WEBRTC_TEST_FAKEAUDIOCAPTUREMODULE_H_ | |
| 38 #define TALK_APP_WEBRTC_TEST_FAKEAUDIOCAPTUREMODULE_H_ | |
| 39 | |
| 40 #include "webrtc/base/basictypes.h" | |
| 41 #include "webrtc/base/criticalsection.h" | |
| 42 #include "webrtc/base/messagehandler.h" | |
| 43 #include "webrtc/base/scoped_ptr.h" | |
| 44 #include "webrtc/base/scoped_ref_ptr.h" | |
| 45 #include "webrtc/common_types.h" | |
| 46 #include "webrtc/modules/audio_device/include/audio_device.h" | |
| 47 | |
| 48 namespace rtc { | |
| 49 class Thread; | |
| 50 } // namespace rtc | |
| 51 | |
| 52 class FakeAudioCaptureModule | |
| 53 : public webrtc::AudioDeviceModule, | |
| 54 public rtc::MessageHandler { | |
| 55 public: | |
| 56 typedef uint16_t Sample; | |
| 57 | |
| 58 // The value for the following constants have been derived by running VoE | |
| 59 // using a real ADM. The constants correspond to 10ms of mono audio at 44kHz. | |
| 60 static const size_t kNumberSamples = 440; | |
| 61 static const size_t kNumberBytesPerSample = sizeof(Sample); | |
| 62 | |
| 63 // Creates a FakeAudioCaptureModule or returns NULL on failure. | |
| 64 static rtc::scoped_refptr<FakeAudioCaptureModule> Create(); | |
| 65 | |
| 66 // Returns the number of frames that have been successfully pulled by the | |
| 67 // instance. Note that correctly detecting success can only be done if the | |
| 68 // pulled frame was generated/pushed from a FakeAudioCaptureModule. | |
| 69 int frames_received() const; | |
| 70 | |
| 71 // Following functions are inherited from webrtc::AudioDeviceModule. | |
| 72 // Only functions called by PeerConnection are implemented, the rest do | |
| 73 // nothing and return success. If a function is not expected to be called by | |
| 74 // PeerConnection an assertion is triggered if it is in fact called. | |
| 75 int64_t TimeUntilNextProcess() override; | |
| 76 int32_t Process() override; | |
| 77 | |
| 78 int32_t ActiveAudioLayer(AudioLayer* audio_layer) const override; | |
| 79 | |
| 80 ErrorCode LastError() const override; | |
| 81 int32_t RegisterEventObserver( | |
| 82 webrtc::AudioDeviceObserver* event_callback) override; | |
| 83 | |
| 84 // Note: Calling this method from a callback may result in deadlock. | |
| 85 int32_t RegisterAudioCallback( | |
| 86 webrtc::AudioTransport* audio_callback) override; | |
| 87 | |
| 88 int32_t Init() override; | |
| 89 int32_t Terminate() override; | |
| 90 bool Initialized() const override; | |
| 91 | |
| 92 int16_t PlayoutDevices() override; | |
| 93 int16_t RecordingDevices() override; | |
| 94 int32_t PlayoutDeviceName(uint16_t index, | |
| 95 char name[webrtc::kAdmMaxDeviceNameSize], | |
| 96 char guid[webrtc::kAdmMaxGuidSize]) override; | |
| 97 int32_t RecordingDeviceName(uint16_t index, | |
| 98 char name[webrtc::kAdmMaxDeviceNameSize], | |
| 99 char guid[webrtc::kAdmMaxGuidSize]) override; | |
| 100 | |
| 101 int32_t SetPlayoutDevice(uint16_t index) override; | |
| 102 int32_t SetPlayoutDevice(WindowsDeviceType device) override; | |
| 103 int32_t SetRecordingDevice(uint16_t index) override; | |
| 104 int32_t SetRecordingDevice(WindowsDeviceType device) override; | |
| 105 | |
| 106 int32_t PlayoutIsAvailable(bool* available) override; | |
| 107 int32_t InitPlayout() override; | |
| 108 bool PlayoutIsInitialized() const override; | |
| 109 int32_t RecordingIsAvailable(bool* available) override; | |
| 110 int32_t InitRecording() override; | |
| 111 bool RecordingIsInitialized() const override; | |
| 112 | |
| 113 int32_t StartPlayout() override; | |
| 114 int32_t StopPlayout() override; | |
| 115 bool Playing() const override; | |
| 116 int32_t StartRecording() override; | |
| 117 int32_t StopRecording() override; | |
| 118 bool Recording() const override; | |
| 119 | |
| 120 int32_t SetAGC(bool enable) override; | |
| 121 bool AGC() const override; | |
| 122 | |
| 123 int32_t SetWaveOutVolume(uint16_t volume_left, | |
| 124 uint16_t volume_right) override; | |
| 125 int32_t WaveOutVolume(uint16_t* volume_left, | |
| 126 uint16_t* volume_right) const override; | |
| 127 | |
| 128 int32_t InitSpeaker() override; | |
| 129 bool SpeakerIsInitialized() const override; | |
| 130 int32_t InitMicrophone() override; | |
| 131 bool MicrophoneIsInitialized() const override; | |
| 132 | |
| 133 int32_t SpeakerVolumeIsAvailable(bool* available) override; | |
| 134 int32_t SetSpeakerVolume(uint32_t volume) override; | |
| 135 int32_t SpeakerVolume(uint32_t* volume) const override; | |
| 136 int32_t MaxSpeakerVolume(uint32_t* max_volume) const override; | |
| 137 int32_t MinSpeakerVolume(uint32_t* min_volume) const override; | |
| 138 int32_t SpeakerVolumeStepSize(uint16_t* step_size) const override; | |
| 139 | |
| 140 int32_t MicrophoneVolumeIsAvailable(bool* available) override; | |
| 141 int32_t SetMicrophoneVolume(uint32_t volume) override; | |
| 142 int32_t MicrophoneVolume(uint32_t* volume) const override; | |
| 143 int32_t MaxMicrophoneVolume(uint32_t* max_volume) const override; | |
| 144 | |
| 145 int32_t MinMicrophoneVolume(uint32_t* min_volume) const override; | |
| 146 int32_t MicrophoneVolumeStepSize(uint16_t* step_size) const override; | |
| 147 | |
| 148 int32_t SpeakerMuteIsAvailable(bool* available) override; | |
| 149 int32_t SetSpeakerMute(bool enable) override; | |
| 150 int32_t SpeakerMute(bool* enabled) const override; | |
| 151 | |
| 152 int32_t MicrophoneMuteIsAvailable(bool* available) override; | |
| 153 int32_t SetMicrophoneMute(bool enable) override; | |
| 154 int32_t MicrophoneMute(bool* enabled) const override; | |
| 155 | |
| 156 int32_t MicrophoneBoostIsAvailable(bool* available) override; | |
| 157 int32_t SetMicrophoneBoost(bool enable) override; | |
| 158 int32_t MicrophoneBoost(bool* enabled) const override; | |
| 159 | |
| 160 int32_t StereoPlayoutIsAvailable(bool* available) const override; | |
| 161 int32_t SetStereoPlayout(bool enable) override; | |
| 162 int32_t StereoPlayout(bool* enabled) const override; | |
| 163 int32_t StereoRecordingIsAvailable(bool* available) const override; | |
| 164 int32_t SetStereoRecording(bool enable) override; | |
| 165 int32_t StereoRecording(bool* enabled) const override; | |
| 166 int32_t SetRecordingChannel(const ChannelType channel) override; | |
| 167 int32_t RecordingChannel(ChannelType* channel) const override; | |
| 168 | |
| 169 int32_t SetPlayoutBuffer(const BufferType type, | |
| 170 uint16_t size_ms = 0) override; | |
| 171 int32_t PlayoutBuffer(BufferType* type, uint16_t* size_ms) const override; | |
| 172 int32_t PlayoutDelay(uint16_t* delay_ms) const override; | |
| 173 int32_t RecordingDelay(uint16_t* delay_ms) const override; | |
| 174 | |
| 175 int32_t CPULoad(uint16_t* load) const override; | |
| 176 | |
| 177 int32_t StartRawOutputFileRecording( | |
| 178 const char pcm_file_name_utf8[webrtc::kAdmMaxFileNameSize]) override; | |
| 179 int32_t StopRawOutputFileRecording() override; | |
| 180 int32_t StartRawInputFileRecording( | |
| 181 const char pcm_file_name_utf8[webrtc::kAdmMaxFileNameSize]) override; | |
| 182 int32_t StopRawInputFileRecording() override; | |
| 183 | |
| 184 int32_t SetRecordingSampleRate(const uint32_t samples_per_sec) override; | |
| 185 int32_t RecordingSampleRate(uint32_t* samples_per_sec) const override; | |
| 186 int32_t SetPlayoutSampleRate(const uint32_t samples_per_sec) override; | |
| 187 int32_t PlayoutSampleRate(uint32_t* samples_per_sec) const override; | |
| 188 | |
| 189 int32_t ResetAudioDevice() override; | |
| 190 int32_t SetLoudspeakerStatus(bool enable) override; | |
| 191 int32_t GetLoudspeakerStatus(bool* enabled) const override; | |
| 192 virtual bool BuiltInAECIsAvailable() const { return false; } | |
| 193 virtual int32_t EnableBuiltInAEC(bool enable) { return -1; } | |
| 194 virtual bool BuiltInAGCIsAvailable() const { return false; } | |
| 195 virtual int32_t EnableBuiltInAGC(bool enable) { return -1; } | |
| 196 virtual bool BuiltInNSIsAvailable() const { return false; } | |
| 197 virtual int32_t EnableBuiltInNS(bool enable) { return -1; } | |
| 198 // End of functions inherited from webrtc::AudioDeviceModule. | |
| 199 | |
| 200 // The following function is inherited from rtc::MessageHandler. | |
| 201 void OnMessage(rtc::Message* msg) override; | |
| 202 | |
| 203 protected: | |
| 204 // The constructor is protected because the class needs to be created as a | |
| 205 // reference counted object (for memory managment reasons). It could be | |
| 206 // exposed in which case the burden of proper instantiation would be put on | |
| 207 // the creator of a FakeAudioCaptureModule instance. To create an instance of | |
| 208 // this class use the Create(..) API. | |
| 209 explicit FakeAudioCaptureModule(); | |
| 210 // The destructor is protected because it is reference counted and should not | |
| 211 // be deleted directly. | |
| 212 virtual ~FakeAudioCaptureModule(); | |
| 213 | |
| 214 private: | |
| 215 // Initializes the state of the FakeAudioCaptureModule. This API is called on | |
| 216 // creation by the Create() API. | |
| 217 bool Initialize(); | |
| 218 // SetBuffer() sets all samples in send_buffer_ to |value|. | |
| 219 void SetSendBuffer(int value); | |
| 220 // Resets rec_buffer_. I.e., sets all rec_buffer_ samples to 0. | |
| 221 void ResetRecBuffer(); | |
| 222 // Returns true if rec_buffer_ contains one or more sample greater than or | |
| 223 // equal to |value|. | |
| 224 bool CheckRecBuffer(int value); | |
| 225 | |
| 226 // Returns true/false depending on if recording or playback has been | |
| 227 // enabled/started. | |
| 228 bool ShouldStartProcessing(); | |
| 229 | |
| 230 // Starts or stops the pushing and pulling of audio frames. | |
| 231 void UpdateProcessing(bool start); | |
| 232 | |
| 233 // Starts the periodic calling of ProcessFrame() in a thread safe way. | |
| 234 void StartProcessP(); | |
| 235 // Periodcally called function that ensures that frames are pulled and pushed | |
| 236 // periodically if enabled/started. | |
| 237 void ProcessFrameP(); | |
| 238 // Pulls frames from the registered webrtc::AudioTransport. | |
| 239 void ReceiveFrameP(); | |
| 240 // Pushes frames to the registered webrtc::AudioTransport. | |
| 241 void SendFrameP(); | |
| 242 | |
| 243 // The time in milliseconds when Process() was last called or 0 if no call | |
| 244 // has been made. | |
| 245 uint32_t last_process_time_ms_; | |
| 246 | |
| 247 // Callback for playout and recording. | |
| 248 webrtc::AudioTransport* audio_callback_; | |
| 249 | |
| 250 bool recording_; // True when audio is being pushed from the instance. | |
| 251 bool playing_; // True when audio is being pulled by the instance. | |
| 252 | |
| 253 bool play_is_initialized_; // True when the instance is ready to pull audio. | |
| 254 bool rec_is_initialized_; // True when the instance is ready to push audio. | |
| 255 | |
| 256 // Input to and output from RecordedDataIsAvailable(..) makes it possible to | |
| 257 // modify the current mic level. The implementation does not care about the | |
| 258 // mic level so it just feeds back what it receives. | |
| 259 uint32_t current_mic_level_; | |
| 260 | |
| 261 // next_frame_time_ is updated in a non-drifting manner to indicate the next | |
| 262 // wall clock time the next frame should be generated and received. started_ | |
| 263 // ensures that next_frame_time_ can be initialized properly on first call. | |
| 264 bool started_; | |
| 265 uint32_t next_frame_time_; | |
| 266 | |
| 267 rtc::scoped_ptr<rtc::Thread> process_thread_; | |
| 268 | |
| 269 // Buffer for storing samples received from the webrtc::AudioTransport. | |
| 270 char rec_buffer_[kNumberSamples * kNumberBytesPerSample]; | |
| 271 // Buffer for samples to send to the webrtc::AudioTransport. | |
| 272 char send_buffer_[kNumberSamples * kNumberBytesPerSample]; | |
| 273 | |
| 274 // Counter of frames received that have samples of high enough amplitude to | |
| 275 // indicate that the frames are not faked somewhere in the audio pipeline | |
| 276 // (e.g. by a jitter buffer). | |
| 277 int frames_received_; | |
| 278 | |
| 279 // Protects variables that are accessed from process_thread_ and | |
| 280 // the main thread. | |
| 281 mutable rtc::CriticalSection crit_; | |
| 282 // Protects |audio_callback_| that is accessed from process_thread_ and | |
| 283 // the main thread. | |
| 284 rtc::CriticalSection crit_callback_; | |
| 285 }; | |
| 286 | |
| 287 #endif // TALK_APP_WEBRTC_TEST_FAKEAUDIOCAPTUREMODULE_H_ | |
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Chromium Code Reviews
Issue 1610243002:
Move talk/app/webrtc to webrtc/api (Closed)
Base URL: https://chromium.googlesource.com/external/webrtc.git@master