henrika | 883d00f | 2018-03-16 10:09:49 +0100 | [diff] [blame^] | 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 MODULES_AUDIO_DEVICE_ANDROID_AAUDIO_PLAYER_H_ |
| 12 | #define MODULES_AUDIO_DEVICE_ANDROID_AAUDIO_PLAYER_H_ |
| 13 | |
| 14 | #include <aaudio/AAudio.h> |
| 15 | #include <memory> |
| 16 | |
| 17 | #include "modules/audio_device/android/aaudio_wrapper.h" |
| 18 | #include "modules/audio_device/include/audio_device_defines.h" |
| 19 | #include "rtc_base/messagehandler.h" |
| 20 | #include "rtc_base/thread.h" |
| 21 | #include "rtc_base/thread_annotations.h" |
| 22 | #include "rtc_base/thread_checker.h" |
| 23 | |
| 24 | namespace webrtc { |
| 25 | |
| 26 | class AudioDeviceBuffer; |
| 27 | class FineAudioBuffer; |
| 28 | class AudioManager; |
| 29 | |
| 30 | // Implements low-latency 16-bit mono PCM audio output support for Android |
| 31 | // using the C based AAudio API. |
| 32 | // |
| 33 | // An instance must be created and destroyed on one and the same thread. |
| 34 | // All public methods must also be called on the same thread. A thread checker |
| 35 | // will DCHECK if any method is called on an invalid thread. Audio buffers |
| 36 | // are requested on a dedicated high-priority thread owned by AAudio. |
| 37 | // |
| 38 | // The existing design forces the user to call InitPlayout() after StopPlayout() |
| 39 | // to be able to call StartPlayout() again. This is in line with how the Java- |
| 40 | // based implementation works. |
| 41 | // |
| 42 | // An audio stream can be disconnected, e.g. when an audio device is removed. |
| 43 | // This implementation will restart the audio stream using the new preferred |
| 44 | // device if such an event happens. |
| 45 | // |
| 46 | // Also supports automatic buffer-size adjustment based on underrun detections |
| 47 | // where the internal AAudio buffer can be increased when needed. It will |
| 48 | // reduce the risk of underruns (~glitches) at the expense of an increased |
| 49 | // latency. |
| 50 | class AAudioPlayer final : public AAudioObserverInterface, |
| 51 | public rtc::MessageHandler { |
| 52 | public: |
| 53 | explicit AAudioPlayer(AudioManager* audio_manager); |
| 54 | ~AAudioPlayer(); |
| 55 | |
| 56 | int Init(); |
| 57 | int Terminate(); |
| 58 | |
| 59 | int InitPlayout(); |
| 60 | bool PlayoutIsInitialized() const; |
| 61 | |
| 62 | int StartPlayout(); |
| 63 | int StopPlayout(); |
| 64 | bool Playing() const; |
| 65 | |
| 66 | void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer); |
| 67 | |
| 68 | // Not implemented in AAudio. |
| 69 | int SpeakerVolumeIsAvailable(bool& available); // NOLINT |
| 70 | int SetSpeakerVolume(uint32_t volume) { return -1; } |
| 71 | int SpeakerVolume(uint32_t& volume) const { return -1; } // NOLINT |
| 72 | int MaxSpeakerVolume(uint32_t& maxVolume) const { return -1; } // NOLINT |
| 73 | int MinSpeakerVolume(uint32_t& minVolume) const { return -1; } // NOLINT |
| 74 | |
| 75 | protected: |
| 76 | // AAudioObserverInterface implementation. |
| 77 | |
| 78 | // For an output stream, this function should render and write |num_frames| |
| 79 | // of data in the streams current data format to the |audio_data| buffer. |
| 80 | // Called on a real-time thread owned by AAudio. |
| 81 | aaudio_data_callback_result_t OnDataCallback(void* audio_data, |
| 82 | int32_t num_frames) override; |
| 83 | // AAudio calls this functions if any error occurs on a callback thread. |
| 84 | // Called on a real-time thread owned by AAudio. |
| 85 | void OnErrorCallback(aaudio_result_t error) override; |
| 86 | |
| 87 | // rtc::MessageHandler used for restart messages from the error-callback |
| 88 | // thread to the main (creating) thread. |
| 89 | void OnMessage(rtc::Message* msg) override; |
| 90 | |
| 91 | private: |
| 92 | // Closes the existing stream and starts a new stream. |
| 93 | void HandleStreamDisconnected(); |
| 94 | |
| 95 | // Ensures that methods are called from the same thread as this object is |
| 96 | // created on. |
| 97 | rtc::ThreadChecker main_thread_checker_; |
| 98 | |
| 99 | // Stores thread ID in first call to AAudioPlayer::OnDataCallback from a |
| 100 | // real-time thread owned by AAudio. Detached during construction of this |
| 101 | // object. |
| 102 | rtc::ThreadChecker thread_checker_aaudio_; |
| 103 | |
| 104 | // The thread on which this object is created on. |
| 105 | rtc::Thread* main_thread_; |
| 106 | |
| 107 | // Wraps all AAudio resources. Contains an output stream using the default |
| 108 | // output audio device. Can be accessed on both the main thread and the |
| 109 | // real-time thread owned by AAudio. See separate AAudio documentation about |
| 110 | // thread safety. |
| 111 | AAudioWrapper aaudio_; |
| 112 | |
| 113 | // FineAudioBuffer takes an AudioDeviceBuffer which delivers audio data |
| 114 | // in chunks of 10ms. It then allows for this data to be pulled in |
| 115 | // a finer or coarser granularity. I.e. interacting with this class instead |
| 116 | // of directly with the AudioDeviceBuffer one can ask for any number of |
| 117 | // audio data samples. |
| 118 | // Example: native buffer size can be 192 audio frames at 48kHz sample rate. |
| 119 | // WebRTC will provide 480 audio frames per 10ms but AAudio asks for 192 |
| 120 | // in each callback (once every 4th ms). This class can then ask for 192 and |
| 121 | // the FineAudioBuffer will ask WebRTC for new data approximately only every |
| 122 | // second callback and also cache non-utilized audio. |
| 123 | std::unique_ptr<FineAudioBuffer> fine_audio_buffer_; |
| 124 | |
| 125 | // Counts number of detected underrun events reported by AAudio. |
| 126 | int32_t underrun_count_ = 0; |
| 127 | |
| 128 | // True only for the first data callback in each audio session. |
| 129 | bool first_data_callback_ = true; |
| 130 | |
| 131 | // Raw pointer handle provided to us in AttachAudioBuffer(). Owned by the |
| 132 | // AudioDeviceModuleImpl class and set by AudioDeviceModule::Create(). |
| 133 | AudioDeviceBuffer* audio_device_buffer_ RTC_GUARDED_BY(main_thread_checker_) = |
| 134 | nullptr; |
| 135 | |
| 136 | bool initialized_ RTC_GUARDED_BY(main_thread_checker_) = false; |
| 137 | bool playing_ RTC_GUARDED_BY(main_thread_checker_) = false; |
| 138 | |
| 139 | // Estimated latency between writing an audio frame to the output stream and |
| 140 | // the time that same frame is played out on the output audio device. |
| 141 | double latency_millis_ RTC_GUARDED_BY(thread_checker_aaudio_) = 0; |
| 142 | }; |
| 143 | |
| 144 | } // namespace webrtc |
| 145 | |
| 146 | #endif // MODULES_AUDIO_DEVICE_ANDROID_AAUDIO_PLAYER_H_ |