ff046c74c5
Finally we are able to remove this class entirely, along with the last vestiges of it's use. I've also removed some legacy files that were only used for windows XP support. BUG=webrtc:7035 Review-Url: https://codereview.webrtc.org/2790533002 Cr-Commit-Position: refs/heads/master@{#17480}
386 lines
13 KiB
C++
386 lines
13 KiB
C++
/*
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* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_MAC_H
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#define WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_MAC_H
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#include <memory>
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#include "webrtc/base/criticalsection.h"
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#include "webrtc/base/thread_annotations.h"
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#include "webrtc/modules/audio_device/audio_device_generic.h"
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#include "webrtc/modules/audio_device/mac/audio_mixer_manager_mac.h"
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#include "webrtc/system_wrappers/include/event_wrapper.h"
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#include <AudioToolbox/AudioConverter.h>
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#include <CoreAudio/CoreAudio.h>
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#include <mach/semaphore.h>
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struct PaUtilRingBuffer;
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namespace rtc {
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class PlatformThread;
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} // namespace rtc
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namespace webrtc {
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class EventWrapper;
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const uint32_t N_REC_SAMPLES_PER_SEC = 48000;
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const uint32_t N_PLAY_SAMPLES_PER_SEC = 48000;
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const uint32_t N_REC_CHANNELS = 1; // default is mono recording
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const uint32_t N_PLAY_CHANNELS = 2; // default is stereo playout
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const uint32_t N_DEVICE_CHANNELS = 64;
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const int kBufferSizeMs = 10;
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const uint32_t ENGINE_REC_BUF_SIZE_IN_SAMPLES =
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N_REC_SAMPLES_PER_SEC * kBufferSizeMs / 1000;
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const uint32_t ENGINE_PLAY_BUF_SIZE_IN_SAMPLES =
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N_PLAY_SAMPLES_PER_SEC * kBufferSizeMs / 1000;
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const int N_BLOCKS_IO = 2;
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const int N_BUFFERS_IN = 2; // Must be at least N_BLOCKS_IO.
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const int N_BUFFERS_OUT = 3; // Must be at least N_BLOCKS_IO.
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const uint32_t TIMER_PERIOD_MS = 2 * 10 * N_BLOCKS_IO * 1000000;
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const uint32_t REC_BUF_SIZE_IN_SAMPLES =
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ENGINE_REC_BUF_SIZE_IN_SAMPLES * N_DEVICE_CHANNELS * N_BUFFERS_IN;
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const uint32_t PLAY_BUF_SIZE_IN_SAMPLES =
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ENGINE_PLAY_BUF_SIZE_IN_SAMPLES * N_PLAY_CHANNELS * N_BUFFERS_OUT;
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const int kGetMicVolumeIntervalMs = 1000;
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class AudioDeviceMac : public AudioDeviceGeneric {
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public:
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AudioDeviceMac(const int32_t id);
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~AudioDeviceMac();
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// Retrieve the currently utilized audio layer
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virtual int32_t ActiveAudioLayer(
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AudioDeviceModule::AudioLayer& audioLayer) const;
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// Main initializaton and termination
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virtual InitStatus Init();
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virtual int32_t Terminate();
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virtual bool Initialized() const;
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// Device enumeration
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virtual int16_t PlayoutDevices();
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virtual int16_t RecordingDevices();
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virtual int32_t PlayoutDeviceName(uint16_t index,
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char name[kAdmMaxDeviceNameSize],
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char guid[kAdmMaxGuidSize]);
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virtual int32_t RecordingDeviceName(uint16_t index,
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char name[kAdmMaxDeviceNameSize],
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char guid[kAdmMaxGuidSize]);
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// Device selection
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virtual int32_t SetPlayoutDevice(uint16_t index);
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virtual int32_t SetPlayoutDevice(AudioDeviceModule::WindowsDeviceType device);
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virtual int32_t SetRecordingDevice(uint16_t index);
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virtual int32_t SetRecordingDevice(
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AudioDeviceModule::WindowsDeviceType device);
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// Audio transport initialization
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virtual int32_t PlayoutIsAvailable(bool& available);
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virtual int32_t InitPlayout();
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virtual bool PlayoutIsInitialized() const;
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virtual int32_t RecordingIsAvailable(bool& available);
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virtual int32_t InitRecording();
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virtual bool RecordingIsInitialized() const;
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// Audio transport control
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virtual int32_t StartPlayout();
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virtual int32_t StopPlayout();
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virtual bool Playing() const;
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virtual int32_t StartRecording();
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virtual int32_t StopRecording();
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virtual bool Recording() const;
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// Microphone Automatic Gain Control (AGC)
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virtual int32_t SetAGC(bool enable);
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virtual bool AGC() const;
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// Volume control based on the Windows Wave API (Windows only)
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virtual int32_t SetWaveOutVolume(uint16_t volumeLeft, uint16_t volumeRight);
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virtual int32_t WaveOutVolume(uint16_t& volumeLeft,
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uint16_t& volumeRight) const;
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// Audio mixer initialization
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virtual int32_t InitSpeaker();
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virtual bool SpeakerIsInitialized() const;
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virtual int32_t InitMicrophone();
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virtual bool MicrophoneIsInitialized() const;
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// Speaker volume controls
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virtual int32_t SpeakerVolumeIsAvailable(bool& available);
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virtual int32_t SetSpeakerVolume(uint32_t volume);
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virtual int32_t SpeakerVolume(uint32_t& volume) const;
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virtual int32_t MaxSpeakerVolume(uint32_t& maxVolume) const;
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virtual int32_t MinSpeakerVolume(uint32_t& minVolume) const;
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virtual int32_t SpeakerVolumeStepSize(uint16_t& stepSize) const;
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// Microphone volume controls
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virtual int32_t MicrophoneVolumeIsAvailable(bool& available);
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virtual int32_t SetMicrophoneVolume(uint32_t volume);
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virtual int32_t MicrophoneVolume(uint32_t& volume) const;
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virtual int32_t MaxMicrophoneVolume(uint32_t& maxVolume) const;
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virtual int32_t MinMicrophoneVolume(uint32_t& minVolume) const;
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virtual int32_t MicrophoneVolumeStepSize(uint16_t& stepSize) const;
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// Microphone mute control
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virtual int32_t MicrophoneMuteIsAvailable(bool& available);
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virtual int32_t SetMicrophoneMute(bool enable);
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virtual int32_t MicrophoneMute(bool& enabled) const;
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// Speaker mute control
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virtual int32_t SpeakerMuteIsAvailable(bool& available);
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virtual int32_t SetSpeakerMute(bool enable);
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virtual int32_t SpeakerMute(bool& enabled) const;
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// Microphone boost control
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virtual int32_t MicrophoneBoostIsAvailable(bool& available);
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virtual int32_t SetMicrophoneBoost(bool enable);
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virtual int32_t MicrophoneBoost(bool& enabled) const;
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// Stereo support
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virtual int32_t StereoPlayoutIsAvailable(bool& available);
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virtual int32_t SetStereoPlayout(bool enable);
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virtual int32_t StereoPlayout(bool& enabled) const;
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virtual int32_t StereoRecordingIsAvailable(bool& available);
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virtual int32_t SetStereoRecording(bool enable);
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virtual int32_t StereoRecording(bool& enabled) const;
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// Delay information and control
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virtual int32_t SetPlayoutBuffer(const AudioDeviceModule::BufferType type,
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uint16_t sizeMS);
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virtual int32_t PlayoutBuffer(AudioDeviceModule::BufferType& type,
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uint16_t& sizeMS) const;
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virtual int32_t PlayoutDelay(uint16_t& delayMS) const;
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virtual int32_t RecordingDelay(uint16_t& delayMS) const;
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// CPU load
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virtual int32_t CPULoad(uint16_t& load) const;
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virtual bool PlayoutWarning() const;
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virtual bool PlayoutError() const;
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virtual bool RecordingWarning() const;
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virtual bool RecordingError() const;
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virtual void ClearPlayoutWarning();
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virtual void ClearPlayoutError();
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virtual void ClearRecordingWarning();
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virtual void ClearRecordingError();
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virtual void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer);
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private:
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virtual int32_t MicrophoneIsAvailable(bool& available);
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virtual int32_t SpeakerIsAvailable(bool& available);
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static void AtomicSet32(int32_t* theValue, int32_t newValue);
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static int32_t AtomicGet32(int32_t* theValue);
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static void logCAMsg(const TraceLevel level,
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const TraceModule module,
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const int32_t id,
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const char* msg,
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const char* err);
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int32_t GetNumberDevices(const AudioObjectPropertyScope scope,
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AudioDeviceID scopedDeviceIds[],
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const uint32_t deviceListLength);
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int32_t GetDeviceName(const AudioObjectPropertyScope scope,
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const uint16_t index,
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char* name);
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int32_t InitDevice(uint16_t userDeviceIndex,
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AudioDeviceID& deviceId,
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bool isInput);
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// Always work with our preferred playout format inside VoE.
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// Then convert the output to the OS setting using an AudioConverter.
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OSStatus SetDesiredPlayoutFormat();
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static OSStatus objectListenerProc(
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AudioObjectID objectId,
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UInt32 numberAddresses,
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const AudioObjectPropertyAddress addresses[],
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void* clientData);
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OSStatus implObjectListenerProc(AudioObjectID objectId,
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UInt32 numberAddresses,
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const AudioObjectPropertyAddress addresses[]);
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int32_t HandleDeviceChange();
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int32_t HandleStreamFormatChange(AudioObjectID objectId,
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AudioObjectPropertyAddress propertyAddress);
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int32_t HandleDataSourceChange(AudioObjectID objectId,
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AudioObjectPropertyAddress propertyAddress);
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int32_t HandleProcessorOverload(AudioObjectPropertyAddress propertyAddress);
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static OSStatus deviceIOProc(AudioDeviceID device,
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const AudioTimeStamp* now,
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const AudioBufferList* inputData,
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const AudioTimeStamp* inputTime,
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AudioBufferList* outputData,
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const AudioTimeStamp* outputTime,
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void* clientData);
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static OSStatus outConverterProc(
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AudioConverterRef audioConverter,
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UInt32* numberDataPackets,
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AudioBufferList* data,
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AudioStreamPacketDescription** dataPacketDescription,
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void* userData);
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static OSStatus inDeviceIOProc(AudioDeviceID device,
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const AudioTimeStamp* now,
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const AudioBufferList* inputData,
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const AudioTimeStamp* inputTime,
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AudioBufferList* outputData,
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const AudioTimeStamp* outputTime,
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void* clientData);
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static OSStatus inConverterProc(
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AudioConverterRef audioConverter,
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UInt32* numberDataPackets,
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AudioBufferList* data,
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AudioStreamPacketDescription** dataPacketDescription,
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void* inUserData);
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OSStatus implDeviceIOProc(const AudioBufferList* inputData,
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const AudioTimeStamp* inputTime,
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AudioBufferList* outputData,
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const AudioTimeStamp* outputTime);
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OSStatus implOutConverterProc(UInt32* numberDataPackets,
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AudioBufferList* data);
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OSStatus implInDeviceIOProc(const AudioBufferList* inputData,
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const AudioTimeStamp* inputTime);
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OSStatus implInConverterProc(UInt32* numberDataPackets,
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AudioBufferList* data);
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static bool RunCapture(void*);
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static bool RunRender(void*);
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bool CaptureWorkerThread();
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bool RenderWorkerThread();
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bool KeyPressed();
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AudioDeviceBuffer* _ptrAudioBuffer;
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rtc::CriticalSection _critSect;
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EventWrapper& _stopEventRec;
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EventWrapper& _stopEvent;
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// TODO(pbos): Replace with direct members, just start/stop, no need to
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// recreate the thread.
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// Only valid/running between calls to StartRecording and StopRecording.
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std::unique_ptr<rtc::PlatformThread> capture_worker_thread_;
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// Only valid/running between calls to StartPlayout and StopPlayout.
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std::unique_ptr<rtc::PlatformThread> render_worker_thread_;
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int32_t _id;
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AudioMixerManagerMac _mixerManager;
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uint16_t _inputDeviceIndex;
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uint16_t _outputDeviceIndex;
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AudioDeviceID _inputDeviceID;
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AudioDeviceID _outputDeviceID;
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#if __MAC_OS_X_VERSION_MAX_ALLOWED >= 1050
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AudioDeviceIOProcID _inDeviceIOProcID;
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AudioDeviceIOProcID _deviceIOProcID;
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#endif
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bool _inputDeviceIsSpecified;
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bool _outputDeviceIsSpecified;
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uint8_t _recChannels;
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uint8_t _playChannels;
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Float32* _captureBufData;
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SInt16* _renderBufData;
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SInt16 _renderConvertData[PLAY_BUF_SIZE_IN_SAMPLES];
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AudioDeviceModule::BufferType _playBufType;
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bool _initialized;
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bool _isShutDown;
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bool _recording;
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bool _playing;
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bool _recIsInitialized;
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bool _playIsInitialized;
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bool _AGC;
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// Atomically set varaibles
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int32_t _renderDeviceIsAlive;
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int32_t _captureDeviceIsAlive;
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bool _twoDevices;
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bool _doStop; // For play if not shared device or play+rec if shared device
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bool _doStopRec; // For rec if not shared device
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bool _macBookPro;
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bool _macBookProPanRight;
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AudioConverterRef _captureConverter;
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AudioConverterRef _renderConverter;
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AudioStreamBasicDescription _outStreamFormat;
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AudioStreamBasicDescription _outDesiredFormat;
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AudioStreamBasicDescription _inStreamFormat;
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AudioStreamBasicDescription _inDesiredFormat;
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uint32_t _captureLatencyUs;
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uint32_t _renderLatencyUs;
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// Atomically set variables
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mutable int32_t _captureDelayUs;
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mutable int32_t _renderDelayUs;
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int32_t _renderDelayOffsetSamples;
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uint16_t _playBufDelayFixed; // fixed playback delay
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uint16_t _playWarning;
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uint16_t _playError;
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uint16_t _recWarning;
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uint16_t _recError;
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PaUtilRingBuffer* _paCaptureBuffer;
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PaUtilRingBuffer* _paRenderBuffer;
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semaphore_t _renderSemaphore;
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semaphore_t _captureSemaphore;
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int _captureBufSizeSamples;
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int _renderBufSizeSamples;
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// Typing detection
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// 0x5c is key "9", after that comes function keys.
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bool prev_key_state_[0x5d];
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int get_mic_volume_counter_ms_;
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};
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} // namespace webrtc
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#endif // MODULES_AUDIO_DEVICE_MAIN_SOURCE_MAC_AUDIO_DEVICE_MAC_H_
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