Objective-C++ style guide changes for iOS ADM
BUG=NONE Review URL: https://codereview.webrtc.org/1379583002 Cr-Commit-Position: refs/heads/master@{#10135}
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@ -43,21 +43,21 @@ class AudioDeviceIOS : public AudioDeviceGeneric {
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int32_t Init() override;
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int32_t Terminate() override;
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bool Initialized() const override { return _initialized; }
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bool Initialized() const override { return initialized_; }
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int32_t InitPlayout() override;
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bool PlayoutIsInitialized() const override { return _playIsInitialized; }
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bool PlayoutIsInitialized() const override { return play_is_initialized_; }
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int32_t InitRecording() override;
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bool RecordingIsInitialized() const override { return _recIsInitialized; }
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bool RecordingIsInitialized() const override { return rec_is_initialized_; }
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int32_t StartPlayout() override;
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int32_t StopPlayout() override;
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bool Playing() const override { return _playing; }
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bool Playing() const override { return playing_; }
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int32_t StartRecording() override;
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int32_t StopRecording() override;
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bool Recording() const override { return _recording; }
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bool Recording() const override { return recording_; }
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int32_t SetLoudspeakerStatus(bool enable) override;
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int32_t GetLoudspeakerStatus(bool& enabled) const override;
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@ -145,13 +145,13 @@ class AudioDeviceIOS : public AudioDeviceGeneric {
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bool PlayoutError() const override;
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bool RecordingWarning() const override;
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bool RecordingError() const override;
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void ClearPlayoutWarning() override{};
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void ClearPlayoutError() override{};
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void ClearRecordingWarning() override{};
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void ClearRecordingError() override{};
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void ClearPlayoutWarning() override {}
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void ClearPlayoutError() override {}
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void ClearRecordingWarning() override {}
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void ClearRecordingError() override {}
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private:
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// Uses current |_playoutParameters| and |_recordParameters| to inform the
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// Uses current |playout_parameters_| and |record_parameters_| to inform the
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// audio device buffer (ADB) about our internal audio parameters.
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void UpdateAudioDeviceBuffer();
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@ -159,7 +159,7 @@ class AudioDeviceIOS : public AudioDeviceGeneric {
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// values may be different once the AVAudioSession has been activated.
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// This method asks for the current hardware parameters and takes actions
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// if they should differ from what we have asked for initially. It also
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// defines |_playoutParameters| and |_recordParameters|.
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// defines |playout_parameters_| and |record_parameters_|.
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void SetupAudioBuffersForActiveAudioSession();
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// Creates a Voice-Processing I/O unit and configures it for full-duplex
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@ -168,7 +168,7 @@ class AudioDeviceIOS : public AudioDeviceGeneric {
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// This method also initializes the created audio unit.
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bool SetupAndInitializeVoiceProcessingAudioUnit();
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// Activates our audio session, creates and initilizes the voice-processing
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// Activates our audio session, creates and initializes the voice-processing
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// audio unit and verifies that we got the preferred native audio parameters.
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bool InitPlayOrRecord();
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@ -178,39 +178,40 @@ class AudioDeviceIOS : public AudioDeviceGeneric {
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// Callback function called on a real-time priority I/O thread from the audio
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// unit. This method is used to signal that recorded audio is available.
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static OSStatus RecordedDataIsAvailable(
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void* inRefCon,
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AudioUnitRenderActionFlags* ioActionFlags,
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const AudioTimeStamp* timeStamp,
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UInt32 inBusNumber,
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UInt32 inNumberFrames,
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AudioBufferList* ioData);
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OSStatus OnRecordedDataIsAvailable(AudioUnitRenderActionFlags* ioActionFlags,
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const AudioTimeStamp* timeStamp,
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UInt32 inBusNumber,
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UInt32 inNumberFrames);
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void* in_ref_con,
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AudioUnitRenderActionFlags* io_action_flags,
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const AudioTimeStamp* time_stamp,
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UInt32 in_bus_number,
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UInt32 in_number_frames,
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AudioBufferList* io_data);
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OSStatus OnRecordedDataIsAvailable(
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AudioUnitRenderActionFlags* io_action_flags,
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const AudioTimeStamp* time_stamp,
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UInt32 in_bus_number,
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UInt32 in_number_frames);
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// Callback function called on a real-time priority I/O thread from the audio
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// unit. This method is used to provide audio samples to the audio unit.
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static OSStatus GetPlayoutData(void* inRefCon,
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AudioUnitRenderActionFlags* ioActionFlags,
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const AudioTimeStamp* timeStamp,
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UInt32 inBusNumber,
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UInt32 inNumberFrames,
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AudioBufferList* ioData);
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OSStatus OnGetPlayoutData(AudioUnitRenderActionFlags* ioActionFlags,
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UInt32 inNumberFrames,
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AudioBufferList* ioData);
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static OSStatus GetPlayoutData(void* in_ref_con,
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AudioUnitRenderActionFlags* io_action_flags,
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const AudioTimeStamp* time_stamp,
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UInt32 in_bus_number,
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UInt32 in_number_frames,
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AudioBufferList* io_data);
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OSStatus OnGetPlayoutData(AudioUnitRenderActionFlags* io_action_flags,
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UInt32 in_number_frames,
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AudioBufferList* io_data);
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private:
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// Ensures that methods are called from the same thread as this object is
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// created on.
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rtc::ThreadChecker _threadChecker;
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rtc::ThreadChecker thread_checker_;
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// Raw pointer handle provided to us in AttachAudioBuffer(). Owned by the
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// AudioDeviceModuleImpl class and called by AudioDeviceModuleImpl::Create().
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// The AudioDeviceBuffer is a member of the AudioDeviceModuleImpl instance
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// and therefore outlives this object.
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AudioDeviceBuffer* _audioDeviceBuffer;
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AudioDeviceBuffer* audio_device_buffer_;
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// Contains audio parameters (sample rate, #channels, buffer size etc.) for
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// the playout and recording sides. These structure is set in two steps:
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@ -220,15 +221,15 @@ class AudioDeviceIOS : public AudioDeviceGeneric {
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// component to the parameters; the native I/O buffer duration.
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// A RTC_CHECK will be hit if we for some reason fail to open an audio session
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// using the specified parameters.
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AudioParameters _playoutParameters;
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AudioParameters _recordParameters;
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AudioParameters playout_parameters_;
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AudioParameters record_parameters_;
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// The Voice-Processing I/O unit has the same characteristics as the
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// Remote I/O unit (supports full duplex low-latency audio input and output)
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// and adds AEC for for two-way duplex communication. It also adds AGC,
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// adjustment of voice-processing quality, and muting. Hence, ideal for
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// VoIP applications.
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AudioUnit _vpioUnit;
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AudioUnit vpio_unit_;
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// FineAudioBuffer takes an AudioDeviceBuffer which delivers audio data
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// in chunks of 10ms. It then allows for this data to be pulled in
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@ -244,37 +245,37 @@ class AudioDeviceIOS : public AudioDeviceGeneric {
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// can provide audio data frames of size 128 and these are accumulated until
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// enough data to supply one 10ms call exists. This 10ms chunk is then sent
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// to WebRTC and the remaining part is stored.
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rtc::scoped_ptr<FineAudioBuffer> _fineAudioBuffer;
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rtc::scoped_ptr<FineAudioBuffer> fine_audio_buffer_;
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// Extra audio buffer to be used by the playout side for rendering audio.
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// The buffer size is given by FineAudioBuffer::RequiredBufferSizeBytes().
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rtc::scoped_ptr<SInt8[]> _playoutAudioBuffer;
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rtc::scoped_ptr<SInt8[]> playout_audio_buffer_;
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// Provides a mechanism for encapsulating one or more buffers of audio data.
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// Only used on the recording side.
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AudioBufferList _audioRecordBufferList;
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AudioBufferList audio_record_buffer_list_;
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// Temporary storage for recorded data. AudioUnitRender() renders into this
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// array as soon as a frame of the desired buffer size has been recorded.
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rtc::scoped_ptr<SInt8[]> _recordAudioBuffer;
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rtc::scoped_ptr<SInt8[]> record_audio_buffer_;
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// Set to 1 when recording is active and 0 otherwise.
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volatile int _recording;
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volatile int recording_;
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// Set to 1 when playout is active and 0 otherwise.
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volatile int _playing;
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volatile int playing_;
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// Set to true after successful call to Init(), false otherwise.
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bool _initialized;
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bool initialized_;
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// Set to true after successful call to InitRecording(), false otherwise.
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bool _recIsInitialized;
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bool rec_is_initialized_;
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// Set to true after successful call to InitPlayout(), false otherwise.
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bool _playIsInitialized;
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bool play_is_initialized_;
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// Audio interruption observer instance.
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void* _audioInterruptionObserver;
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void* audio_interruption_observer_;
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};
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} // namespace webrtc
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@ -177,34 +177,34 @@ static void LogDeviceInfo() {
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#endif // !defined(NDEBUG)
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AudioDeviceIOS::AudioDeviceIOS()
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: _audioDeviceBuffer(nullptr),
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_vpioUnit(nullptr),
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_recording(0),
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_playing(0),
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_initialized(false),
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_recIsInitialized(false),
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_playIsInitialized(false),
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_audioInterruptionObserver(nullptr) {
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: audio_device_buffer_(nullptr),
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vpio_unit_(nullptr),
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recording_(0),
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playing_(0),
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initialized_(false),
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rec_is_initialized_(false),
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play_is_initialized_(false),
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audio_interruption_observer_(nullptr) {
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LOGI() << "ctor" << ios::GetCurrentThreadDescription();
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}
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AudioDeviceIOS::~AudioDeviceIOS() {
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LOGI() << "~dtor";
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RTC_DCHECK(_threadChecker.CalledOnValidThread());
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RTC_DCHECK(thread_checker_.CalledOnValidThread());
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Terminate();
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}
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void AudioDeviceIOS::AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) {
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LOGI() << "AttachAudioBuffer";
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RTC_DCHECK(audioBuffer);
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RTC_DCHECK(_threadChecker.CalledOnValidThread());
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_audioDeviceBuffer = audioBuffer;
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RTC_DCHECK(thread_checker_.CalledOnValidThread());
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audio_device_buffer_ = audioBuffer;
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}
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int32_t AudioDeviceIOS::Init() {
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LOGI() << "Init";
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RTC_DCHECK(_threadChecker.CalledOnValidThread());
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if (_initialized) {
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RTC_DCHECK(thread_checker_.CalledOnValidThread());
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if (initialized_) {
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return 0;
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}
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#if !defined(NDEBUG)
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@ -214,119 +214,119 @@ int32_t AudioDeviceIOS::Init() {
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// here. They have not been set and confirmed yet since ActivateAudioSession()
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// is not called until audio is about to start. However, it makes sense to
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// store the parameters now and then verify at a later stage.
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_playoutParameters.reset(kPreferredSampleRate, kPreferredNumberOfChannels);
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_recordParameters.reset(kPreferredSampleRate, kPreferredNumberOfChannels);
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playout_parameters_.reset(kPreferredSampleRate, kPreferredNumberOfChannels);
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record_parameters_.reset(kPreferredSampleRate, kPreferredNumberOfChannels);
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// Ensure that the audio device buffer (ADB) knows about the internal audio
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// parameters. Note that, even if we are unable to get a mono audio session,
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// we will always tell the I/O audio unit to do a channel format conversion
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// to guarantee mono on the "input side" of the audio unit.
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UpdateAudioDeviceBuffer();
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_initialized = true;
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initialized_ = true;
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return 0;
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}
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int32_t AudioDeviceIOS::Terminate() {
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LOGI() << "Terminate";
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RTC_DCHECK(_threadChecker.CalledOnValidThread());
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if (!_initialized) {
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RTC_DCHECK(thread_checker_.CalledOnValidThread());
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if (!initialized_) {
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return 0;
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}
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ShutdownPlayOrRecord();
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_initialized = false;
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initialized_ = false;
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return 0;
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}
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int32_t AudioDeviceIOS::InitPlayout() {
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LOGI() << "InitPlayout";
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RTC_DCHECK(_threadChecker.CalledOnValidThread());
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RTC_DCHECK(_initialized);
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RTC_DCHECK(!_playIsInitialized);
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RTC_DCHECK(!_playing);
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if (!_recIsInitialized) {
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RTC_DCHECK(thread_checker_.CalledOnValidThread());
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RTC_DCHECK(initialized_);
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RTC_DCHECK(!play_is_initialized_);
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RTC_DCHECK(!playing_);
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if (!rec_is_initialized_) {
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if (!InitPlayOrRecord()) {
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LOG_F(LS_ERROR) << "InitPlayOrRecord failed!";
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return -1;
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}
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}
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_playIsInitialized = true;
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play_is_initialized_ = true;
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return 0;
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}
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int32_t AudioDeviceIOS::InitRecording() {
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LOGI() << "InitRecording";
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RTC_DCHECK(_threadChecker.CalledOnValidThread());
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RTC_DCHECK(_initialized);
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RTC_DCHECK(!_recIsInitialized);
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RTC_DCHECK(!_recording);
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if (!_playIsInitialized) {
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RTC_DCHECK(thread_checker_.CalledOnValidThread());
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RTC_DCHECK(initialized_);
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RTC_DCHECK(!rec_is_initialized_);
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RTC_DCHECK(!recording_);
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if (!play_is_initialized_) {
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if (!InitPlayOrRecord()) {
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LOG_F(LS_ERROR) << "InitPlayOrRecord failed!";
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return -1;
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}
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}
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_recIsInitialized = true;
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rec_is_initialized_ = true;
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return 0;
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}
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int32_t AudioDeviceIOS::StartPlayout() {
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LOGI() << "StartPlayout";
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RTC_DCHECK(_threadChecker.CalledOnValidThread());
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RTC_DCHECK(_playIsInitialized);
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RTC_DCHECK(!_playing);
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_fineAudioBuffer->ResetPlayout();
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if (!_recording) {
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OSStatus result = AudioOutputUnitStart(_vpioUnit);
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RTC_DCHECK(thread_checker_.CalledOnValidThread());
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RTC_DCHECK(play_is_initialized_);
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RTC_DCHECK(!playing_);
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fine_audio_buffer_->ResetPlayout();
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if (!recording_) {
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OSStatus result = AudioOutputUnitStart(vpio_unit_);
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if (result != noErr) {
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LOG_F(LS_ERROR) << "AudioOutputUnitStart failed: " << result;
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return -1;
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}
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}
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rtc::AtomicOps::ReleaseStore(&_playing, 1);
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rtc::AtomicOps::ReleaseStore(&playing_, 1);
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return 0;
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}
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int32_t AudioDeviceIOS::StopPlayout() {
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LOGI() << "StopPlayout";
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RTC_DCHECK(_threadChecker.CalledOnValidThread());
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if (!_playIsInitialized || !_playing) {
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RTC_DCHECK(thread_checker_.CalledOnValidThread());
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if (!play_is_initialized_ || !playing_) {
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return 0;
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}
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if (!_recording) {
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if (!recording_) {
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ShutdownPlayOrRecord();
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}
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_playIsInitialized = false;
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rtc::AtomicOps::ReleaseStore(&_playing, 0);
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play_is_initialized_ = false;
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rtc::AtomicOps::ReleaseStore(&playing_, 0);
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return 0;
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}
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int32_t AudioDeviceIOS::StartRecording() {
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LOGI() << "StartRecording";
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RTC_DCHECK(_threadChecker.CalledOnValidThread());
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RTC_DCHECK(_recIsInitialized);
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RTC_DCHECK(!_recording);
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_fineAudioBuffer->ResetRecord();
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if (!_playing) {
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OSStatus result = AudioOutputUnitStart(_vpioUnit);
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RTC_DCHECK(thread_checker_.CalledOnValidThread());
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RTC_DCHECK(rec_is_initialized_);
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RTC_DCHECK(!recording_);
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fine_audio_buffer_->ResetRecord();
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if (!playing_) {
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OSStatus result = AudioOutputUnitStart(vpio_unit_);
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if (result != noErr) {
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LOG_F(LS_ERROR) << "AudioOutputUnitStart failed: " << result;
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return -1;
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}
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}
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rtc::AtomicOps::ReleaseStore(&_recording, 1);
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rtc::AtomicOps::ReleaseStore(&recording_, 1);
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return 0;
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}
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int32_t AudioDeviceIOS::StopRecording() {
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LOGI() << "StopRecording";
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RTC_DCHECK(_threadChecker.CalledOnValidThread());
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if (!_recIsInitialized || !_recording) {
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RTC_DCHECK(thread_checker_.CalledOnValidThread());
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if (!rec_is_initialized_ || !recording_) {
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return 0;
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}
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if (!_playing) {
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if (!playing_) {
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ShutdownPlayOrRecord();
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}
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_recIsInitialized = false;
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rtc::AtomicOps::ReleaseStore(&_recording, 0);
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rec_is_initialized_ = false;
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rtc::AtomicOps::ReleaseStore(&recording_, 0);
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return 0;
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}
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@ -377,17 +377,17 @@ int32_t AudioDeviceIOS::RecordingDelay(uint16_t& delayMS) const {
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int AudioDeviceIOS::GetPlayoutAudioParameters(AudioParameters* params) const {
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LOGI() << "GetPlayoutAudioParameters";
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RTC_DCHECK(_playoutParameters.is_valid());
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RTC_DCHECK(_threadChecker.CalledOnValidThread());
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*params = _playoutParameters;
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RTC_DCHECK(playout_parameters_.is_valid());
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RTC_DCHECK(thread_checker_.CalledOnValidThread());
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*params = playout_parameters_;
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return 0;
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}
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int AudioDeviceIOS::GetRecordAudioParameters(AudioParameters* params) const {
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LOGI() << "GetRecordAudioParameters";
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RTC_DCHECK(_recordParameters.is_valid());
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RTC_DCHECK(_threadChecker.CalledOnValidThread());
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*params = _recordParameters;
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RTC_DCHECK(record_parameters_.is_valid());
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RTC_DCHECK(thread_checker_.CalledOnValidThread());
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*params = record_parameters_;
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return 0;
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}
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@ -395,12 +395,13 @@ void AudioDeviceIOS::UpdateAudioDeviceBuffer() {
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LOGI() << "UpdateAudioDevicebuffer";
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// AttachAudioBuffer() is called at construction by the main class but check
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// just in case.
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RTC_DCHECK(_audioDeviceBuffer) << "AttachAudioBuffer must be called first";
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RTC_DCHECK(audio_device_buffer_) << "AttachAudioBuffer must be called first";
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// Inform the audio device buffer (ADB) about the new audio format.
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_audioDeviceBuffer->SetPlayoutSampleRate(_playoutParameters.sample_rate());
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_audioDeviceBuffer->SetPlayoutChannels(_playoutParameters.channels());
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_audioDeviceBuffer->SetRecordingSampleRate(_recordParameters.sample_rate());
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_audioDeviceBuffer->SetRecordingChannels(_recordParameters.channels());
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audio_device_buffer_->SetPlayoutSampleRate(playout_parameters_.sample_rate());
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audio_device_buffer_->SetPlayoutChannels(playout_parameters_.channels());
|
||||
audio_device_buffer_->SetRecordingSampleRate(
|
||||
record_parameters_.sample_rate());
|
||||
audio_device_buffer_->SetRecordingChannels(record_parameters_.channels());
|
||||
}
|
||||
|
||||
void AudioDeviceIOS::SetupAudioBuffersForActiveAudioSession() {
|
||||
@ -416,7 +417,7 @@ void AudioDeviceIOS::SetupAudioBuffersForActiveAudioSession() {
|
||||
// Log a warning message for the case when we are unable to set the preferred
|
||||
// hardware sample rate but continue and use the non-ideal sample rate after
|
||||
// reinitializing the audio parameters.
|
||||
if (session.sampleRate != _playoutParameters.sample_rate()) {
|
||||
if (session.sampleRate != playout_parameters_.sample_rate()) {
|
||||
LOG(LS_WARNING)
|
||||
<< "Failed to enable an audio session with the preferred sample rate!";
|
||||
}
|
||||
@ -426,18 +427,18 @@ void AudioDeviceIOS::SetupAudioBuffersForActiveAudioSession() {
|
||||
// number of audio frames.
|
||||
// Example: IO buffer size = 0.008 seconds <=> 128 audio frames at 16kHz.
|
||||
// Hence, 128 is the size we expect to see in upcoming render callbacks.
|
||||
_playoutParameters.reset(session.sampleRate, _playoutParameters.channels(),
|
||||
playout_parameters_.reset(session.sampleRate, playout_parameters_.channels(),
|
||||
session.IOBufferDuration);
|
||||
RTC_DCHECK(playout_parameters_.is_complete());
|
||||
record_parameters_.reset(session.sampleRate, record_parameters_.channels(),
|
||||
session.IOBufferDuration);
|
||||
RTC_DCHECK(_playoutParameters.is_complete());
|
||||
_recordParameters.reset(session.sampleRate, _recordParameters.channels(),
|
||||
session.IOBufferDuration);
|
||||
RTC_DCHECK(_recordParameters.is_complete());
|
||||
RTC_DCHECK(record_parameters_.is_complete());
|
||||
LOG(LS_INFO) << " frames per I/O buffer: "
|
||||
<< _playoutParameters.frames_per_buffer();
|
||||
<< playout_parameters_.frames_per_buffer();
|
||||
LOG(LS_INFO) << " bytes per I/O buffer: "
|
||||
<< _playoutParameters.GetBytesPerBuffer();
|
||||
RTC_DCHECK_EQ(_playoutParameters.GetBytesPerBuffer(),
|
||||
_recordParameters.GetBytesPerBuffer());
|
||||
<< playout_parameters_.GetBytesPerBuffer();
|
||||
RTC_DCHECK_EQ(playout_parameters_.GetBytesPerBuffer(),
|
||||
record_parameters_.GetBytesPerBuffer());
|
||||
|
||||
// Update the ADB parameters since the sample rate might have changed.
|
||||
UpdateAudioDeviceBuffer();
|
||||
@ -445,71 +446,71 @@ void AudioDeviceIOS::SetupAudioBuffersForActiveAudioSession() {
|
||||
// Create a modified audio buffer class which allows us to ask for,
|
||||
// or deliver, any number of samples (and not only multiple of 10ms) to match
|
||||
// the native audio unit buffer size.
|
||||
RTC_DCHECK(_audioDeviceBuffer);
|
||||
_fineAudioBuffer.reset(new FineAudioBuffer(
|
||||
_audioDeviceBuffer, _playoutParameters.GetBytesPerBuffer(),
|
||||
_playoutParameters.sample_rate()));
|
||||
RTC_DCHECK(audio_device_buffer_);
|
||||
fine_audio_buffer_.reset(new FineAudioBuffer(
|
||||
audio_device_buffer_, playout_parameters_.GetBytesPerBuffer(),
|
||||
playout_parameters_.sample_rate()));
|
||||
|
||||
// The extra/temporary playoutbuffer must be of this size to avoid
|
||||
// unnecessary memcpy while caching data between successive callbacks.
|
||||
const int requiredPlayoutBufferSize =
|
||||
_fineAudioBuffer->RequiredPlayoutBufferSizeBytes();
|
||||
const int required_playout_buffer_size =
|
||||
fine_audio_buffer_->RequiredPlayoutBufferSizeBytes();
|
||||
LOG(LS_INFO) << " required playout buffer size: "
|
||||
<< requiredPlayoutBufferSize;
|
||||
_playoutAudioBuffer.reset(new SInt8[requiredPlayoutBufferSize]);
|
||||
<< required_playout_buffer_size;
|
||||
playout_audio_buffer_.reset(new SInt8[required_playout_buffer_size]);
|
||||
|
||||
// Allocate AudioBuffers to be used as storage for the received audio.
|
||||
// The AudioBufferList structure works as a placeholder for the
|
||||
// AudioBuffer structure, which holds a pointer to the actual data buffer
|
||||
// in |_recordAudioBuffer|. Recorded audio will be rendered into this memory
|
||||
// in |record_audio_buffer_|. Recorded audio will be rendered into this memory
|
||||
// at each input callback when calling AudioUnitRender().
|
||||
const int dataByteSize = _recordParameters.GetBytesPerBuffer();
|
||||
_recordAudioBuffer.reset(new SInt8[dataByteSize]);
|
||||
_audioRecordBufferList.mNumberBuffers = 1;
|
||||
AudioBuffer* audioBuffer = &_audioRecordBufferList.mBuffers[0];
|
||||
audioBuffer->mNumberChannels = _recordParameters.channels();
|
||||
audioBuffer->mDataByteSize = dataByteSize;
|
||||
audioBuffer->mData = _recordAudioBuffer.get();
|
||||
const int data_byte_size = record_parameters_.GetBytesPerBuffer();
|
||||
record_audio_buffer_.reset(new SInt8[data_byte_size]);
|
||||
audio_record_buffer_list_.mNumberBuffers = 1;
|
||||
AudioBuffer* audio_buffer = &audio_record_buffer_list_.mBuffers[0];
|
||||
audio_buffer->mNumberChannels = record_parameters_.channels();
|
||||
audio_buffer->mDataByteSize = data_byte_size;
|
||||
audio_buffer->mData = record_audio_buffer_.get();
|
||||
}
|
||||
|
||||
bool AudioDeviceIOS::SetupAndInitializeVoiceProcessingAudioUnit() {
|
||||
LOGI() << "SetupAndInitializeVoiceProcessingAudioUnit";
|
||||
RTC_DCHECK(!_vpioUnit);
|
||||
RTC_DCHECK(!vpio_unit_);
|
||||
// Create an audio component description to identify the Voice-Processing
|
||||
// I/O audio unit.
|
||||
AudioComponentDescription vpioUnitDescription;
|
||||
vpioUnitDescription.componentType = kAudioUnitType_Output;
|
||||
vpioUnitDescription.componentSubType = kAudioUnitSubType_VoiceProcessingIO;
|
||||
vpioUnitDescription.componentManufacturer = kAudioUnitManufacturer_Apple;
|
||||
vpioUnitDescription.componentFlags = 0;
|
||||
vpioUnitDescription.componentFlagsMask = 0;
|
||||
AudioComponentDescription vpio_unit_description;
|
||||
vpio_unit_description.componentType = kAudioUnitType_Output;
|
||||
vpio_unit_description.componentSubType = kAudioUnitSubType_VoiceProcessingIO;
|
||||
vpio_unit_description.componentManufacturer = kAudioUnitManufacturer_Apple;
|
||||
vpio_unit_description.componentFlags = 0;
|
||||
vpio_unit_description.componentFlagsMask = 0;
|
||||
// Obtain an audio unit instance given the description.
|
||||
AudioComponent foundVpioUnitRef =
|
||||
AudioComponentFindNext(nullptr, &vpioUnitDescription);
|
||||
AudioComponent found_vpio_unit_ref =
|
||||
AudioComponentFindNext(nullptr, &vpio_unit_description);
|
||||
|
||||
// Create a Voice-Processing IO audio unit.
|
||||
LOG_AND_RETURN_IF_ERROR(
|
||||
AudioComponentInstanceNew(foundVpioUnitRef, &_vpioUnit),
|
||||
AudioComponentInstanceNew(found_vpio_unit_ref, &vpio_unit_),
|
||||
"Failed to create a VoiceProcessingIO audio unit");
|
||||
|
||||
// A VP I/O unit's bus 1 connects to input hardware (microphone). Enable
|
||||
// input on the input scope of the input element.
|
||||
AudioUnitElement inputBus = 1;
|
||||
UInt32 enableInput = 1;
|
||||
AudioUnitElement input_bus = 1;
|
||||
UInt32 enable_input = 1;
|
||||
LOG_AND_RETURN_IF_ERROR(
|
||||
AudioUnitSetProperty(_vpioUnit, kAudioOutputUnitProperty_EnableIO,
|
||||
kAudioUnitScope_Input, inputBus, &enableInput,
|
||||
sizeof(enableInput)),
|
||||
AudioUnitSetProperty(vpio_unit_, kAudioOutputUnitProperty_EnableIO,
|
||||
kAudioUnitScope_Input, input_bus, &enable_input,
|
||||
sizeof(enable_input)),
|
||||
"Failed to enable input on input scope of input element");
|
||||
|
||||
// A VP I/O unit's bus 0 connects to output hardware (speaker). Enable
|
||||
// output on the output scope of the output element.
|
||||
AudioUnitElement outputBus = 0;
|
||||
UInt32 enableOutput = 1;
|
||||
AudioUnitElement output_bus = 0;
|
||||
UInt32 enable_output = 1;
|
||||
LOG_AND_RETURN_IF_ERROR(
|
||||
AudioUnitSetProperty(_vpioUnit, kAudioOutputUnitProperty_EnableIO,
|
||||
kAudioUnitScope_Output, outputBus, &enableOutput,
|
||||
sizeof(enableOutput)),
|
||||
AudioUnitSetProperty(vpio_unit_, kAudioOutputUnitProperty_EnableIO,
|
||||
kAudioUnitScope_Output, output_bus, &enable_output,
|
||||
sizeof(enable_output)),
|
||||
"Failed to enable output on output scope of output element");
|
||||
|
||||
// Set the application formats for input and output:
|
||||
@ -517,72 +518,73 @@ bool AudioDeviceIOS::SetupAndInitializeVoiceProcessingAudioUnit() {
|
||||
// - avoid resampling in the I/O unit by using the hardware sample rate
|
||||
// - linear PCM => noncompressed audio data format with one frame per packet
|
||||
// - no need to specify interleaving since only mono is supported
|
||||
AudioStreamBasicDescription applicationFormat = {0};
|
||||
UInt32 size = sizeof(applicationFormat);
|
||||
RTC_DCHECK_EQ(_playoutParameters.sample_rate(),
|
||||
_recordParameters.sample_rate());
|
||||
AudioStreamBasicDescription application_format = {0};
|
||||
UInt32 size = sizeof(application_format);
|
||||
RTC_DCHECK_EQ(playout_parameters_.sample_rate(),
|
||||
record_parameters_.sample_rate());
|
||||
RTC_DCHECK_EQ(1, kPreferredNumberOfChannels);
|
||||
applicationFormat.mSampleRate = _playoutParameters.sample_rate();
|
||||
applicationFormat.mFormatID = kAudioFormatLinearPCM;
|
||||
applicationFormat.mFormatFlags =
|
||||
application_format.mSampleRate = playout_parameters_.sample_rate();
|
||||
application_format.mFormatID = kAudioFormatLinearPCM;
|
||||
application_format.mFormatFlags =
|
||||
kLinearPCMFormatFlagIsSignedInteger | kLinearPCMFormatFlagIsPacked;
|
||||
applicationFormat.mBytesPerPacket = kBytesPerSample;
|
||||
applicationFormat.mFramesPerPacket = 1; // uncompressed
|
||||
applicationFormat.mBytesPerFrame = kBytesPerSample;
|
||||
applicationFormat.mChannelsPerFrame = kPreferredNumberOfChannels;
|
||||
applicationFormat.mBitsPerChannel = 8 * kBytesPerSample;
|
||||
application_format.mBytesPerPacket = kBytesPerSample;
|
||||
application_format.mFramesPerPacket = 1; // uncompressed
|
||||
application_format.mBytesPerFrame = kBytesPerSample;
|
||||
application_format.mChannelsPerFrame = kPreferredNumberOfChannels;
|
||||
application_format.mBitsPerChannel = 8 * kBytesPerSample;
|
||||
#if !defined(NDEBUG)
|
||||
LogABSD(applicationFormat);
|
||||
LogABSD(application_format);
|
||||
#endif
|
||||
|
||||
// Set the application format on the output scope of the input element/bus.
|
||||
LOG_AND_RETURN_IF_ERROR(
|
||||
AudioUnitSetProperty(_vpioUnit, kAudioUnitProperty_StreamFormat,
|
||||
kAudioUnitScope_Output, inputBus, &applicationFormat,
|
||||
size),
|
||||
AudioUnitSetProperty(vpio_unit_, kAudioUnitProperty_StreamFormat,
|
||||
kAudioUnitScope_Output, input_bus,
|
||||
&application_format, size),
|
||||
"Failed to set application format on output scope of input element");
|
||||
|
||||
// Set the application format on the input scope of the output element/bus.
|
||||
LOG_AND_RETURN_IF_ERROR(
|
||||
AudioUnitSetProperty(_vpioUnit, kAudioUnitProperty_StreamFormat,
|
||||
kAudioUnitScope_Input, outputBus, &applicationFormat,
|
||||
size),
|
||||
AudioUnitSetProperty(vpio_unit_, kAudioUnitProperty_StreamFormat,
|
||||
kAudioUnitScope_Input, output_bus,
|
||||
&application_format, size),
|
||||
"Failed to set application format on input scope of output element");
|
||||
|
||||
// Specify the callback function that provides audio samples to the audio
|
||||
// unit.
|
||||
AURenderCallbackStruct renderCallback;
|
||||
renderCallback.inputProc = GetPlayoutData;
|
||||
renderCallback.inputProcRefCon = this;
|
||||
AURenderCallbackStruct render_callback;
|
||||
render_callback.inputProc = GetPlayoutData;
|
||||
render_callback.inputProcRefCon = this;
|
||||
LOG_AND_RETURN_IF_ERROR(
|
||||
AudioUnitSetProperty(_vpioUnit, kAudioUnitProperty_SetRenderCallback,
|
||||
kAudioUnitScope_Input, outputBus, &renderCallback,
|
||||
sizeof(renderCallback)),
|
||||
AudioUnitSetProperty(vpio_unit_, kAudioUnitProperty_SetRenderCallback,
|
||||
kAudioUnitScope_Input, output_bus, &render_callback,
|
||||
sizeof(render_callback)),
|
||||
"Failed to specify the render callback on the output element");
|
||||
|
||||
// Disable AU buffer allocation for the recorder, we allocate our own.
|
||||
// TODO(henrika): not sure that it actually saves resource to make this call.
|
||||
UInt32 flag = 0;
|
||||
LOG_AND_RETURN_IF_ERROR(
|
||||
AudioUnitSetProperty(_vpioUnit, kAudioUnitProperty_ShouldAllocateBuffer,
|
||||
kAudioUnitScope_Output, inputBus, &flag,
|
||||
AudioUnitSetProperty(vpio_unit_, kAudioUnitProperty_ShouldAllocateBuffer,
|
||||
kAudioUnitScope_Output, input_bus, &flag,
|
||||
sizeof(flag)),
|
||||
"Failed to disable buffer allocation on the input element");
|
||||
|
||||
// Specify the callback to be called by the I/O thread to us when input audio
|
||||
// is available. The recorded samples can then be obtained by calling the
|
||||
// AudioUnitRender() method.
|
||||
AURenderCallbackStruct inputCallback;
|
||||
inputCallback.inputProc = RecordedDataIsAvailable;
|
||||
inputCallback.inputProcRefCon = this;
|
||||
AURenderCallbackStruct input_callback;
|
||||
input_callback.inputProc = RecordedDataIsAvailable;
|
||||
input_callback.inputProcRefCon = this;
|
||||
LOG_AND_RETURN_IF_ERROR(
|
||||
AudioUnitSetProperty(_vpioUnit, kAudioOutputUnitProperty_SetInputCallback,
|
||||
kAudioUnitScope_Global, inputBus, &inputCallback,
|
||||
sizeof(inputCallback)),
|
||||
AudioUnitSetProperty(vpio_unit_,
|
||||
kAudioOutputUnitProperty_SetInputCallback,
|
||||
kAudioUnitScope_Global, input_bus, &input_callback,
|
||||
sizeof(input_callback)),
|
||||
"Failed to specify the input callback on the input element");
|
||||
|
||||
// Initialize the Voice-Processing I/O unit instance.
|
||||
LOG_AND_RETURN_IF_ERROR(AudioUnitInitialize(_vpioUnit),
|
||||
LOG_AND_RETURN_IF_ERROR(AudioUnitInitialize(vpio_unit_),
|
||||
"Failed to initialize the Voice-Processing I/O unit");
|
||||
return true;
|
||||
}
|
||||
@ -617,9 +619,8 @@ bool AudioDeviceIOS::InitPlayOrRecord() {
|
||||
switch (type) {
|
||||
case AVAudioSessionInterruptionTypeBegan:
|
||||
// At this point our audio session has been deactivated and
|
||||
// the
|
||||
// audio unit render callbacks no longer occur. Nothing to
|
||||
// do.
|
||||
// the audio unit render callbacks no longer occur.
|
||||
// Nothing to do.
|
||||
break;
|
||||
case AVAudioSessionInterruptionTypeEnded: {
|
||||
NSError* error = nil;
|
||||
@ -631,8 +632,8 @@ bool AudioDeviceIOS::InitPlayOrRecord() {
|
||||
// Post interruption the audio unit render callbacks don't
|
||||
// automatically continue, so we restart the unit manually
|
||||
// here.
|
||||
AudioOutputUnitStop(_vpioUnit);
|
||||
AudioOutputUnitStart(_vpioUnit);
|
||||
AudioOutputUnitStop(vpio_unit_);
|
||||
AudioOutputUnitStart(vpio_unit_);
|
||||
break;
|
||||
}
|
||||
}
|
||||
@ -640,32 +641,32 @@ bool AudioDeviceIOS::InitPlayOrRecord() {
|
||||
// Increment refcount on observer using ARC bridge. Instance variable is a
|
||||
// void* instead of an id because header is included in other pure C++
|
||||
// files.
|
||||
_audioInterruptionObserver = (__bridge_retained void*)observer;
|
||||
audio_interruption_observer_ = (__bridge_retained void*)observer;
|
||||
return true;
|
||||
}
|
||||
|
||||
bool AudioDeviceIOS::ShutdownPlayOrRecord() {
|
||||
LOGI() << "ShutdownPlayOrRecord";
|
||||
if (_audioInterruptionObserver != nullptr) {
|
||||
if (audio_interruption_observer_ != nullptr) {
|
||||
NSNotificationCenter* center = [NSNotificationCenter defaultCenter];
|
||||
// Transfer ownership of observer back to ARC, which will dealloc the
|
||||
// observer once it exits this scope.
|
||||
id observer = (__bridge_transfer id)_audioInterruptionObserver;
|
||||
id observer = (__bridge_transfer id)audio_interruption_observer_;
|
||||
[center removeObserver:observer];
|
||||
_audioInterruptionObserver = nullptr;
|
||||
audio_interruption_observer_ = nullptr;
|
||||
}
|
||||
// Close and delete the voice-processing I/O unit.
|
||||
OSStatus result = -1;
|
||||
if (nullptr != _vpioUnit) {
|
||||
result = AudioOutputUnitStop(_vpioUnit);
|
||||
if (nullptr != vpio_unit_) {
|
||||
result = AudioOutputUnitStop(vpio_unit_);
|
||||
if (result != noErr) {
|
||||
LOG_F(LS_ERROR) << "AudioOutputUnitStop failed: " << result;
|
||||
}
|
||||
result = AudioComponentInstanceDispose(_vpioUnit);
|
||||
result = AudioComponentInstanceDispose(vpio_unit_);
|
||||
if (result != noErr) {
|
||||
LOG_F(LS_ERROR) << "AudioComponentInstanceDispose failed: " << result;
|
||||
}
|
||||
_vpioUnit = nullptr;
|
||||
vpio_unit_ = nullptr;
|
||||
}
|
||||
// All I/O should be stopped or paused prior to deactivating the audio
|
||||
// session, hence we deactivate as last action.
|
||||
@ -675,36 +676,38 @@ bool AudioDeviceIOS::ShutdownPlayOrRecord() {
|
||||
}
|
||||
|
||||
OSStatus AudioDeviceIOS::RecordedDataIsAvailable(
|
||||
void* inRefCon,
|
||||
AudioUnitRenderActionFlags* ioActionFlags,
|
||||
const AudioTimeStamp* inTimeStamp,
|
||||
UInt32 inBusNumber,
|
||||
UInt32 inNumberFrames,
|
||||
AudioBufferList* ioData) {
|
||||
RTC_DCHECK_EQ(1u, inBusNumber);
|
||||
RTC_DCHECK(!ioData); // no buffer should be allocated for input at this stage
|
||||
AudioDeviceIOS* audio_device_ios = static_cast<AudioDeviceIOS*>(inRefCon);
|
||||
void* in_ref_con,
|
||||
AudioUnitRenderActionFlags* io_action_flags,
|
||||
const AudioTimeStamp* in_time_stamp,
|
||||
UInt32 in_bus_number,
|
||||
UInt32 in_number_frames,
|
||||
AudioBufferList* io_data) {
|
||||
RTC_DCHECK_EQ(1u, in_bus_number);
|
||||
RTC_DCHECK(
|
||||
!io_data); // no buffer should be allocated for input at this stage
|
||||
AudioDeviceIOS* audio_device_ios = static_cast<AudioDeviceIOS*>(in_ref_con);
|
||||
return audio_device_ios->OnRecordedDataIsAvailable(
|
||||
ioActionFlags, inTimeStamp, inBusNumber, inNumberFrames);
|
||||
io_action_flags, in_time_stamp, in_bus_number, in_number_frames);
|
||||
}
|
||||
|
||||
OSStatus AudioDeviceIOS::OnRecordedDataIsAvailable(
|
||||
AudioUnitRenderActionFlags* ioActionFlags,
|
||||
const AudioTimeStamp* inTimeStamp,
|
||||
UInt32 inBusNumber,
|
||||
UInt32 inNumberFrames) {
|
||||
RTC_DCHECK_EQ(_recordParameters.frames_per_buffer(), inNumberFrames);
|
||||
AudioUnitRenderActionFlags* io_action_flags,
|
||||
const AudioTimeStamp* in_time_stamp,
|
||||
UInt32 in_bus_number,
|
||||
UInt32 in_number_frames) {
|
||||
RTC_DCHECK_EQ(record_parameters_.frames_per_buffer(), in_number_frames);
|
||||
OSStatus result = noErr;
|
||||
// Simply return if recording is not enabled.
|
||||
if (!rtc::AtomicOps::AcquireLoad(&_recording))
|
||||
if (!rtc::AtomicOps::AcquireLoad(&recording_))
|
||||
return result;
|
||||
RTC_DCHECK_EQ(record_parameters_.frames_per_buffer(), in_number_frames);
|
||||
// Obtain the recorded audio samples by initiating a rendering cycle.
|
||||
// Since it happens on the input bus, the |ioData| parameter is a reference
|
||||
// Since it happens on the input bus, the |io_data| parameter is a reference
|
||||
// to the preallocated audio buffer list that the audio unit renders into.
|
||||
// TODO(henrika): should error handling be improved?
|
||||
AudioBufferList* ioData = &_audioRecordBufferList;
|
||||
result = AudioUnitRender(_vpioUnit, ioActionFlags, inTimeStamp, inBusNumber,
|
||||
inNumberFrames, ioData);
|
||||
AudioBufferList* io_data = &audio_record_buffer_list_;
|
||||
result = AudioUnitRender(vpio_unit_, io_action_flags, in_time_stamp,
|
||||
in_bus_number, in_number_frames, io_data);
|
||||
if (result != noErr) {
|
||||
LOG_F(LS_ERROR) << "AudioOutputUnitStart failed: " << result;
|
||||
return result;
|
||||
@ -712,53 +715,53 @@ OSStatus AudioDeviceIOS::OnRecordedDataIsAvailable(
|
||||
// Get a pointer to the recorded audio and send it to the WebRTC ADB.
|
||||
// Use the FineAudioBuffer instance to convert between native buffer size
|
||||
// and the 10ms buffer size used by WebRTC.
|
||||
const UInt32 dataSizeInBytes = ioData->mBuffers[0].mDataByteSize;
|
||||
RTC_CHECK_EQ(dataSizeInBytes / kBytesPerSample, inNumberFrames);
|
||||
SInt8* data = static_cast<SInt8*>(ioData->mBuffers[0].mData);
|
||||
_fineAudioBuffer->DeliverRecordedData(data, dataSizeInBytes,
|
||||
kFixedPlayoutDelayEstimate,
|
||||
kFixedRecordDelayEstimate);
|
||||
const UInt32 data_size_in_bytes = io_data->mBuffers[0].mDataByteSize;
|
||||
RTC_CHECK_EQ(data_size_in_bytes / kBytesPerSample, in_number_frames);
|
||||
SInt8* data = static_cast<SInt8*>(io_data->mBuffers[0].mData);
|
||||
fine_audio_buffer_->DeliverRecordedData(data, data_size_in_bytes,
|
||||
kFixedPlayoutDelayEstimate,
|
||||
kFixedRecordDelayEstimate);
|
||||
return noErr;
|
||||
}
|
||||
|
||||
OSStatus AudioDeviceIOS::GetPlayoutData(
|
||||
void* inRefCon,
|
||||
AudioUnitRenderActionFlags* ioActionFlags,
|
||||
const AudioTimeStamp* inTimeStamp,
|
||||
UInt32 inBusNumber,
|
||||
UInt32 inNumberFrames,
|
||||
AudioBufferList* ioData) {
|
||||
RTC_DCHECK_EQ(0u, inBusNumber);
|
||||
RTC_DCHECK(ioData);
|
||||
AudioDeviceIOS* audio_device_ios = static_cast<AudioDeviceIOS*>(inRefCon);
|
||||
return audio_device_ios->OnGetPlayoutData(ioActionFlags, inNumberFrames,
|
||||
ioData);
|
||||
void* in_ref_con,
|
||||
AudioUnitRenderActionFlags* io_action_flags,
|
||||
const AudioTimeStamp* in_time_stamp,
|
||||
UInt32 in_bus_number,
|
||||
UInt32 in_number_frames,
|
||||
AudioBufferList* io_data) {
|
||||
RTC_DCHECK_EQ(0u, in_bus_number);
|
||||
RTC_DCHECK(io_data);
|
||||
AudioDeviceIOS* audio_device_ios = static_cast<AudioDeviceIOS*>(in_ref_con);
|
||||
return audio_device_ios->OnGetPlayoutData(io_action_flags, in_number_frames,
|
||||
io_data);
|
||||
}
|
||||
|
||||
OSStatus AudioDeviceIOS::OnGetPlayoutData(
|
||||
AudioUnitRenderActionFlags* ioActionFlags,
|
||||
UInt32 inNumberFrames,
|
||||
AudioBufferList* ioData) {
|
||||
AudioUnitRenderActionFlags* io_action_flags,
|
||||
UInt32 in_number_frames,
|
||||
AudioBufferList* io_data) {
|
||||
// Verify 16-bit, noninterleaved mono PCM signal format.
|
||||
RTC_DCHECK_EQ(1u, ioData->mNumberBuffers);
|
||||
RTC_DCHECK_EQ(1u, ioData->mBuffers[0].mNumberChannels);
|
||||
RTC_DCHECK_EQ(1u, io_data->mNumberBuffers);
|
||||
RTC_DCHECK_EQ(1u, io_data->mBuffers[0].mNumberChannels);
|
||||
// Get pointer to internal audio buffer to which new audio data shall be
|
||||
// written.
|
||||
const UInt32 dataSizeInBytes = ioData->mBuffers[0].mDataByteSize;
|
||||
RTC_CHECK_EQ(dataSizeInBytes / kBytesPerSample, inNumberFrames);
|
||||
SInt8* destination = static_cast<SInt8*>(ioData->mBuffers[0].mData);
|
||||
const UInt32 dataSizeInBytes = io_data->mBuffers[0].mDataByteSize;
|
||||
RTC_CHECK_EQ(dataSizeInBytes / kBytesPerSample, in_number_frames);
|
||||
SInt8* destination = static_cast<SInt8*>(io_data->mBuffers[0].mData);
|
||||
// Produce silence and give audio unit a hint about it if playout is not
|
||||
// activated.
|
||||
if (!rtc::AtomicOps::AcquireLoad(&_playing)) {
|
||||
*ioActionFlags |= kAudioUnitRenderAction_OutputIsSilence;
|
||||
if (!rtc::AtomicOps::AcquireLoad(&playing_)) {
|
||||
*io_action_flags |= kAudioUnitRenderAction_OutputIsSilence;
|
||||
memset(destination, 0, dataSizeInBytes);
|
||||
return noErr;
|
||||
}
|
||||
// Read decoded 16-bit PCM samples from WebRTC (using a size that matches
|
||||
// the native I/O audio unit) to a preallocated intermediate buffer and
|
||||
// copy the result to the audio buffer in the |ioData| destination.
|
||||
SInt8* source = _playoutAudioBuffer.get();
|
||||
_fineAudioBuffer->GetPlayoutData(source);
|
||||
// copy the result to the audio buffer in the |io_data| destination.
|
||||
SInt8* source = playout_audio_buffer_.get();
|
||||
fine_audio_buffer_->GetPlayoutData(source);
|
||||
memcpy(destination, source, dataSizeInBytes);
|
||||
return noErr;
|
||||
}
|
||||
|
||||
Loading…
x
Reference in New Issue
Block a user