blob: fcfe7a6e8bc55cd872af966a8753e2ec94c92c74 [file] [log] [blame]
* Copyright (c) 2022 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.
#include <memory>
#import "components/audio/RTCAudioDevice.h"
#include "modules/audio_device/audio_device_buffer.h"
#include "modules/audio_device/include/audio_device.h"
#include "rtc_base/thread.h"
@class ObjCAudioDeviceDelegate;
namespace webrtc {
class FineAudioBuffer;
namespace objc_adm {
class ObjCAudioDeviceModule : public AudioDeviceModule {
explicit ObjCAudioDeviceModule(id<RTC_OBJC_TYPE(RTCAudioDevice)> audio_device);
~ObjCAudioDeviceModule() override;
// Retrieve the currently utilized audio layer
int32_t ActiveAudioLayer(AudioLayer* audioLayer) const override;
// Full-duplex transportation of PCM audio
int32_t RegisterAudioCallback(AudioTransport* audioCallback) override;
// Main initialization and termination
int32_t Init() override;
int32_t Terminate() override;
bool Initialized() const override;
// Device enumeration
int16_t PlayoutDevices() override;
int16_t RecordingDevices() override;
int32_t PlayoutDeviceName(uint16_t index,
char name[kAdmMaxDeviceNameSize],
char guid[kAdmMaxGuidSize]) override;
int32_t RecordingDeviceName(uint16_t index,
char name[kAdmMaxDeviceNameSize],
char guid[kAdmMaxGuidSize]) override;
// Device selection
int32_t SetPlayoutDevice(uint16_t index) override;
int32_t SetPlayoutDevice(WindowsDeviceType device) override;
int32_t SetRecordingDevice(uint16_t index) override;
int32_t SetRecordingDevice(WindowsDeviceType device) override;
// Audio transport initialization
int32_t PlayoutIsAvailable(bool* available) override;
int32_t InitPlayout() override;
bool PlayoutIsInitialized() const override;
int32_t RecordingIsAvailable(bool* available) override;
int32_t InitRecording() override;
bool RecordingIsInitialized() const override;
// Audio transport control
int32_t StartPlayout() override;
int32_t StopPlayout() override;
bool Playing() const override;
int32_t StartRecording() override;
int32_t StopRecording() override;
bool Recording() const override;
// Audio mixer initialization
int32_t InitSpeaker() override;
bool SpeakerIsInitialized() const override;
int32_t InitMicrophone() override;
bool MicrophoneIsInitialized() const override;
// Speaker volume controls
int32_t SpeakerVolumeIsAvailable(bool* available) override;
int32_t SetSpeakerVolume(uint32_t volume) override;
int32_t SpeakerVolume(uint32_t* volume) const override;
int32_t MaxSpeakerVolume(uint32_t* maxVolume) const override;
int32_t MinSpeakerVolume(uint32_t* minVolume) const override;
// Microphone volume controls
int32_t MicrophoneVolumeIsAvailable(bool* available) override;
int32_t SetMicrophoneVolume(uint32_t volume) override;
int32_t MicrophoneVolume(uint32_t* volume) const override;
int32_t MaxMicrophoneVolume(uint32_t* maxVolume) const override;
int32_t MinMicrophoneVolume(uint32_t* minVolume) const override;
// Speaker mute control
int32_t SpeakerMuteIsAvailable(bool* available) override;
int32_t SetSpeakerMute(bool enable) override;
int32_t SpeakerMute(bool* enabled) const override;
// Microphone mute control
int32_t MicrophoneMuteIsAvailable(bool* available) override;
int32_t SetMicrophoneMute(bool enable) override;
int32_t MicrophoneMute(bool* enabled) const override;
// Stereo support
int32_t StereoPlayoutIsAvailable(bool* available) const override;
int32_t SetStereoPlayout(bool enable) override;
int32_t StereoPlayout(bool* enabled) const override;
int32_t StereoRecordingIsAvailable(bool* available) const override;
int32_t SetStereoRecording(bool enable) override;
int32_t StereoRecording(bool* enabled) const override;
// Playout delay
int32_t PlayoutDelay(uint16_t* delayMS) const override;
// Only supported on Android.
bool BuiltInAECIsAvailable() const override;
bool BuiltInAGCIsAvailable() const override;
bool BuiltInNSIsAvailable() const override;
// Enables the built-in audio effects. Only supported on Android.
int32_t EnableBuiltInAEC(bool enable) override;
int32_t EnableBuiltInAGC(bool enable) override;
int32_t EnableBuiltInNS(bool enable) override;
// Play underrun count. Only supported on Android.
int32_t GetPlayoutUnderrunCount() const override;
#if defined(WEBRTC_IOS)
int GetPlayoutAudioParameters(AudioParameters* params) const override;
int GetRecordAudioParameters(AudioParameters* params) const override;
#endif // WEBRTC_IOS
OSStatus OnDeliverRecordedData(AudioUnitRenderActionFlags* flags,
const AudioTimeStamp* time_stamp,
NSInteger bus_number,
UInt32 num_frames,
const AudioBufferList* io_data,
void* render_context,
RTC_OBJC_TYPE(RTCAudioDeviceRenderRecordedDataBlock) render_block);
OSStatus OnGetPlayoutData(AudioUnitRenderActionFlags* flags,
const AudioTimeStamp* time_stamp,
NSInteger bus_number,
UInt32 num_frames,
AudioBufferList* io_data);
// Notifies `ObjCAudioDeviceModule` that at least one of the audio input
// parameters or audio input latency of `RTCAudioDevice` has changed. It necessary to
// update `record_parameters_` with current audio parameter of `RTCAudioDevice`
// via `UpdateAudioParameters` and if parameters are actually change then
// ADB parameters are updated with `UpdateInputAudioDeviceBuffer`. Audio input latency
// stored in `cached_recording_delay_ms_` is also updated with current latency
// of `RTCAudioDevice`.
void HandleAudioInputParametersChange();
// Same as `HandleAudioInputParametersChange` but should be called when audio output
// parameters of `RTCAudioDevice` has changed.
void HandleAudioOutputParametersChange();
// Notifies `ObjCAudioDeviceModule` about audio input interruption happen due to
// any reason so `ObjCAudioDeviceModule` is can prepare to restart of audio IO.
void HandleAudioInputInterrupted();
// Same as `ObjCAudioDeviceModule` but should be called when audio output
// is interrupted.
void HandleAudioOutputInterrupted();
// Update our audio parameters if they are different from current device audio parameters
// Returns true when our parameters are update, false - otherwise.
// `ObjCAudioDeviceModule` has audio device buffer (ADB) which has audio parameters
// of playout & recording. The ADB is configured to work with specific sample rate & channel
// count. `ObjCAudioDeviceModule` stores audio parameters which were used to configure ADB in the
// fields `playout_parameters_` and `recording_parameters_`.
// `RTCAudioDevice` protocol has its own audio parameters exposed as individual properties.
// `RTCAudioDevice` audio parameters might change when playout/recording is already in progress,
// for example, when device is switched. `RTCAudioDevice` audio parameters must be kept in sync
// with ADB audio parameters. This method is invoked when `RTCAudioDevice` reports that it's audio
// parameters (`device_params`) are changed and it detects if there any difference with our
// current audio parameters (`params`). Our parameters are updated in case of actual change and
// method returns true. In case of actual change there is follow-up call to either
// `UpdateOutputAudioDeviceBuffer` or `UpdateInputAudioDeviceBuffer` to apply updated
// `playout_parameters_` or `recording_parameters_` to ADB.
bool UpdateAudioParameters(AudioParameters& params, const AudioParameters& device_params);
// Update our cached audio latency with device latency. Device latency is reported by
// `RTCAudioDevice` object. Whenever latency is changed, `RTCAudioDevice` is obliged to notify ADM
// about the change via `HandleAudioInputParametersChange` or `HandleAudioOutputParametersChange`.
// Current device IO latency is cached in the atomic field and used from audio IO thread
// to be reported to audio device buffer. It is highly recommended by Apple not to call any
// ObjC methods from audio IO thread, that is why implementation relies on caching latency
// into a field and being notified when latency is changed, which is the case when device
// is switched.
void UpdateAudioDelay(std::atomic<int>& delay_ms, const NSTimeInterval device_latency);
// Uses current `playout_parameters_` to inform the audio device buffer (ADB)
// about our internal audio parameters.
void UpdateOutputAudioDeviceBuffer();
// Uses current `record_parameters_` to inform the audio device buffer (ADB)
// about our internal audio parameters.
void UpdateInputAudioDeviceBuffer();
id<RTC_OBJC_TYPE(RTCAudioDevice)> audio_device_;
const std::unique_ptr<TaskQueueFactory> task_queue_factory_;
// AudioDeviceBuffer is a buffer to consume audio recorded by `RTCAudioDevice`
// and provide audio to be played via `RTCAudioDevice`.
// Audio PCMs could have different sample rate and channels count, but expected
// to be in 16-bit integer interleaved linear PCM format.
// The current parameters ADB configured to work with is stored in field
// `playout_parameters_` for playout and `record_parameters_` for recording.
// These parameters and ADB must kept in sync with `RTCAudioDevice` audio parameters.
std::unique_ptr<AudioDeviceBuffer> audio_device_buffer_;
// Set to 1 when recording is active and 0 otherwise.
std::atomic<bool> recording_ = false;
// Set to 1 when playout is active and 0 otherwise.
std::atomic<bool> playing_ = false;
// Stores cached value of `RTCAudioDevice outputLatency` to be used from
// audio IO thread. Latency is updated on audio output parameters change.
std::atomic<int> cached_playout_delay_ms_ = 0;
// Same as `cached_playout_delay_ms_` but for audio input
std::atomic<int> cached_recording_delay_ms_ = 0;
// Thread that is initialized audio device module.
rtc::Thread* thread_;
// Ensures that methods are called from the same thread as this object is
// initialized on.
SequenceChecker thread_checker_;
// I/O audio thread checker.
SequenceChecker io_playout_thread_checker_;
SequenceChecker io_record_thread_checker_;
bool is_initialized_ RTC_GUARDED_BY(thread_checker_) = false;
bool is_playout_initialized_ RTC_GUARDED_BY(thread_checker_) = false;
bool is_recording_initialized_ RTC_GUARDED_BY(thread_checker_) = false;
// Contains audio parameters (sample rate, #channels, buffer size etc.) for
// the playout and recording sides.
AudioParameters playout_parameters_;
AudioParameters record_parameters_;
// `FineAudioBuffer` takes an `AudioDeviceBuffer` which delivers audio data
// in chunks of 10ms. `RTCAudioDevice` might deliver recorded data in
// chunks which are not 10ms long. `FineAudioBuffer` implements adaptation
// from undetermined chunk size to 10ms chunks.
std::unique_ptr<FineAudioBuffer> record_fine_audio_buffer_;
// Same as `record_fine_audio_buffer_` but for audio output.
std::unique_ptr<FineAudioBuffer> playout_fine_audio_buffer_;
// Temporary storage for recorded data.
rtc::BufferT<int16_t> record_audio_buffer_;
// Delegate object provided to RTCAudioDevice during initialization
ObjCAudioDeviceDelegate* audio_device_delegate_;
} // namespace objc_adm
} // namespace webrtc