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henrike@webrtc.org28e20752013-07-10 00:45:36 +00001/*
kjellanderb24317b2016-02-10 07:54:43 -08002 * Copyright 2012 The WebRTC project authors. All Rights Reserved.
henrike@webrtc.org28e20752013-07-10 00:45:36 +00003 *
kjellanderb24317b2016-02-10 07:54:43 -08004 * 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.org28e20752013-07-10 00:45:36 +00009 */
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
Henrik Kjellander15583c12016-02-10 10:53:12 +010020#ifndef WEBRTC_API_TEST_FAKEAUDIOCAPTUREMODULE_H_
21#define WEBRTC_API_TEST_FAKEAUDIOCAPTUREMODULE_H_
henrike@webrtc.org28e20752013-07-10 00:45:36 +000022
kwibergd1fe2812016-04-27 06:47:29 -070023#include <memory>
24
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +000025#include "webrtc/base/basictypes.h"
26#include "webrtc/base/criticalsection.h"
27#include "webrtc/base/messagehandler.h"
28#include "webrtc/base/scoped_ref_ptr.h"
henrike@webrtc.org28e20752013-07-10 00:45:36 +000029#include "webrtc/common_types.h"
30#include "webrtc/modules/audio_device/include/audio_device.h"
31
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +000032namespace rtc {
henrike@webrtc.org28e20752013-07-10 00:45:36 +000033class Thread;
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +000034} // namespace rtc
henrike@webrtc.org28e20752013-07-10 00:45:36 +000035
36class FakeAudioCaptureModule
37 : public webrtc::AudioDeviceModule,
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +000038 public rtc::MessageHandler {
henrike@webrtc.org28e20752013-07-10 00:45:36 +000039 public:
Peter Boström0c4e06b2015-10-07 12:23:21 +020040 typedef uint16_t Sample;
henrike@webrtc.org28e20752013-07-10 00:45:36 +000041
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 Kastingdce40cf2015-08-24 14:52:23 -070044 static const size_t kNumberSamples = 440;
45 static const size_t kNumberBytesPerSample = sizeof(Sample);
henrike@webrtc.org28e20752013-07-10 00:45:36 +000046
47 // Creates a FakeAudioCaptureModule or returns NULL on failure.
deadbeefee8c6d32015-08-13 14:27:18 -070048 static rtc::scoped_refptr<FakeAudioCaptureModule> Create();
henrike@webrtc.org28e20752013-07-10 00:45:36 +000049
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
55 // Following functions are inherited from webrtc::AudioDeviceModule.
56 // Only functions called by PeerConnection are implemented, the rest do
57 // nothing and return success. If a function is not expected to be called by
58 // PeerConnection an assertion is triggered if it is in fact called.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +000059 int64_t TimeUntilNextProcess() override;
pbosa26ac922016-02-25 04:50:01 -080060 void Process() override;
henrike@webrtc.org28e20752013-07-10 00:45:36 +000061
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +000062 int32_t ActiveAudioLayer(AudioLayer* audio_layer) const override;
henrike@webrtc.org28e20752013-07-10 00:45:36 +000063
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +000064 ErrorCode LastError() const override;
65 int32_t RegisterEventObserver(
66 webrtc::AudioDeviceObserver* event_callback) override;
henrike@webrtc.org28e20752013-07-10 00:45:36 +000067
wu@webrtc.org8804a292013-10-22 23:09:20 +000068 // Note: Calling this method from a callback may result in deadlock.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +000069 int32_t RegisterAudioCallback(
70 webrtc::AudioTransport* audio_callback) override;
henrike@webrtc.org28e20752013-07-10 00:45:36 +000071
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +000072 int32_t Init() override;
73 int32_t Terminate() override;
74 bool Initialized() const override;
henrike@webrtc.org28e20752013-07-10 00:45:36 +000075
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +000076 int16_t PlayoutDevices() override;
77 int16_t RecordingDevices() override;
78 int32_t PlayoutDeviceName(uint16_t index,
79 char name[webrtc::kAdmMaxDeviceNameSize],
80 char guid[webrtc::kAdmMaxGuidSize]) override;
81 int32_t RecordingDeviceName(uint16_t index,
82 char name[webrtc::kAdmMaxDeviceNameSize],
83 char guid[webrtc::kAdmMaxGuidSize]) override;
henrike@webrtc.org28e20752013-07-10 00:45:36 +000084
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +000085 int32_t SetPlayoutDevice(uint16_t index) override;
86 int32_t SetPlayoutDevice(WindowsDeviceType device) override;
87 int32_t SetRecordingDevice(uint16_t index) override;
88 int32_t SetRecordingDevice(WindowsDeviceType device) override;
henrike@webrtc.org28e20752013-07-10 00:45:36 +000089
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +000090 int32_t PlayoutIsAvailable(bool* available) override;
91 int32_t InitPlayout() override;
92 bool PlayoutIsInitialized() const override;
93 int32_t RecordingIsAvailable(bool* available) override;
94 int32_t InitRecording() override;
95 bool RecordingIsInitialized() const override;
henrike@webrtc.org28e20752013-07-10 00:45:36 +000096
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +000097 int32_t StartPlayout() override;
98 int32_t StopPlayout() override;
99 bool Playing() const override;
100 int32_t StartRecording() override;
101 int32_t StopRecording() override;
102 bool Recording() const override;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000103
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000104 int32_t SetAGC(bool enable) override;
105 bool AGC() const override;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000106
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000107 int32_t SetWaveOutVolume(uint16_t volume_left,
108 uint16_t volume_right) override;
109 int32_t WaveOutVolume(uint16_t* volume_left,
110 uint16_t* volume_right) const override;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000111
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000112 int32_t InitSpeaker() override;
113 bool SpeakerIsInitialized() const override;
114 int32_t InitMicrophone() override;
115 bool MicrophoneIsInitialized() const override;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000116
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000117 int32_t SpeakerVolumeIsAvailable(bool* available) override;
118 int32_t SetSpeakerVolume(uint32_t volume) override;
119 int32_t SpeakerVolume(uint32_t* volume) const override;
120 int32_t MaxSpeakerVolume(uint32_t* max_volume) const override;
121 int32_t MinSpeakerVolume(uint32_t* min_volume) const override;
122 int32_t SpeakerVolumeStepSize(uint16_t* step_size) const override;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000123
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000124 int32_t MicrophoneVolumeIsAvailable(bool* available) override;
125 int32_t SetMicrophoneVolume(uint32_t volume) override;
126 int32_t MicrophoneVolume(uint32_t* volume) const override;
127 int32_t MaxMicrophoneVolume(uint32_t* max_volume) const override;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000128
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000129 int32_t MinMicrophoneVolume(uint32_t* min_volume) const override;
130 int32_t MicrophoneVolumeStepSize(uint16_t* step_size) const override;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000131
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000132 int32_t SpeakerMuteIsAvailable(bool* available) override;
133 int32_t SetSpeakerMute(bool enable) override;
134 int32_t SpeakerMute(bool* enabled) const override;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000135
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000136 int32_t MicrophoneMuteIsAvailable(bool* available) override;
137 int32_t SetMicrophoneMute(bool enable) override;
138 int32_t MicrophoneMute(bool* enabled) const override;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000139
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000140 int32_t MicrophoneBoostIsAvailable(bool* available) override;
141 int32_t SetMicrophoneBoost(bool enable) override;
142 int32_t MicrophoneBoost(bool* enabled) const override;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000143
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000144 int32_t StereoPlayoutIsAvailable(bool* available) const override;
145 int32_t SetStereoPlayout(bool enable) override;
146 int32_t StereoPlayout(bool* enabled) const override;
147 int32_t StereoRecordingIsAvailable(bool* available) const override;
148 int32_t SetStereoRecording(bool enable) override;
149 int32_t StereoRecording(bool* enabled) const override;
150 int32_t SetRecordingChannel(const ChannelType channel) override;
151 int32_t RecordingChannel(ChannelType* channel) const override;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000152
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000153 int32_t SetPlayoutBuffer(const BufferType type,
154 uint16_t size_ms = 0) override;
155 int32_t PlayoutBuffer(BufferType* type, uint16_t* size_ms) const override;
156 int32_t PlayoutDelay(uint16_t* delay_ms) const override;
157 int32_t RecordingDelay(uint16_t* delay_ms) const override;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000158
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000159 int32_t CPULoad(uint16_t* load) const override;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000160
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000161 int32_t StartRawOutputFileRecording(
162 const char pcm_file_name_utf8[webrtc::kAdmMaxFileNameSize]) override;
163 int32_t StopRawOutputFileRecording() override;
164 int32_t StartRawInputFileRecording(
165 const char pcm_file_name_utf8[webrtc::kAdmMaxFileNameSize]) override;
166 int32_t StopRawInputFileRecording() override;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000167
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000168 int32_t SetRecordingSampleRate(const uint32_t samples_per_sec) override;
169 int32_t RecordingSampleRate(uint32_t* samples_per_sec) const override;
170 int32_t SetPlayoutSampleRate(const uint32_t samples_per_sec) override;
171 int32_t PlayoutSampleRate(uint32_t* samples_per_sec) const override;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000172
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000173 int32_t ResetAudioDevice() override;
174 int32_t SetLoudspeakerStatus(bool enable) override;
175 int32_t GetLoudspeakerStatus(bool* enabled) const override;
nisseef8b61e2016-04-29 06:09:15 -0700176 bool BuiltInAECIsAvailable() const override { return false; }
177 int32_t EnableBuiltInAEC(bool enable) override { return -1; }
178 bool BuiltInAGCIsAvailable() const override { return false; }
179 int32_t EnableBuiltInAGC(bool enable) override { return -1; }
180 bool BuiltInNSIsAvailable() const override { return false; }
181 int32_t EnableBuiltInNS(bool enable) override { return -1; }
maxmorin88e31a32016-08-16 00:56:09 -0700182#if defined(WEBRTC_IOS)
183 int GetPlayoutAudioParameters(
184 webrtc::AudioParameters* params) const override {
185 return -1;
186 }
187 int GetRecordAudioParameters(webrtc::AudioParameters* params) const override {
188 return -1;
189 }
190#endif // WEBRTC_IOS
191
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000192 // End of functions inherited from webrtc::AudioDeviceModule.
193
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000194 // The following function is inherited from rtc::MessageHandler.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000195 void OnMessage(rtc::Message* msg) override;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000196
197 protected:
198 // The constructor is protected because the class needs to be created as a
199 // reference counted object (for memory managment reasons). It could be
200 // exposed in which case the burden of proper instantiation would be put on
201 // the creator of a FakeAudioCaptureModule instance. To create an instance of
202 // this class use the Create(..) API.
deadbeefee8c6d32015-08-13 14:27:18 -0700203 explicit FakeAudioCaptureModule();
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000204 // The destructor is protected because it is reference counted and should not
205 // be deleted directly.
206 virtual ~FakeAudioCaptureModule();
207
208 private:
209 // Initializes the state of the FakeAudioCaptureModule. This API is called on
210 // creation by the Create() API.
211 bool Initialize();
212 // SetBuffer() sets all samples in send_buffer_ to |value|.
213 void SetSendBuffer(int value);
214 // Resets rec_buffer_. I.e., sets all rec_buffer_ samples to 0.
215 void ResetRecBuffer();
216 // Returns true if rec_buffer_ contains one or more sample greater than or
217 // equal to |value|.
218 bool CheckRecBuffer(int value);
219
wu@webrtc.org8804a292013-10-22 23:09:20 +0000220 // Returns true/false depending on if recording or playback has been
221 // enabled/started.
222 bool ShouldStartProcessing();
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000223
wu@webrtc.org8804a292013-10-22 23:09:20 +0000224 // Starts or stops the pushing and pulling of audio frames.
225 void UpdateProcessing(bool start);
226
227 // Starts the periodic calling of ProcessFrame() in a thread safe way.
228 void StartProcessP();
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000229 // Periodcally called function that ensures that frames are pulled and pushed
230 // periodically if enabled/started.
231 void ProcessFrameP();
232 // Pulls frames from the registered webrtc::AudioTransport.
233 void ReceiveFrameP();
234 // Pushes frames to the registered webrtc::AudioTransport.
235 void SendFrameP();
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000236
237 // The time in milliseconds when Process() was last called or 0 if no call
238 // has been made.
Honghai Zhang82d78622016-05-06 11:29:15 -0700239 int64_t last_process_time_ms_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000240
241 // Callback for playout and recording.
242 webrtc::AudioTransport* audio_callback_;
243
244 bool recording_; // True when audio is being pushed from the instance.
245 bool playing_; // True when audio is being pulled by the instance.
246
247 bool play_is_initialized_; // True when the instance is ready to pull audio.
248 bool rec_is_initialized_; // True when the instance is ready to push audio.
249
250 // Input to and output from RecordedDataIsAvailable(..) makes it possible to
251 // modify the current mic level. The implementation does not care about the
252 // mic level so it just feeds back what it receives.
253 uint32_t current_mic_level_;
254
255 // next_frame_time_ is updated in a non-drifting manner to indicate the next
256 // wall clock time the next frame should be generated and received. started_
257 // ensures that next_frame_time_ can be initialized properly on first call.
258 bool started_;
Honghai Zhang82d78622016-05-06 11:29:15 -0700259 int64_t next_frame_time_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000260
kwibergd1fe2812016-04-27 06:47:29 -0700261 std::unique_ptr<rtc::Thread> process_thread_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000262
263 // Buffer for storing samples received from the webrtc::AudioTransport.
264 char rec_buffer_[kNumberSamples * kNumberBytesPerSample];
265 // Buffer for samples to send to the webrtc::AudioTransport.
266 char send_buffer_[kNumberSamples * kNumberBytesPerSample];
267
268 // Counter of frames received that have samples of high enough amplitude to
269 // indicate that the frames are not faked somewhere in the audio pipeline
270 // (e.g. by a jitter buffer).
271 int frames_received_;
wu@webrtc.org8804a292013-10-22 23:09:20 +0000272
273 // Protects variables that are accessed from process_thread_ and
274 // the main thread.
pbos5ad935c2016-01-25 03:52:44 -0800275 rtc::CriticalSection crit_;
wu@webrtc.org8804a292013-10-22 23:09:20 +0000276 // Protects |audio_callback_| that is accessed from process_thread_ and
277 // the main thread.
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000278 rtc::CriticalSection crit_callback_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000279};
280
Henrik Kjellander15583c12016-02-10 10:53:12 +0100281#endif // WEBRTC_API_TEST_FAKEAUDIOCAPTUREMODULE_H_