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Diffstat (limited to 'third_party/libwebrtc/sdk/objc/native/src/audio/audio_device_ios.h')
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1 files changed, 306 insertions, 0 deletions
diff --git a/third_party/libwebrtc/sdk/objc/native/src/audio/audio_device_ios.h b/third_party/libwebrtc/sdk/objc/native/src/audio/audio_device_ios.h new file mode 100644 index 0000000000..a86acb56fe --- /dev/null +++ b/third_party/libwebrtc/sdk/objc/native/src/audio/audio_device_ios.h @@ -0,0 +1,306 @@ +/* + * 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 SDK_OBJC_NATIVE_SRC_AUDIO_AUDIO_DEVICE_IOS_H_ +#define SDK_OBJC_NATIVE_SRC_AUDIO_AUDIO_DEVICE_IOS_H_ + +#include <atomic> +#include <memory> + +#include "api/scoped_refptr.h" +#include "api/sequence_checker.h" +#include "api/task_queue/pending_task_safety_flag.h" +#include "audio_session_observer.h" +#include "modules/audio_device/audio_device_generic.h" +#include "rtc_base/buffer.h" +#include "rtc_base/thread.h" +#include "rtc_base/thread_annotations.h" +#include "sdk/objc/base/RTCMacros.h" +#include "voice_processing_audio_unit.h" + +RTC_FWD_DECL_OBJC_CLASS(RTCNativeAudioSessionDelegateAdapter); + +namespace webrtc { + +class FineAudioBuffer; + +namespace ios_adm { + +// Implements full duplex 16-bit mono PCM audio support for iOS using a +// Voice-Processing (VP) I/O audio unit in Core Audio. The VP I/O audio unit +// supports audio echo cancellation. It also adds automatic gain control, +// adjustment of voice-processing quality and muting. +// +// An instance must be created and destroyed on one and the same thread. +// All supported public methods must also be called on the same thread. +// A thread checker will RTC_DCHECK if any supported method is called on an +// invalid thread. +// +// Recorded audio will be delivered on a real-time internal I/O thread in the +// audio unit. The audio unit will also ask for audio data to play out on this +// same thread. +class AudioDeviceIOS : public AudioDeviceGeneric, + public AudioSessionObserver, + public VoiceProcessingAudioUnitObserver { + public: + explicit AudioDeviceIOS(bool bypass_voice_processing); + ~AudioDeviceIOS() override; + + void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) override; + + InitStatus Init() override; + int32_t Terminate() override; + bool Initialized() const override; + + int32_t InitPlayout() override; + bool PlayoutIsInitialized() const override; + + int32_t InitRecording() override; + bool RecordingIsInitialized() const override; + + int32_t StartPlayout() override; + int32_t StopPlayout() override; + bool Playing() const override; + + int32_t StartRecording() override; + int32_t StopRecording() override; + bool Recording() const override; + + // These methods returns hard-coded delay values and not dynamic delay + // estimates. The reason is that iOS supports a built-in AEC and the WebRTC + // AEC will always be disabled in the Libjingle layer to avoid running two + // AEC implementations at the same time. And, it saves resources to avoid + // updating these delay values continuously. + // TODO(henrika): it would be possible to mark these two methods as not + // implemented since they are only called for A/V-sync purposes today and + // A/V-sync is not supported on iOS. However, we avoid adding error messages + // the log by using these dummy implementations instead. + int32_t PlayoutDelay(uint16_t& delayMS) const override; + + // No implementation for playout underrun on iOS. We override it to avoid a + // periodic log that it isn't available from the base class. + int32_t GetPlayoutUnderrunCount() const override { return -1; } + + // Native audio parameters stored during construction. + // These methods are unique for the iOS implementation. + int GetPlayoutAudioParameters(AudioParameters* params) const override; + int GetRecordAudioParameters(AudioParameters* params) const override; + + // These methods are currently not fully implemented on iOS: + + // See audio_device_not_implemented.cc for trivial implementations. + int32_t ActiveAudioLayer( + AudioDeviceModule::AudioLayer& audioLayer) const override; + int32_t PlayoutIsAvailable(bool& available) override; + int32_t RecordingIsAvailable(bool& available) override; + 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; + int32_t SetPlayoutDevice(uint16_t index) override; + int32_t SetPlayoutDevice( + AudioDeviceModule::WindowsDeviceType device) override; + int32_t SetRecordingDevice(uint16_t index) override; + int32_t SetRecordingDevice( + AudioDeviceModule::WindowsDeviceType device) override; + int32_t InitSpeaker() override; + bool SpeakerIsInitialized() const override; + int32_t InitMicrophone() override; + bool MicrophoneIsInitialized() const override; + 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; + 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; + int32_t MicrophoneMuteIsAvailable(bool& available) override; + int32_t SetMicrophoneMute(bool enable) override; + int32_t MicrophoneMute(bool& enabled) const override; + int32_t SpeakerMuteIsAvailable(bool& available) override; + int32_t SetSpeakerMute(bool enable) override; + int32_t SpeakerMute(bool& enabled) const override; + int32_t StereoPlayoutIsAvailable(bool& available) override; + int32_t SetStereoPlayout(bool enable) override; + int32_t StereoPlayout(bool& enabled) const override; + int32_t StereoRecordingIsAvailable(bool& available) override; + int32_t SetStereoRecording(bool enable) override; + int32_t StereoRecording(bool& enabled) const override; + + // AudioSessionObserver methods. May be called from any thread. + void OnInterruptionBegin() override; + void OnInterruptionEnd() override; + void OnValidRouteChange() override; + void OnCanPlayOrRecordChange(bool can_play_or_record) override; + void OnChangedOutputVolume() override; + + // VoiceProcessingAudioUnitObserver methods. + OSStatus OnDeliverRecordedData(AudioUnitRenderActionFlags* flags, + const AudioTimeStamp* time_stamp, + UInt32 bus_number, + UInt32 num_frames, + AudioBufferList* io_data) override; + OSStatus OnGetPlayoutData(AudioUnitRenderActionFlags* flags, + const AudioTimeStamp* time_stamp, + UInt32 bus_number, + UInt32 num_frames, + AudioBufferList* io_data) override; + + bool IsInterrupted(); + + private: + // Called by the relevant AudioSessionObserver methods on `thread_`. + void HandleInterruptionBegin(); + void HandleInterruptionEnd(); + void HandleValidRouteChange(); + void HandleCanPlayOrRecordChange(bool can_play_or_record); + void HandleSampleRateChange(); + void HandlePlayoutGlitchDetected(); + void HandleOutputVolumeChange(); + + // Uses current `playout_parameters_` and `record_parameters_` to inform the + // audio device buffer (ADB) about our internal audio parameters. + void UpdateAudioDeviceBuffer(); + + // Since the preferred audio parameters are only hints to the OS, the actual + // values may be different once the AVAudioSession has been activated. + // This method asks for the current hardware parameters and takes actions + // if they should differ from what we have asked for initially. It also + // defines `playout_parameters_` and `record_parameters_`. + void SetupAudioBuffersForActiveAudioSession(); + + // Creates the audio unit. + bool CreateAudioUnit(); + + // Updates the audio unit state based on current state. + void UpdateAudioUnit(bool can_play_or_record); + + // Configures the audio session for WebRTC. + bool ConfigureAudioSession(); + + // Like above, but requires caller to already hold session lock. + bool ConfigureAudioSessionLocked(); + + // Unconfigures the audio session. + void UnconfigureAudioSession(); + + // Activates our audio session, creates and initializes the voice-processing + // audio unit and verifies that we got the preferred native audio parameters. + bool InitPlayOrRecord(); + + // Closes and deletes the voice-processing I/O unit. + void ShutdownPlayOrRecord(); + + // Resets thread-checkers before a call is restarted. + void PrepareForNewStart(); + + // Determines whether voice processing should be enabled or disabled. + const bool bypass_voice_processing_; + + // Native I/O audio thread checker. + SequenceChecker io_thread_checker_; + + // Thread that this object is created on. + rtc::Thread* thread_; + + // Raw pointer handle provided to us in AttachAudioBuffer(). Owned by the + // AudioDeviceModuleImpl class and called by AudioDeviceModule::Create(). + // The AudioDeviceBuffer is a member of the AudioDeviceModuleImpl instance + // and therefore outlives this object. + AudioDeviceBuffer* audio_device_buffer_; + + // Contains audio parameters (sample rate, #channels, buffer size etc.) for + // the playout and recording sides. These structure is set in two steps: + // first, native sample rate and #channels are defined in Init(). Next, the + // audio session is activated and we verify that the preferred parameters + // were granted by the OS. At this stage it is also possible to add a third + // component to the parameters; the native I/O buffer duration. + // A RTC_CHECK will be hit if we for some reason fail to open an audio session + // using the specified parameters. + AudioParameters playout_parameters_; + AudioParameters record_parameters_; + + // The AudioUnit used to play and record audio. + std::unique_ptr<VoiceProcessingAudioUnit> audio_unit_; + + // FineAudioBuffer takes an AudioDeviceBuffer which delivers audio data + // in chunks of 10ms. It then allows for this data to be pulled in + // a finer or coarser granularity. I.e. interacting with this class instead + // of directly with the AudioDeviceBuffer one can ask for any number of + // audio data samples. Is also supports a similar scheme for the recording + // side. + // Example: native buffer size can be 128 audio frames at 16kHz sample rate. + // WebRTC will provide 480 audio frames per 10ms but iOS asks for 128 + // in each callback (one every 8ms). This class can then ask for 128 and the + // FineAudioBuffer will ask WebRTC for new data only when needed and also + // cache non-utilized audio between callbacks. On the recording side, iOS + // can provide audio data frames of size 128 and these are accumulated until + // enough data to supply one 10ms call exists. This 10ms chunk is then sent + // to WebRTC and the remaining part is stored. + std::unique_ptr<FineAudioBuffer> fine_audio_buffer_; + + // Temporary storage for recorded data. AudioUnitRender() renders into this + // array as soon as a frame of the desired buffer size has been recorded. + // On real iOS devices, the size will be fixed and set once. For iOS + // simulators, the size can vary from callback to callback and the size + // will be changed dynamically to account for this behavior. + rtc::BufferT<int16_t> record_audio_buffer_; + + // Set to 1 when recording is active and 0 otherwise. + std::atomic<int> recording_; + + // Set to 1 when playout is active and 0 otherwise. + std::atomic<int> playing_; + + // Set to true after successful call to Init(), false otherwise. + bool initialized_ RTC_GUARDED_BY(thread_); + + // Set to true after successful call to InitRecording() or InitPlayout(), + // false otherwise. + bool audio_is_initialized_; + + // Set to true if audio session is interrupted, false otherwise. + bool is_interrupted_; + + // Audio interruption observer instance. + RTCNativeAudioSessionDelegateAdapter* audio_session_observer_ + RTC_GUARDED_BY(thread_); + + // Set to true if we've activated the audio session. + bool has_configured_session_ RTC_GUARDED_BY(thread_); + + // Counts number of detected audio glitches on the playout side. + int64_t num_detected_playout_glitches_ RTC_GUARDED_BY(thread_); + int64_t last_playout_time_ RTC_GUARDED_BY(io_thread_checker_); + + // Counts number of playout callbacks per call. + // The value is updated on the native I/O thread and later read on the + // creating `thread_` but at this stage no audio is active. + // Hence, it is a "thread safe" design and no lock is needed. + int64_t num_playout_callbacks_; + + // Contains the time for when the last output volume change was detected. + int64_t last_output_volume_change_time_ RTC_GUARDED_BY(thread_); + + // Avoids running pending task after `this` is Terminated. + rtc::scoped_refptr<PendingTaskSafetyFlag> safety_ = + PendingTaskSafetyFlag::Create(); +}; +} // namespace ios_adm +} // namespace webrtc + +#endif // SDK_OBJC_NATIVE_SRC_AUDIO_AUDIO_DEVICE_IOS_H_ |