1 /* 2 * Copyright (c) 2018 The WebRTC project authors. All Rights Reserved. 3 * 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. 9 */ 10 11 #ifndef SDK_ANDROID_SRC_JNI_AUDIO_DEVICE_AAUDIO_PLAYER_H_ 12 #define SDK_ANDROID_SRC_JNI_AUDIO_DEVICE_AAUDIO_PLAYER_H_ 13 14 #include <aaudio/AAudio.h> 15 16 #include <memory> 17 18 #include "absl/types/optional.h" 19 #include "api/sequence_checker.h" 20 #include "api/task_queue/task_queue_base.h" 21 #include "modules/audio_device/audio_device_buffer.h" 22 #include "modules/audio_device/include/audio_device_defines.h" 23 #include "rtc_base/thread_annotations.h" 24 #include "sdk/android/src/jni/audio_device/aaudio_wrapper.h" 25 #include "sdk/android/src/jni/audio_device/audio_device_module.h" 26 27 namespace webrtc { 28 29 class AudioDeviceBuffer; 30 class FineAudioBuffer; 31 32 namespace jni { 33 34 // Implements low-latency 16-bit mono PCM audio output support for Android 35 // using the C based AAudio API. 36 // 37 // An instance must be created and destroyed on one and the same thread. 38 // All public methods must also be called on the same thread. A thread checker 39 // will DCHECK if any method is called on an invalid thread. Audio buffers 40 // are requested on a dedicated high-priority thread owned by AAudio. 41 // 42 // The existing design forces the user to call InitPlayout() after StopPlayout() 43 // to be able to call StartPlayout() again. This is in line with how the Java- 44 // based implementation works. 45 // 46 // An audio stream can be disconnected, e.g. when an audio device is removed. 47 // This implementation will restart the audio stream using the new preferred 48 // device if such an event happens. 49 // 50 // Also supports automatic buffer-size adjustment based on underrun detections 51 // where the internal AAudio buffer can be increased when needed. It will 52 // reduce the risk of underruns (~glitches) at the expense of an increased 53 // latency. 54 class AAudioPlayer final : public AudioOutput, public AAudioObserverInterface { 55 public: 56 explicit AAudioPlayer(const AudioParameters& audio_parameters); 57 ~AAudioPlayer() override; 58 59 int Init() override; 60 int Terminate() override; 61 62 int InitPlayout() override; 63 bool PlayoutIsInitialized() const override; 64 65 int StartPlayout() override; 66 int StopPlayout() override; 67 bool Playing() const override; 68 69 void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) override; 70 71 // Not implemented in AAudio. 72 bool SpeakerVolumeIsAvailable() override; 73 int SetSpeakerVolume(uint32_t volume) override; 74 absl::optional<uint32_t> SpeakerVolume() const override; 75 absl::optional<uint32_t> MaxSpeakerVolume() const override; 76 absl::optional<uint32_t> MinSpeakerVolume() const override; 77 78 protected: 79 // AAudioObserverInterface implementation. 80 81 // For an output stream, this function should render and write `num_frames` 82 // of data in the streams current data format to the `audio_data` buffer. 83 // Called on a real-time thread owned by AAudio. 84 aaudio_data_callback_result_t OnDataCallback(void* audio_data, 85 int32_t num_frames) override; 86 // AAudio calls this functions if any error occurs on a callback thread. 87 // Called on a real-time thread owned by AAudio. 88 void OnErrorCallback(aaudio_result_t error) override; 89 90 private: 91 // TODO(henrika): Implement. GetPlayoutUnderrunCount()92 int GetPlayoutUnderrunCount() override { return 0; } 93 94 // Closes the existing stream and starts a new stream. 95 void HandleStreamDisconnected(); 96 97 // Ensures that methods are called from the same thread as this object is 98 // created on. 99 SequenceChecker main_thread_checker_; 100 101 // Stores thread ID in first call to AAudioPlayer::OnDataCallback from a 102 // real-time thread owned by AAudio. Detached during construction of this 103 // object. 104 SequenceChecker thread_checker_aaudio_; 105 106 // The thread on which this object is created on. 107 TaskQueueBase* main_thread_; 108 109 // Wraps all AAudio resources. Contains an output stream using the default 110 // output audio device. Can be accessed on both the main thread and the 111 // real-time thread owned by AAudio. See separate AAudio documentation about 112 // thread safety. 113 AAudioWrapper aaudio_; 114 115 // FineAudioBuffer takes an AudioDeviceBuffer which delivers audio data 116 // in chunks of 10ms. It then allows for this data to be pulled in 117 // a finer or coarser granularity. I.e. interacting with this class instead 118 // of directly with the AudioDeviceBuffer one can ask for any number of 119 // audio data samples. 120 // Example: native buffer size can be 192 audio frames at 48kHz sample rate. 121 // WebRTC will provide 480 audio frames per 10ms but AAudio asks for 192 122 // in each callback (once every 4th ms). This class can then ask for 192 and 123 // the FineAudioBuffer will ask WebRTC for new data approximately only every 124 // second callback and also cache non-utilized audio. 125 std::unique_ptr<FineAudioBuffer> fine_audio_buffer_; 126 127 // Counts number of detected underrun events reported by AAudio. 128 int32_t underrun_count_ = 0; 129 130 // True only for the first data callback in each audio session. 131 bool first_data_callback_ = true; 132 133 // Raw pointer handle provided to us in AttachAudioBuffer(). Owned by the 134 // AudioDeviceModuleImpl class and set by AudioDeviceModule::Create(). 135 AudioDeviceBuffer* audio_device_buffer_ RTC_GUARDED_BY(main_thread_checker_) = 136 nullptr; 137 138 bool initialized_ RTC_GUARDED_BY(main_thread_checker_) = false; 139 bool playing_ RTC_GUARDED_BY(main_thread_checker_) = false; 140 141 // Estimated latency between writing an audio frame to the output stream and 142 // the time that same frame is played out on the output audio device. 143 double latency_millis_ RTC_GUARDED_BY(thread_checker_aaudio_) = 0; 144 }; 145 146 } // namespace jni 147 148 } // namespace webrtc 149 150 #endif // SDK_ANDROID_SRC_JNI_AUDIO_DEVICE_AAUDIO_PLAYER_H_ 151