| /* |
| * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. |
| * |
| * Use of this source code is governed by a BSD-style license |
| * that can be found in the LICENSE file in the root of the source |
| * tree. An additional intellectual property rights grant can be found |
| * in the file PATENTS. All contributing project authors may |
| * be found in the AUTHORS file in the root of the source tree. |
| */ |
| |
| #ifndef WEBRTC_MODULES_AUDIO_DEVICE_AUDIO_DEVICE_BUFFER_H_ |
| #define WEBRTC_MODULES_AUDIO_DEVICE_AUDIO_DEVICE_BUFFER_H_ |
| |
| #include "webrtc/base/criticalsection.h" |
| #include "webrtc/base/task_queue.h" |
| #include "webrtc/base/thread_checker.h" |
| #include "webrtc/modules/audio_device/include/audio_device.h" |
| #include "webrtc/system_wrappers/include/file_wrapper.h" |
| #include "webrtc/typedefs.h" |
| |
| namespace webrtc { |
| class CriticalSectionWrapper; |
| |
| const uint32_t kPulsePeriodMs = 1000; |
| const size_t kMaxBufferSizeBytes = 3840; // 10ms in stereo @ 96kHz |
| |
| class AudioDeviceObserver; |
| |
| class AudioDeviceBuffer { |
| public: |
| AudioDeviceBuffer(); |
| virtual ~AudioDeviceBuffer(); |
| |
| void SetId(uint32_t id) {}; |
| int32_t RegisterAudioCallback(AudioTransport* audioCallback); |
| |
| int32_t InitPlayout(); |
| int32_t InitRecording(); |
| |
| virtual int32_t SetRecordingSampleRate(uint32_t fsHz); |
| virtual int32_t SetPlayoutSampleRate(uint32_t fsHz); |
| int32_t RecordingSampleRate() const; |
| int32_t PlayoutSampleRate() const; |
| |
| virtual int32_t SetRecordingChannels(size_t channels); |
| virtual int32_t SetPlayoutChannels(size_t channels); |
| size_t RecordingChannels() const; |
| size_t PlayoutChannels() const; |
| int32_t SetRecordingChannel(const AudioDeviceModule::ChannelType channel); |
| int32_t RecordingChannel(AudioDeviceModule::ChannelType& channel) const; |
| |
| virtual int32_t SetRecordedBuffer(const void* audioBuffer, size_t nSamples); |
| int32_t SetCurrentMicLevel(uint32_t level); |
| virtual void SetVQEData(int playDelayMS, int recDelayMS, int clockDrift); |
| virtual int32_t DeliverRecordedData(); |
| uint32_t NewMicLevel() const; |
| |
| virtual int32_t RequestPlayoutData(size_t nSamples); |
| virtual int32_t GetPlayoutData(void* audioBuffer); |
| |
| int32_t StartInputFileRecording(const char fileName[kAdmMaxFileNameSize]); |
| int32_t StopInputFileRecording(); |
| int32_t StartOutputFileRecording(const char fileName[kAdmMaxFileNameSize]); |
| int32_t StopOutputFileRecording(); |
| |
| int32_t SetTypingStatus(bool typingStatus); |
| |
| private: |
| // Posts the first delayed task in the task queue and starts the periodic |
| // timer. |
| void StartTimer(); |
| |
| // Called periodically on the internal thread created by the TaskQueue. |
| void LogStats(); |
| |
| // Updates counters in each play/record callback but does it on the task |
| // queue to ensure that they can be read by LogStats() without any locks since |
| // each task is serialized by the task queue. |
| void UpdateRecStats(size_t num_samples); |
| void UpdatePlayStats(size_t num_samples); |
| |
| // Ensures that methods are called on the same thread as the thread that |
| // creates this object. |
| rtc::ThreadChecker thread_checker_; |
| |
| rtc::CriticalSection _critSect; |
| rtc::CriticalSection _critSectCb; |
| |
| AudioTransport* _ptrCbAudioTransport; |
| |
| // Task queue used to invoke LogStats() periodically. Tasks are executed on a |
| // worker thread but it does not necessarily have to be the same thread for |
| // each task. |
| rtc::TaskQueue task_queue_; |
| |
| // Ensures that the timer is only started once. |
| bool timer_has_started_; |
| |
| uint32_t _recSampleRate; |
| uint32_t _playSampleRate; |
| |
| size_t _recChannels; |
| size_t _playChannels; |
| |
| // selected recording channel (left/right/both) |
| AudioDeviceModule::ChannelType _recChannel; |
| |
| // 2 or 4 depending on mono or stereo |
| size_t _recBytesPerSample; |
| size_t _playBytesPerSample; |
| |
| // 10ms in stereo @ 96kHz |
| int8_t _recBuffer[kMaxBufferSizeBytes]; |
| |
| // one sample <=> 2 or 4 bytes |
| size_t _recSamples; |
| size_t _recSize; // in bytes |
| |
| // 10ms in stereo @ 96kHz |
| int8_t _playBuffer[kMaxBufferSizeBytes]; |
| |
| // one sample <=> 2 or 4 bytes |
| size_t _playSamples; |
| size_t _playSize; // in bytes |
| |
| FileWrapper& _recFile; |
| FileWrapper& _playFile; |
| |
| uint32_t _currentMicLevel; |
| uint32_t _newMicLevel; |
| |
| bool _typingStatus; |
| |
| int _playDelayMS; |
| int _recDelayMS; |
| int _clockDrift; |
| int high_delay_counter_; |
| |
| // Counts number of times LogStats() has been called. |
| size_t num_stat_reports_; |
| |
| // Total number of recording callbacks where the source provides 10ms audio |
| // data each time. |
| uint64_t rec_callbacks_; |
| |
| // Total number of recording callbacks stored at the last timer task. |
| uint64_t last_rec_callbacks_; |
| |
| // Total number of playback callbacks where the sink asks for 10ms audio |
| // data each time. |
| uint64_t play_callbacks_; |
| |
| // Total number of playout callbacks stored at the last timer task. |
| uint64_t last_play_callbacks_; |
| |
| // Total number of recorded audio samples. |
| uint64_t rec_samples_; |
| |
| // Total number of recorded samples stored at the previous timer task. |
| uint64_t last_rec_samples_; |
| |
| // Total number of played audio samples. |
| uint64_t play_samples_; |
| |
| // Total number of played samples stored at the previous timer task. |
| uint64_t last_play_samples_; |
| |
| // Time stamp of last stat report. |
| uint64_t last_log_stat_time_; |
| }; |
| |
| } // namespace webrtc |
| |
| #endif // WEBRTC_MODULES_AUDIO_DEVICE_AUDIO_DEVICE_BUFFER_H_ |