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platform-external-webrtc/webrtc/modules/audio_device/mac/audio_device_mac.cc
braveyao@webrtc.org 346a64b9b5 Mac would force bluetooth playout working with 8kHz/1ch if capturing/rendering shares the same device, e.g. changing from 44.1kHz/2ch as default. 
So in the HandleStreamFormatChange() callback, we need to re-initiate the playout as same as what we do in InitPlayout(). Here we merely copy those codes out from InitPlayout() into a new SetDesiredPlayoutFormat() function for the invoking from the two places.
Previously, HandleStreamFormatChange only re-creates the AudioConverter, which is not enough. We also need to reset the buffer size and refresh the latency.

BUG=4240
TEST=Manual Test
R=andrew@webrtc.org, henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36029004

Cr-Commit-Position: refs/heads/master@{#8815}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8815 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-21 01:06:14 +00:00

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/*
* 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.
*/
#include "webrtc/base/arraysize.h"
#include "webrtc/base/checks.h"
#include "webrtc/modules/audio_device/audio_device_config.h"
#include "webrtc/modules/audio_device/audio_device_utility.h"
#include "webrtc/modules/audio_device/mac/audio_device_mac.h"
#include "webrtc/modules/audio_device/mac/portaudio/pa_ringbuffer.h"
#include "webrtc/system_wrappers/interface/event_wrapper.h"
#include "webrtc/system_wrappers/interface/thread_wrapper.h"
#include "webrtc/system_wrappers/interface/trace.h"
#include <ApplicationServices/ApplicationServices.h>
#include <libkern/OSAtomic.h> // OSAtomicCompareAndSwap()
#include <mach/mach.h> // mach_task_self()
#include <sys/sysctl.h> // sysctlbyname()
namespace webrtc
{
#define WEBRTC_CA_RETURN_ON_ERR(expr) \
do { \
err = expr; \
if (err != noErr) { \
logCAMsg(kTraceError, kTraceAudioDevice, _id, \
"Error in " #expr, (const char *)&err); \
return -1; \
} \
} while(0)
#define WEBRTC_CA_LOG_ERR(expr) \
do { \
err = expr; \
if (err != noErr) { \
logCAMsg(kTraceError, kTraceAudioDevice, _id, \
"Error in " #expr, (const char *)&err); \
} \
} while(0)
#define WEBRTC_CA_LOG_WARN(expr) \
do { \
err = expr; \
if (err != noErr) { \
logCAMsg(kTraceWarning, kTraceAudioDevice, _id, \
"Error in " #expr, (const char *)&err); \
} \
} while(0)
enum
{
MaxNumberDevices = 64
};
void AudioDeviceMac::AtomicSet32(int32_t* theValue, int32_t newValue)
{
while (1)
{
int32_t oldValue = *theValue;
if (OSAtomicCompareAndSwap32Barrier(oldValue, newValue, theValue)
== true)
{
return;
}
}
}
int32_t AudioDeviceMac::AtomicGet32(int32_t* theValue)
{
while (1)
{
int32_t value = *theValue;
if (OSAtomicCompareAndSwap32Barrier(value, value, theValue) == true)
{
return value;
}
}
}
// CoreAudio errors are best interpreted as four character strings.
void AudioDeviceMac::logCAMsg(const TraceLevel level,
const TraceModule module,
const int32_t id, const char *msg,
const char *err)
{
DCHECK(msg != NULL);
DCHECK(err != NULL);
#ifdef WEBRTC_ARCH_BIG_ENDIAN
WEBRTC_TRACE(level, module, id, "%s: %.4s", msg, err);
#else
// We need to flip the characters in this case.
WEBRTC_TRACE(level, module, id, "%s: %.1s%.1s%.1s%.1s", msg, err + 3, err
+ 2, err + 1, err);
#endif
}
AudioDeviceMac::AudioDeviceMac(const int32_t id) :
_ptrAudioBuffer(NULL),
_critSect(*CriticalSectionWrapper::CreateCriticalSection()),
_stopEventRec(*EventWrapper::Create()),
_stopEvent(*EventWrapper::Create()),
_id(id),
_mixerManager(id),
_inputDeviceIndex(0),
_outputDeviceIndex(0),
_inputDeviceID(kAudioObjectUnknown),
_outputDeviceID(kAudioObjectUnknown),
_inputDeviceIsSpecified(false),
_outputDeviceIsSpecified(false),
_recChannels(N_REC_CHANNELS),
_playChannels(N_PLAY_CHANNELS),
_captureBufData(NULL),
_renderBufData(NULL),
_playBufType(AudioDeviceModule::kFixedBufferSize),
_initialized(false),
_isShutDown(false),
_recording(false),
_playing(false),
_recIsInitialized(false),
_playIsInitialized(false),
_AGC(false),
_renderDeviceIsAlive(1),
_captureDeviceIsAlive(1),
_twoDevices(true),
_doStop(false),
_doStopRec(false),
_macBookPro(false),
_macBookProPanRight(false),
_captureLatencyUs(0),
_renderLatencyUs(0),
_captureDelayUs(0),
_renderDelayUs(0),
_renderDelayOffsetSamples(0),
_playBufDelayFixed(20),
_playWarning(0),
_playError(0),
_recWarning(0),
_recError(0),
_paCaptureBuffer(NULL),
_paRenderBuffer(NULL),
_captureBufSizeSamples(0),
_renderBufSizeSamples(0),
prev_key_state_()
{
WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, id,
"%s created", __FUNCTION__);
DCHECK(&_stopEvent != NULL);
DCHECK(&_stopEventRec != NULL);
memset(_renderConvertData, 0, sizeof(_renderConvertData));
memset(&_outStreamFormat, 0, sizeof(AudioStreamBasicDescription));
memset(&_outDesiredFormat, 0, sizeof(AudioStreamBasicDescription));
memset(&_inStreamFormat, 0, sizeof(AudioStreamBasicDescription));
memset(&_inDesiredFormat, 0, sizeof(AudioStreamBasicDescription));
}
AudioDeviceMac::~AudioDeviceMac()
{
WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id,
"%s destroyed", __FUNCTION__);
if (!_isShutDown)
{
Terminate();
}
DCHECK(!capture_worker_thread_.get());
DCHECK(!render_worker_thread_.get());
if (_paRenderBuffer)
{
delete _paRenderBuffer;
_paRenderBuffer = NULL;
}
if (_paCaptureBuffer)
{
delete _paCaptureBuffer;
_paCaptureBuffer = NULL;
}
if (_renderBufData)
{
delete[] _renderBufData;
_renderBufData = NULL;
}
if (_captureBufData)
{
delete[] _captureBufData;
_captureBufData = NULL;
}
kern_return_t kernErr = KERN_SUCCESS;
kernErr = semaphore_destroy(mach_task_self(), _renderSemaphore);
if (kernErr != KERN_SUCCESS)
{
WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
" semaphore_destroy() error: %d", kernErr);
}
kernErr = semaphore_destroy(mach_task_self(), _captureSemaphore);
if (kernErr != KERN_SUCCESS)
{
WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
" semaphore_destroy() error: %d", kernErr);
}
delete &_stopEvent;
delete &_stopEventRec;
delete &_critSect;
}
// ============================================================================
// API
// ============================================================================
void AudioDeviceMac::AttachAudioBuffer(AudioDeviceBuffer* audioBuffer)
{
CriticalSectionScoped lock(&_critSect);
_ptrAudioBuffer = audioBuffer;
// inform the AudioBuffer about default settings for this implementation
_ptrAudioBuffer->SetRecordingSampleRate(N_REC_SAMPLES_PER_SEC);
_ptrAudioBuffer->SetPlayoutSampleRate(N_PLAY_SAMPLES_PER_SEC);
_ptrAudioBuffer->SetRecordingChannels(N_REC_CHANNELS);
_ptrAudioBuffer->SetPlayoutChannels(N_PLAY_CHANNELS);
}
int32_t AudioDeviceMac::ActiveAudioLayer(
AudioDeviceModule::AudioLayer& audioLayer) const
{
audioLayer = AudioDeviceModule::kPlatformDefaultAudio;
return 0;
}
int32_t AudioDeviceMac::Init()
{
CriticalSectionScoped lock(&_critSect);
if (_initialized)
{
return 0;
}
OSStatus err = noErr;
_isShutDown = false;
// PortAudio ring buffers require an elementCount which is a power of two.
if (_renderBufData == NULL)
{
UInt32 powerOfTwo = 1;
while (powerOfTwo < PLAY_BUF_SIZE_IN_SAMPLES)
{
powerOfTwo <<= 1;
}
_renderBufSizeSamples = powerOfTwo;
_renderBufData = new SInt16[_renderBufSizeSamples];
}
if (_paRenderBuffer == NULL)
{
_paRenderBuffer = new PaUtilRingBuffer;
PaRingBufferSize bufSize = -1;
bufSize = PaUtil_InitializeRingBuffer(_paRenderBuffer, sizeof(SInt16),
_renderBufSizeSamples,
_renderBufData);
if (bufSize == -1)
{
WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice,
_id, " PaUtil_InitializeRingBuffer() error");
return -1;
}
}
if (_captureBufData == NULL)
{
UInt32 powerOfTwo = 1;
while (powerOfTwo < REC_BUF_SIZE_IN_SAMPLES)
{
powerOfTwo <<= 1;
}
_captureBufSizeSamples = powerOfTwo;
_captureBufData = new Float32[_captureBufSizeSamples];
}
if (_paCaptureBuffer == NULL)
{
_paCaptureBuffer = new PaUtilRingBuffer;
PaRingBufferSize bufSize = -1;
bufSize = PaUtil_InitializeRingBuffer(_paCaptureBuffer,
sizeof(Float32),
_captureBufSizeSamples,
_captureBufData);
if (bufSize == -1)
{
WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice,
_id, " PaUtil_InitializeRingBuffer() error");
return -1;
}
}
kern_return_t kernErr = KERN_SUCCESS;
kernErr = semaphore_create(mach_task_self(), &_renderSemaphore,
SYNC_POLICY_FIFO, 0);
if (kernErr != KERN_SUCCESS)
{
WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice, _id,
" semaphore_create() error: %d", kernErr);
return -1;
}
kernErr = semaphore_create(mach_task_self(), &_captureSemaphore,
SYNC_POLICY_FIFO, 0);
if (kernErr != KERN_SUCCESS)
{
WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice, _id,
" semaphore_create() error: %d", kernErr);
return -1;
}
// Setting RunLoop to NULL here instructs HAL to manage its own thread for
// notifications. This was the default behaviour on OS X 10.5 and earlier,
// but now must be explicitly specified. HAL would otherwise try to use the
// main thread to issue notifications.
AudioObjectPropertyAddress propertyAddress = {
kAudioHardwarePropertyRunLoop,
kAudioObjectPropertyScopeGlobal,
kAudioObjectPropertyElementMaster };
CFRunLoopRef runLoop = NULL;
UInt32 size = sizeof(CFRunLoopRef);
WEBRTC_CA_RETURN_ON_ERR(AudioObjectSetPropertyData(kAudioObjectSystemObject,
&propertyAddress, 0, NULL, size, &runLoop));
// Listen for any device changes.
propertyAddress.mSelector = kAudioHardwarePropertyDevices;
WEBRTC_CA_LOG_ERR(AudioObjectAddPropertyListener(kAudioObjectSystemObject,
&propertyAddress, &objectListenerProc, this));
// Determine if this is a MacBook Pro
_macBookPro = false;
_macBookProPanRight = false;
char buf[128];
size_t length = sizeof(buf);
memset(buf, 0, length);
int intErr = sysctlbyname("hw.model", buf, &length, NULL, 0);
if (intErr != 0)
{
WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
" Error in sysctlbyname(): %d", err);
} else
{
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
" Hardware model: %s", buf);
if (strncmp(buf, "MacBookPro", 10) == 0)
{
_macBookPro = true;
}
}
_playWarning = 0;
_playError = 0;
_recWarning = 0;
_recError = 0;
_initialized = true;
return 0;
}
int32_t AudioDeviceMac::Terminate()
{
if (!_initialized)
{
return 0;
}
if (_recording)
{
WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
" Recording must be stopped");
return -1;
}
if (_playing)
{
WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
" Playback must be stopped");
return -1;
}
_critSect.Enter();
_mixerManager.Close();
OSStatus err = noErr;
int retVal = 0;
AudioObjectPropertyAddress propertyAddress = {
kAudioHardwarePropertyDevices, kAudioObjectPropertyScopeGlobal,
kAudioObjectPropertyElementMaster };
WEBRTC_CA_LOG_WARN(AudioObjectRemovePropertyListener(kAudioObjectSystemObject,
&propertyAddress, &objectListenerProc, this));
err = AudioHardwareUnload();
if (err != noErr)
{
logCAMsg(kTraceError, kTraceAudioDevice, _id,
"Error in AudioHardwareUnload()", (const char*) &err);
retVal = -1;
}
_isShutDown = true;
_initialized = false;
_outputDeviceIsSpecified = false;
_inputDeviceIsSpecified = false;
_critSect.Leave();
return retVal;
}
bool AudioDeviceMac::Initialized() const
{
return (_initialized);
}
int32_t AudioDeviceMac::SpeakerIsAvailable(bool& available)
{
bool wasInitialized = _mixerManager.SpeakerIsInitialized();
// Make an attempt to open up the
// output mixer corresponding to the currently selected output device.
//
if (!wasInitialized && InitSpeaker() == -1)
{
available = false;
return 0;
}
// Given that InitSpeaker was successful, we know that a valid speaker
// exists.
available = true;
// Close the initialized output mixer
//
if (!wasInitialized)
{
_mixerManager.CloseSpeaker();
}
return 0;
}
int32_t AudioDeviceMac::InitSpeaker()
{
CriticalSectionScoped lock(&_critSect);
if (_playing)
{
return -1;
}
if (InitDevice(_outputDeviceIndex, _outputDeviceID, false) == -1)
{
return -1;
}
if (_inputDeviceID == _outputDeviceID)
{
_twoDevices = false;
} else
{
_twoDevices = true;
}
if (_mixerManager.OpenSpeaker(_outputDeviceID) == -1)
{
return -1;
}
return 0;
}
int32_t AudioDeviceMac::MicrophoneIsAvailable(bool& available)
{
bool wasInitialized = _mixerManager.MicrophoneIsInitialized();
// Make an attempt to open up the
// input mixer corresponding to the currently selected output device.
//
if (!wasInitialized && InitMicrophone() == -1)
{
available = false;
return 0;
}
// Given that InitMicrophone was successful, we know that a valid microphone
// exists.
available = true;
// Close the initialized input mixer
//
if (!wasInitialized)
{
_mixerManager.CloseMicrophone();
}
return 0;
}
int32_t AudioDeviceMac::InitMicrophone()
{
CriticalSectionScoped lock(&_critSect);
if (_recording)
{
return -1;
}
if (InitDevice(_inputDeviceIndex, _inputDeviceID, true) == -1)
{
return -1;
}
if (_inputDeviceID == _outputDeviceID)
{
_twoDevices = false;
} else
{
_twoDevices = true;
}
if (_mixerManager.OpenMicrophone(_inputDeviceID) == -1)
{
return -1;
}
return 0;
}
bool AudioDeviceMac::SpeakerIsInitialized() const
{
return (_mixerManager.SpeakerIsInitialized());
}
bool AudioDeviceMac::MicrophoneIsInitialized() const
{
return (_mixerManager.MicrophoneIsInitialized());
}
int32_t AudioDeviceMac::SpeakerVolumeIsAvailable(bool& available)
{
bool wasInitialized = _mixerManager.SpeakerIsInitialized();
// Make an attempt to open up the
// output mixer corresponding to the currently selected output device.
//
if (!wasInitialized && InitSpeaker() == -1)
{
// If we end up here it means that the selected speaker has no volume
// control.
available = false;
return 0;
}
// Given that InitSpeaker was successful, we know that a volume control exists
//
available = true;
// Close the initialized output mixer
//
if (!wasInitialized)
{
_mixerManager.CloseSpeaker();
}
return 0;
}
int32_t AudioDeviceMac::SetSpeakerVolume(uint32_t volume)
{
return (_mixerManager.SetSpeakerVolume(volume));
}
int32_t AudioDeviceMac::SpeakerVolume(uint32_t& volume) const
{
uint32_t level(0);
if (_mixerManager.SpeakerVolume(level) == -1)
{
return -1;
}
volume = level;
return 0;
}
int32_t AudioDeviceMac::SetWaveOutVolume(uint16_t volumeLeft,
uint16_t volumeRight)
{
WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
" API call not supported on this platform");
return -1;
}
int32_t
AudioDeviceMac::WaveOutVolume(uint16_t& /*volumeLeft*/,
uint16_t& /*volumeRight*/) const
{
WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
" API call not supported on this platform");
return -1;
}
int32_t AudioDeviceMac::MaxSpeakerVolume(uint32_t& maxVolume) const
{
uint32_t maxVol(0);
if (_mixerManager.MaxSpeakerVolume(maxVol) == -1)
{
return -1;
}
maxVolume = maxVol;
return 0;
}
int32_t AudioDeviceMac::MinSpeakerVolume(uint32_t& minVolume) const
{
uint32_t minVol(0);
if (_mixerManager.MinSpeakerVolume(minVol) == -1)
{
return -1;
}
minVolume = minVol;
return 0;
}
int32_t
AudioDeviceMac::SpeakerVolumeStepSize(uint16_t& stepSize) const
{
uint16_t delta(0);
if (_mixerManager.SpeakerVolumeStepSize(delta) == -1)
{
return -1;
}
stepSize = delta;
return 0;
}
int32_t AudioDeviceMac::SpeakerMuteIsAvailable(bool& available)
{
bool isAvailable(false);
bool wasInitialized = _mixerManager.SpeakerIsInitialized();
// Make an attempt to open up the
// output mixer corresponding to the currently selected output device.
//
if (!wasInitialized && InitSpeaker() == -1)
{
// If we end up here it means that the selected speaker has no volume
// control, hence it is safe to state that there is no mute control
// already at this stage.
available = false;
return 0;
}
// Check if the selected speaker has a mute control
//
_mixerManager.SpeakerMuteIsAvailable(isAvailable);
available = isAvailable;
// Close the initialized output mixer
//
if (!wasInitialized)
{
_mixerManager.CloseSpeaker();
}
return 0;
}
int32_t AudioDeviceMac::SetSpeakerMute(bool enable)
{
return (_mixerManager.SetSpeakerMute(enable));
}
int32_t AudioDeviceMac::SpeakerMute(bool& enabled) const
{
bool muted(0);
if (_mixerManager.SpeakerMute(muted) == -1)
{
return -1;
}
enabled = muted;
return 0;
}
int32_t AudioDeviceMac::MicrophoneMuteIsAvailable(bool& available)
{
bool isAvailable(false);
bool wasInitialized = _mixerManager.MicrophoneIsInitialized();
// Make an attempt to open up the
// input mixer corresponding to the currently selected input device.
//
if (!wasInitialized && InitMicrophone() == -1)
{
// If we end up here it means that the selected microphone has no volume
// control, hence it is safe to state that there is no boost control
// already at this stage.
available = false;
return 0;
}
// Check if the selected microphone has a mute control
//
_mixerManager.MicrophoneMuteIsAvailable(isAvailable);
available = isAvailable;
// Close the initialized input mixer
//
if (!wasInitialized)
{
_mixerManager.CloseMicrophone();
}
return 0;
}
int32_t AudioDeviceMac::SetMicrophoneMute(bool enable)
{
return (_mixerManager.SetMicrophoneMute(enable));
}
int32_t AudioDeviceMac::MicrophoneMute(bool& enabled) const
{
bool muted(0);
if (_mixerManager.MicrophoneMute(muted) == -1)
{
return -1;
}
enabled = muted;
return 0;
}
int32_t AudioDeviceMac::MicrophoneBoostIsAvailable(bool& available)
{
bool isAvailable(false);
bool wasInitialized = _mixerManager.MicrophoneIsInitialized();
// Enumerate all avaliable microphone and make an attempt to open up the
// input mixer corresponding to the currently selected input device.
//
if (!wasInitialized && InitMicrophone() == -1)
{
// If we end up here it means that the selected microphone has no volume
// control, hence it is safe to state that there is no boost control
// already at this stage.
available = false;
return 0;
}
// Check if the selected microphone has a boost control
//
_mixerManager.MicrophoneBoostIsAvailable(isAvailable);
available = isAvailable;
// Close the initialized input mixer
//
if (!wasInitialized)
{
_mixerManager.CloseMicrophone();
}
return 0;
}
int32_t AudioDeviceMac::SetMicrophoneBoost(bool enable)
{
return (_mixerManager.SetMicrophoneBoost(enable));
}
int32_t AudioDeviceMac::MicrophoneBoost(bool& enabled) const
{
bool onOff(0);
if (_mixerManager.MicrophoneBoost(onOff) == -1)
{
return -1;
}
enabled = onOff;
return 0;
}
int32_t AudioDeviceMac::StereoRecordingIsAvailable(bool& available)
{
bool isAvailable(false);
bool wasInitialized = _mixerManager.MicrophoneIsInitialized();
if (!wasInitialized && InitMicrophone() == -1)
{
// Cannot open the specified device
available = false;
return 0;
}
// Check if the selected microphone can record stereo
//
_mixerManager.StereoRecordingIsAvailable(isAvailable);
available = isAvailable;
// Close the initialized input mixer
//
if (!wasInitialized)
{
_mixerManager.CloseMicrophone();
}
return 0;
}
int32_t AudioDeviceMac::SetStereoRecording(bool enable)
{
if (enable)
_recChannels = 2;
else
_recChannels = 1;
return 0;
}
int32_t AudioDeviceMac::StereoRecording(bool& enabled) const
{
if (_recChannels == 2)
enabled = true;
else
enabled = false;
return 0;
}
int32_t AudioDeviceMac::StereoPlayoutIsAvailable(bool& available)
{
bool isAvailable(false);
bool wasInitialized = _mixerManager.SpeakerIsInitialized();
if (!wasInitialized && InitSpeaker() == -1)
{
// Cannot open the specified device
available = false;
return 0;
}
// Check if the selected microphone can record stereo
//
_mixerManager.StereoPlayoutIsAvailable(isAvailable);
available = isAvailable;
// Close the initialized input mixer
//
if (!wasInitialized)
{
_mixerManager.CloseSpeaker();
}
return 0;
}
int32_t AudioDeviceMac::SetStereoPlayout(bool enable)
{
if (enable)
_playChannels = 2;
else
_playChannels = 1;
return 0;
}
int32_t AudioDeviceMac::StereoPlayout(bool& enabled) const
{
if (_playChannels == 2)
enabled = true;
else
enabled = false;
return 0;
}
int32_t AudioDeviceMac::SetAGC(bool enable)
{
_AGC = enable;
return 0;
}
bool AudioDeviceMac::AGC() const
{
return _AGC;
}
int32_t AudioDeviceMac::MicrophoneVolumeIsAvailable(bool& available)
{
bool wasInitialized = _mixerManager.MicrophoneIsInitialized();
// Make an attempt to open up the
// input mixer corresponding to the currently selected output device.
//
if (!wasInitialized && InitMicrophone() == -1)
{
// If we end up here it means that the selected microphone has no volume
// control.
available = false;
return 0;
}
// Given that InitMicrophone was successful, we know that a volume control
// exists
//
available = true;
// Close the initialized input mixer
//
if (!wasInitialized)
{
_mixerManager.CloseMicrophone();
}
return 0;
}
int32_t AudioDeviceMac::SetMicrophoneVolume(uint32_t volume)
{
return (_mixerManager.SetMicrophoneVolume(volume));
}
int32_t AudioDeviceMac::MicrophoneVolume(uint32_t& volume) const
{
uint32_t level(0);
if (_mixerManager.MicrophoneVolume(level) == -1)
{
WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
" failed to retrive current microphone level");
return -1;
}
volume = level;
return 0;
}
int32_t
AudioDeviceMac::MaxMicrophoneVolume(uint32_t& maxVolume) const
{
uint32_t maxVol(0);
if (_mixerManager.MaxMicrophoneVolume(maxVol) == -1)
{
return -1;
}
maxVolume = maxVol;
return 0;
}
int32_t
AudioDeviceMac::MinMicrophoneVolume(uint32_t& minVolume) const
{
uint32_t minVol(0);
if (_mixerManager.MinMicrophoneVolume(minVol) == -1)
{
return -1;
}
minVolume = minVol;
return 0;
}
int32_t
AudioDeviceMac::MicrophoneVolumeStepSize(uint16_t& stepSize) const
{
uint16_t delta(0);
if (_mixerManager.MicrophoneVolumeStepSize(delta) == -1)
{
return -1;
}
stepSize = delta;
return 0;
}
int16_t AudioDeviceMac::PlayoutDevices()
{
AudioDeviceID playDevices[MaxNumberDevices];
return GetNumberDevices(kAudioDevicePropertyScopeOutput, playDevices,
MaxNumberDevices);
}
int32_t AudioDeviceMac::SetPlayoutDevice(uint16_t index)
{
CriticalSectionScoped lock(&_critSect);
if (_playIsInitialized)
{
return -1;
}
AudioDeviceID playDevices[MaxNumberDevices];
uint32_t nDevices = GetNumberDevices(kAudioDevicePropertyScopeOutput,
playDevices, MaxNumberDevices);
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
" number of availiable waveform-audio output devices is %u",
nDevices);
if (index > (nDevices - 1))
{
WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
" device index is out of range [0,%u]", (nDevices - 1));
return -1;
}
_outputDeviceIndex = index;
_outputDeviceIsSpecified = true;
return 0;
}
int32_t AudioDeviceMac::SetPlayoutDevice(
AudioDeviceModule::WindowsDeviceType /*device*/)
{
WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
"WindowsDeviceType not supported");
return -1;
}
int32_t AudioDeviceMac::PlayoutDeviceName(
uint16_t index,
char name[kAdmMaxDeviceNameSize],
char guid[kAdmMaxGuidSize])
{
const uint16_t nDevices(PlayoutDevices());
if ((index > (nDevices - 1)) || (name == NULL))
{
return -1;
}
memset(name, 0, kAdmMaxDeviceNameSize);
if (guid != NULL)
{
memset(guid, 0, kAdmMaxGuidSize);
}
return GetDeviceName(kAudioDevicePropertyScopeOutput, index, name);
}
int32_t AudioDeviceMac::RecordingDeviceName(
uint16_t index,
char name[kAdmMaxDeviceNameSize],
char guid[kAdmMaxGuidSize])
{
const uint16_t nDevices(RecordingDevices());
if ((index > (nDevices - 1)) || (name == NULL))
{
return -1;
}
memset(name, 0, kAdmMaxDeviceNameSize);
if (guid != NULL)
{
memset(guid, 0, kAdmMaxGuidSize);
}
return GetDeviceName(kAudioDevicePropertyScopeInput, index, name);
}
int16_t AudioDeviceMac::RecordingDevices()
{
AudioDeviceID recDevices[MaxNumberDevices];
return GetNumberDevices(kAudioDevicePropertyScopeInput, recDevices,
MaxNumberDevices);
}
int32_t AudioDeviceMac::SetRecordingDevice(uint16_t index)
{
if (_recIsInitialized)
{
return -1;
}
AudioDeviceID recDevices[MaxNumberDevices];
uint32_t nDevices = GetNumberDevices(kAudioDevicePropertyScopeInput,
recDevices, MaxNumberDevices);
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
" number of availiable waveform-audio input devices is %u",
nDevices);
if (index > (nDevices - 1))
{
WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
" device index is out of range [0,%u]", (nDevices - 1));
return -1;
}
_inputDeviceIndex = index;
_inputDeviceIsSpecified = true;
return 0;
}
int32_t
AudioDeviceMac::SetRecordingDevice(AudioDeviceModule::WindowsDeviceType /*device*/)
{
WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
"WindowsDeviceType not supported");
return -1;
}
int32_t AudioDeviceMac::PlayoutIsAvailable(bool& available)
{
available = true;
// Try to initialize the playout side
if (InitPlayout() == -1)
{
available = false;
}
// We destroy the IOProc created by InitPlayout() in implDeviceIOProc().
// We must actually start playout here in order to have the IOProc
// deleted by calling StopPlayout().
if (StartPlayout() == -1)
{
available = false;
}
// Cancel effect of initialization
if (StopPlayout() == -1)
{
available = false;
}
return 0;
}
int32_t AudioDeviceMac::RecordingIsAvailable(bool& available)
{
available = true;
// Try to initialize the recording side
if (InitRecording() == -1)
{
available = false;
}
// We destroy the IOProc created by InitRecording() in implInDeviceIOProc().
// We must actually start recording here in order to have the IOProc
// deleted by calling StopRecording().
if (StartRecording() == -1)
{
available = false;
}
// Cancel effect of initialization
if (StopRecording() == -1)
{
available = false;
}
return 0;
}
int32_t AudioDeviceMac::InitPlayout()
{
CriticalSectionScoped lock(&_critSect);
if (_playing)
{
return -1;
}
if (!_outputDeviceIsSpecified)
{
return -1;
}
if (_playIsInitialized)
{
return 0;
}
// Initialize the speaker (devices might have been added or removed)
if (InitSpeaker() == -1)
{
WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
" InitSpeaker() failed");
}
if (!MicrophoneIsInitialized())
{
// Make this call to check if we are using
// one or two devices (_twoDevices)
bool available = false;
if (MicrophoneIsAvailable(available) == -1)
{
WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
" MicrophoneIsAvailable() failed");
}
}
PaUtil_FlushRingBuffer(_paRenderBuffer);
OSStatus err = noErr;
UInt32 size = 0;
_renderDelayOffsetSamples = 0;
_renderDelayUs = 0;
_renderLatencyUs = 0;
_renderDeviceIsAlive = 1;
_doStop = false;
// The internal microphone of a MacBook Pro is located under the left speaker
// grille. When the internal speakers are in use, we want to fully stereo
// pan to the right.
AudioObjectPropertyAddress
propertyAddress = { kAudioDevicePropertyDataSource,
kAudioDevicePropertyScopeOutput, 0 };
if (_macBookPro)
{
_macBookProPanRight = false;
Boolean hasProperty = AudioObjectHasProperty(_outputDeviceID,
&propertyAddress);
if (hasProperty)
{
UInt32 dataSource = 0;
size = sizeof(dataSource);
WEBRTC_CA_LOG_WARN(AudioObjectGetPropertyData(_outputDeviceID,
&propertyAddress, 0, NULL, &size, &dataSource));
if (dataSource == 'ispk')
{
_macBookProPanRight = true;
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice,
_id,
"MacBook Pro using internal speakers; stereo"
" panning right");
} else
{
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice,
_id, "MacBook Pro not using internal speakers");
}
// Add a listener to determine if the status changes.
WEBRTC_CA_LOG_WARN(AudioObjectAddPropertyListener(_outputDeviceID,
&propertyAddress, &objectListenerProc, this));
}
}
// Get current stream description
propertyAddress.mSelector = kAudioDevicePropertyStreamFormat;
memset(&_outStreamFormat, 0, sizeof(_outStreamFormat));
size = sizeof(_outStreamFormat);
WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(_outputDeviceID,
&propertyAddress, 0, NULL, &size, &_outStreamFormat));
if (_outStreamFormat.mFormatID != kAudioFormatLinearPCM)
{
logCAMsg(kTraceError, kTraceAudioDevice, _id,
"Unacceptable output stream format -> mFormatID",
(const char *) &_outStreamFormat.mFormatID);
return -1;
}
if (_outStreamFormat.mChannelsPerFrame > N_DEVICE_CHANNELS)
{
WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
"Too many channels on output device (mChannelsPerFrame = %d)",
_outStreamFormat.mChannelsPerFrame);
return -1;
}
if (_outStreamFormat.mFormatFlags & kAudioFormatFlagIsNonInterleaved)
{
WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
"Non-interleaved audio data is not supported.",
"AudioHardware streams should not have this format.");
return -1;
}
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
"Ouput stream format:");
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
"mSampleRate = %f, mChannelsPerFrame = %u",
_outStreamFormat.mSampleRate,
_outStreamFormat.mChannelsPerFrame);
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
"mBytesPerPacket = %u, mFramesPerPacket = %u",
_outStreamFormat.mBytesPerPacket,
_outStreamFormat.mFramesPerPacket);
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
"mBytesPerFrame = %u, mBitsPerChannel = %u",
_outStreamFormat.mBytesPerFrame,
_outStreamFormat.mBitsPerChannel);
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
"mFormatFlags = %u",
_outStreamFormat.mFormatFlags);
logCAMsg(kTraceInfo, kTraceAudioDevice, _id, "mFormatID",
(const char *) &_outStreamFormat.mFormatID);
// Our preferred format to work with.
if (_outStreamFormat.mChannelsPerFrame < 2)
{
// Disable stereo playout when we only have one channel on the device.
_playChannels = 1;
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
"Stereo playout unavailable on this device");
}
WEBRTC_CA_RETURN_ON_ERR(SetDesiredPlayoutFormat());
// Listen for format changes.
propertyAddress.mSelector = kAudioDevicePropertyStreamFormat;
WEBRTC_CA_RETURN_ON_ERR(AudioObjectAddPropertyListener(_outputDeviceID,
&propertyAddress,
&objectListenerProc,
this));
// Listen for processor overloads.
propertyAddress.mSelector = kAudioDeviceProcessorOverload;
WEBRTC_CA_LOG_WARN(AudioObjectAddPropertyListener(_outputDeviceID,
&propertyAddress,
&objectListenerProc,
this));
if (_twoDevices || !_recIsInitialized)
{
WEBRTC_CA_RETURN_ON_ERR(AudioDeviceCreateIOProcID(_outputDeviceID,
deviceIOProc,
this,
&_deviceIOProcID));
}
_playIsInitialized = true;
return 0;
}
int32_t AudioDeviceMac::InitRecording()
{
CriticalSectionScoped lock(&_critSect);
if (_recording)
{
return -1;
}
if (!_inputDeviceIsSpecified)
{
return -1;
}
if (_recIsInitialized)
{
return 0;
}
// Initialize the microphone (devices might have been added or removed)
if (InitMicrophone() == -1)
{
WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
" InitMicrophone() failed");
}
if (!SpeakerIsInitialized())
{
// Make this call to check if we are using
// one or two devices (_twoDevices)
bool available = false;
if (SpeakerIsAvailable(available) == -1)
{
WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
" SpeakerIsAvailable() failed");
}
}
OSStatus err = noErr;
UInt32 size = 0;
PaUtil_FlushRingBuffer(_paCaptureBuffer);
_captureDelayUs = 0;
_captureLatencyUs = 0;
_captureDeviceIsAlive = 1;
_doStopRec = false;
// Get current stream description
AudioObjectPropertyAddress
propertyAddress = { kAudioDevicePropertyStreamFormat,
kAudioDevicePropertyScopeInput, 0 };
memset(&_inStreamFormat, 0, sizeof(_inStreamFormat));
size = sizeof(_inStreamFormat);
WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(_inputDeviceID,
&propertyAddress, 0, NULL, &size, &_inStreamFormat));
if (_inStreamFormat.mFormatID != kAudioFormatLinearPCM)
{
logCAMsg(kTraceError, kTraceAudioDevice, _id,
"Unacceptable input stream format -> mFormatID",
(const char *) &_inStreamFormat.mFormatID);
return -1;
}
if (_inStreamFormat.mChannelsPerFrame > N_DEVICE_CHANNELS)
{
WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
"Too many channels on input device (mChannelsPerFrame = %d)",
_inStreamFormat.mChannelsPerFrame);
return -1;
}
const int io_block_size_samples = _inStreamFormat.mChannelsPerFrame *
_inStreamFormat.mSampleRate / 100 * N_BLOCKS_IO;
if (io_block_size_samples > _captureBufSizeSamples)
{
WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
"Input IO block size (%d) is larger than ring buffer (%u)",
io_block_size_samples, _captureBufSizeSamples);
return -1;
}
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
" Input stream format:");
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
" mSampleRate = %f, mChannelsPerFrame = %u",
_inStreamFormat.mSampleRate, _inStreamFormat.mChannelsPerFrame);
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
" mBytesPerPacket = %u, mFramesPerPacket = %u",
_inStreamFormat.mBytesPerPacket,
_inStreamFormat.mFramesPerPacket);
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
" mBytesPerFrame = %u, mBitsPerChannel = %u",
_inStreamFormat.mBytesPerFrame,
_inStreamFormat.mBitsPerChannel);
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
" mFormatFlags = %u",
_inStreamFormat.mFormatFlags);
logCAMsg(kTraceInfo, kTraceAudioDevice, _id, "mFormatID",
(const char *) &_inStreamFormat.mFormatID);
// Our preferred format to work with
if (_inStreamFormat.mChannelsPerFrame >= 2 && (_recChannels == 2))
{
_inDesiredFormat.mChannelsPerFrame = 2;
} else
{
// Disable stereo recording when we only have one channel on the device.
_inDesiredFormat.mChannelsPerFrame = 1;
_recChannels = 1;
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
"Stereo recording unavailable on this device");
}
if (_ptrAudioBuffer)
{
// Update audio buffer with the selected parameters
_ptrAudioBuffer->SetRecordingSampleRate(N_REC_SAMPLES_PER_SEC);
_ptrAudioBuffer->SetRecordingChannels((uint8_t) _recChannels);
}
_inDesiredFormat.mSampleRate = N_REC_SAMPLES_PER_SEC;
_inDesiredFormat.mBytesPerPacket = _inDesiredFormat.mChannelsPerFrame
* sizeof(SInt16);
_inDesiredFormat.mFramesPerPacket = 1;
_inDesiredFormat.mBytesPerFrame = _inDesiredFormat.mChannelsPerFrame
* sizeof(SInt16);
_inDesiredFormat.mBitsPerChannel = sizeof(SInt16) * 8;
_inDesiredFormat.mFormatFlags = kLinearPCMFormatFlagIsSignedInteger
| kLinearPCMFormatFlagIsPacked;
#ifdef WEBRTC_ARCH_BIG_ENDIAN
_inDesiredFormat.mFormatFlags |= kLinearPCMFormatFlagIsBigEndian;
#endif
_inDesiredFormat.mFormatID = kAudioFormatLinearPCM;
WEBRTC_CA_RETURN_ON_ERR(AudioConverterNew(&_inStreamFormat, &_inDesiredFormat,
&_captureConverter));
// First try to set buffer size to desired value (10 ms * N_BLOCKS_IO)
// TODO(xians): investigate this block.
UInt32 bufByteCount = (UInt32)((_inStreamFormat.mSampleRate / 1000.0)
* 10.0 * N_BLOCKS_IO * _inStreamFormat.mChannelsPerFrame
* sizeof(Float32));
if (_inStreamFormat.mFramesPerPacket != 0)
{
if (bufByteCount % _inStreamFormat.mFramesPerPacket != 0)
{
bufByteCount = ((UInt32)(bufByteCount
/ _inStreamFormat.mFramesPerPacket) + 1)
* _inStreamFormat.mFramesPerPacket;
}
}
// Ensure the buffer size is within the acceptable range provided by the device.
propertyAddress.mSelector = kAudioDevicePropertyBufferSizeRange;
AudioValueRange range;
size = sizeof(range);
WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(_inputDeviceID,
&propertyAddress, 0, NULL, &size, &range));
if (range.mMinimum > bufByteCount)
{
bufByteCount = range.mMinimum;
} else if (range.mMaximum < bufByteCount)
{
bufByteCount = range.mMaximum;
}
propertyAddress.mSelector = kAudioDevicePropertyBufferSize;
size = sizeof(bufByteCount);
WEBRTC_CA_RETURN_ON_ERR(AudioObjectSetPropertyData(_inputDeviceID,
&propertyAddress, 0, NULL, size, &bufByteCount));
// Get capture device latency
propertyAddress.mSelector = kAudioDevicePropertyLatency;
UInt32 latency = 0;
size = sizeof(UInt32);
WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(_inputDeviceID,
&propertyAddress, 0, NULL, &size, &latency));
_captureLatencyUs = (UInt32)((1.0e6 * latency)
/ _inStreamFormat.mSampleRate);
// Get capture stream latency
propertyAddress.mSelector = kAudioDevicePropertyStreams;
AudioStreamID stream = 0;
size = sizeof(AudioStreamID);
WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(_inputDeviceID,
&propertyAddress, 0, NULL, &size, &stream));
propertyAddress.mSelector = kAudioStreamPropertyLatency;
size = sizeof(UInt32);
latency = 0;
WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(_inputDeviceID,
&propertyAddress, 0, NULL, &size, &latency));
_captureLatencyUs += (UInt32)((1.0e6 * latency)
/ _inStreamFormat.mSampleRate);
// Listen for format changes
// TODO(xians): should we be using kAudioDevicePropertyDeviceHasChanged?
propertyAddress.mSelector = kAudioDevicePropertyStreamFormat;
WEBRTC_CA_RETURN_ON_ERR(AudioObjectAddPropertyListener(_inputDeviceID,
&propertyAddress, &objectListenerProc, this));
// Listen for processor overloads
propertyAddress.mSelector = kAudioDeviceProcessorOverload;
WEBRTC_CA_LOG_WARN(AudioObjectAddPropertyListener(_inputDeviceID,
&propertyAddress, &objectListenerProc, this));
if (_twoDevices)
{
WEBRTC_CA_RETURN_ON_ERR(AudioDeviceCreateIOProcID(_inputDeviceID,
inDeviceIOProc, this, &_inDeviceIOProcID));
} else if (!_playIsInitialized)
{
WEBRTC_CA_RETURN_ON_ERR(AudioDeviceCreateIOProcID(_inputDeviceID,
deviceIOProc, this, &_deviceIOProcID));
}
// Mark recording side as initialized
_recIsInitialized = true;
return 0;
}
int32_t AudioDeviceMac::StartRecording()
{
CriticalSectionScoped lock(&_critSect);
if (!_recIsInitialized)
{
return -1;
}
if (_recording)
{
return 0;
}
if (!_initialized)
{
WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
" Recording worker thread has not been started");
return -1;
}
DCHECK(!capture_worker_thread_.get());
capture_worker_thread_ =
ThreadWrapper::CreateThread(RunCapture, this, "CaptureWorkerThread");
DCHECK(capture_worker_thread_.get());
capture_worker_thread_->Start();
capture_worker_thread_->SetPriority(kRealtimePriority);
OSStatus err = noErr;
if (_twoDevices)
{
WEBRTC_CA_RETURN_ON_ERR(AudioDeviceStart(_inputDeviceID, _inDeviceIOProcID));
} else if (!_playing)
{
WEBRTC_CA_RETURN_ON_ERR(AudioDeviceStart(_inputDeviceID, _deviceIOProcID));
}
_recording = true;
return 0;
}
int32_t AudioDeviceMac::StopRecording()
{
CriticalSectionScoped lock(&_critSect);
if (!_recIsInitialized)
{
return 0;
}
OSStatus err = noErr;
// Stop device
int32_t captureDeviceIsAlive = AtomicGet32(&_captureDeviceIsAlive);
if (_twoDevices)
{
if (_recording && captureDeviceIsAlive == 1)
{
_recording = false;
_doStopRec = true; // Signal to io proc to stop audio device
_critSect.Leave(); // Cannot be under lock, risk of deadlock
if (kEventTimeout == _stopEventRec.Wait(2000))
{
CriticalSectionScoped critScoped(&_critSect);
WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
" Timed out stopping the capture IOProc. "
"We may have failed to detect a device removal.");
WEBRTC_CA_LOG_WARN(AudioDeviceStop(_inputDeviceID,
_inDeviceIOProcID));
WEBRTC_CA_LOG_WARN(
AudioDeviceDestroyIOProcID(_inputDeviceID,
_inDeviceIOProcID));
}
_critSect.Enter();
_doStopRec = false;
WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
" Recording stopped");
}
}
else
{
// We signal a stop for a shared device even when rendering has
// not yet ended. This is to ensure the IOProc will return early as
// intended (by checking |_recording|) before accessing
// resources we free below (e.g. the capture converter).
//
// In the case of a shared devcie, the IOProc will verify
// rendering has ended before stopping itself.
if (_recording && captureDeviceIsAlive == 1)
{
_recording = false;
_doStop = true; // Signal to io proc to stop audio device
_critSect.Leave(); // Cannot be under lock, risk of deadlock
if (kEventTimeout == _stopEvent.Wait(2000))
{
CriticalSectionScoped critScoped(&_critSect);
WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
" Timed out stopping the shared IOProc. "
"We may have failed to detect a device removal.");
// We assume rendering on a shared device has stopped as well if
// the IOProc times out.
WEBRTC_CA_LOG_WARN(AudioDeviceStop(_outputDeviceID,
_deviceIOProcID));
WEBRTC_CA_LOG_WARN(AudioDeviceDestroyIOProcID(_outputDeviceID,
_deviceIOProcID));
}
_critSect.Enter();
_doStop = false;
WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
" Recording stopped (shared)");
}
}
// Setting this signal will allow the worker thread to be stopped.
AtomicSet32(&_captureDeviceIsAlive, 0);
if (capture_worker_thread_.get()) {
_critSect.Leave();
capture_worker_thread_->Stop();
capture_worker_thread_.reset();
_critSect.Enter();
}
WEBRTC_CA_LOG_WARN(AudioConverterDispose(_captureConverter));
// Remove listeners.
AudioObjectPropertyAddress
propertyAddress = { kAudioDevicePropertyStreamFormat,
kAudioDevicePropertyScopeInput, 0 };
WEBRTC_CA_LOG_WARN(AudioObjectRemovePropertyListener(_inputDeviceID,
&propertyAddress, &objectListenerProc, this));
propertyAddress.mSelector = kAudioDeviceProcessorOverload;
WEBRTC_CA_LOG_WARN(AudioObjectRemovePropertyListener(_inputDeviceID,
&propertyAddress, &objectListenerProc, this));
_recIsInitialized = false;
_recording = false;
return 0;
}
bool AudioDeviceMac::RecordingIsInitialized() const
{
return (_recIsInitialized);
}
bool AudioDeviceMac::Recording() const
{
return (_recording);
}
bool AudioDeviceMac::PlayoutIsInitialized() const
{
return (_playIsInitialized);
}
int32_t AudioDeviceMac::StartPlayout()
{
CriticalSectionScoped lock(&_critSect);
if (!_playIsInitialized)
{
return -1;
}
if (_playing)
{
return 0;
}
DCHECK(!render_worker_thread_.get());
render_worker_thread_ =
ThreadWrapper::CreateThread(RunRender, this, "RenderWorkerThread");
render_worker_thread_->Start();
render_worker_thread_->SetPriority(kRealtimePriority);
if (_twoDevices || !_recording)
{
OSStatus err = noErr;
WEBRTC_CA_RETURN_ON_ERR(AudioDeviceStart(_outputDeviceID, _deviceIOProcID));
}
_playing = true;
return 0;
}
int32_t AudioDeviceMac::StopPlayout()
{
CriticalSectionScoped lock(&_critSect);
if (!_playIsInitialized)
{
return 0;
}
OSStatus err = noErr;
int32_t renderDeviceIsAlive = AtomicGet32(&_renderDeviceIsAlive);
if (_playing && renderDeviceIsAlive == 1)
{
// We signal a stop for a shared device even when capturing has not
// yet ended. This is to ensure the IOProc will return early as
// intended (by checking |_playing|) before accessing resources we
// free below (e.g. the render converter).
//
// In the case of a shared device, the IOProc will verify capturing
// has ended before stopping itself.
_playing = false;
_doStop = true; // Signal to io proc to stop audio device
_critSect.Leave(); // Cannot be under lock, risk of deadlock
if (kEventTimeout == _stopEvent.Wait(2000))
{
CriticalSectionScoped critScoped(&_critSect);
WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
" Timed out stopping the render IOProc. "
"We may have failed to detect a device removal.");
// We assume capturing on a shared device has stopped as well if the
// IOProc times out.
WEBRTC_CA_LOG_WARN(AudioDeviceStop(_outputDeviceID,
_deviceIOProcID));
WEBRTC_CA_LOG_WARN(AudioDeviceDestroyIOProcID(_outputDeviceID,
_deviceIOProcID));
}
_critSect.Enter();
_doStop = false;
WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
"Playout stopped");
}
// Setting this signal will allow the worker thread to be stopped.
AtomicSet32(&_renderDeviceIsAlive, 0);
if (render_worker_thread_.get()) {
_critSect.Leave();
render_worker_thread_->Stop();
render_worker_thread_.reset();
_critSect.Enter();
}
WEBRTC_CA_LOG_WARN(AudioConverterDispose(_renderConverter));
// Remove listeners.
AudioObjectPropertyAddress propertyAddress = {
kAudioDevicePropertyStreamFormat, kAudioDevicePropertyScopeOutput,
0 };
WEBRTC_CA_LOG_WARN(AudioObjectRemovePropertyListener(_outputDeviceID,
&propertyAddress, &objectListenerProc, this));
propertyAddress.mSelector = kAudioDeviceProcessorOverload;
WEBRTC_CA_LOG_WARN(AudioObjectRemovePropertyListener(_outputDeviceID,
&propertyAddress, &objectListenerProc, this));
if (_macBookPro)
{
Boolean hasProperty = AudioObjectHasProperty(_outputDeviceID,
&propertyAddress);
if (hasProperty)
{
propertyAddress.mSelector = kAudioDevicePropertyDataSource;
WEBRTC_CA_LOG_WARN(AudioObjectRemovePropertyListener(_outputDeviceID,
&propertyAddress, &objectListenerProc, this));
}
}
_playIsInitialized = false;
_playing = false;
return 0;
}
int32_t AudioDeviceMac::PlayoutDelay(uint16_t& delayMS) const
{
int32_t renderDelayUs = AtomicGet32(&_renderDelayUs);
delayMS = static_cast<uint16_t> (1e-3 * (renderDelayUs + _renderLatencyUs) +
0.5);
return 0;
}
int32_t AudioDeviceMac::RecordingDelay(uint16_t& delayMS) const
{
int32_t captureDelayUs = AtomicGet32(&_captureDelayUs);
delayMS = static_cast<uint16_t> (1e-3 * (captureDelayUs +
_captureLatencyUs) + 0.5);
return 0;
}
bool AudioDeviceMac::Playing() const
{
return (_playing);
}
int32_t AudioDeviceMac::SetPlayoutBuffer(
const AudioDeviceModule::BufferType type,
uint16_t sizeMS)
{
if (type != AudioDeviceModule::kFixedBufferSize)
{
WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
" Adaptive buffer size not supported on this platform");
return -1;
}
_playBufType = type;
_playBufDelayFixed = sizeMS;
return 0;
}
int32_t AudioDeviceMac::PlayoutBuffer(
AudioDeviceModule::BufferType& type,
uint16_t& sizeMS) const
{
type = _playBufType;
sizeMS = _playBufDelayFixed;
return 0;
}
// Not implemented for Mac.
int32_t AudioDeviceMac::CPULoad(uint16_t& /*load*/) const
{
WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
" API call not supported on this platform");
return -1;
}
bool AudioDeviceMac::PlayoutWarning() const
{
return (_playWarning > 0);
}
bool AudioDeviceMac::PlayoutError() const
{
return (_playError > 0);
}
bool AudioDeviceMac::RecordingWarning() const
{
return (_recWarning > 0);
}
bool AudioDeviceMac::RecordingError() const
{
return (_recError > 0);
}
void AudioDeviceMac::ClearPlayoutWarning()
{
_playWarning = 0;
}
void AudioDeviceMac::ClearPlayoutError()
{
_playError = 0;
}
void AudioDeviceMac::ClearRecordingWarning()
{
_recWarning = 0;
}
void AudioDeviceMac::ClearRecordingError()
{
_recError = 0;
}
// ============================================================================
// Private Methods
// ============================================================================
int32_t
AudioDeviceMac::GetNumberDevices(const AudioObjectPropertyScope scope,
AudioDeviceID scopedDeviceIds[],
const uint32_t deviceListLength)
{
OSStatus err = noErr;
AudioObjectPropertyAddress propertyAddress = {
kAudioHardwarePropertyDevices, kAudioObjectPropertyScopeGlobal,
kAudioObjectPropertyElementMaster };
UInt32 size = 0;
WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyDataSize(kAudioObjectSystemObject,
&propertyAddress, 0, NULL, &size));
if (size == 0)
{
WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
"No devices");
return 0;
}
AudioDeviceID* deviceIds = (AudioDeviceID*) malloc(size);
UInt32 numberDevices = size / sizeof(AudioDeviceID);
AudioBufferList* bufferList = NULL;
UInt32 numberScopedDevices = 0;
// First check if there is a default device and list it
UInt32 hardwareProperty = 0;
if (scope == kAudioDevicePropertyScopeOutput)
{
hardwareProperty = kAudioHardwarePropertyDefaultOutputDevice;
} else
{
hardwareProperty = kAudioHardwarePropertyDefaultInputDevice;
}
AudioObjectPropertyAddress
propertyAddressDefault = { hardwareProperty,
kAudioObjectPropertyScopeGlobal,
kAudioObjectPropertyElementMaster };
AudioDeviceID usedID;
UInt32 uintSize = sizeof(UInt32);
WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(kAudioObjectSystemObject,
&propertyAddressDefault, 0, NULL, &uintSize, &usedID));
if (usedID != kAudioDeviceUnknown)
{
scopedDeviceIds[numberScopedDevices] = usedID;
numberScopedDevices++;
} else
{
WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
"GetNumberDevices(): Default device unknown");
}
// Then list the rest of the devices
bool listOK = true;
WEBRTC_CA_LOG_ERR(AudioObjectGetPropertyData(kAudioObjectSystemObject,
&propertyAddress, 0, NULL, &size, deviceIds));
if (err != noErr)
{
listOK = false;
} else
{
propertyAddress.mSelector = kAudioDevicePropertyStreamConfiguration;
propertyAddress.mScope = scope;
propertyAddress.mElement = 0;
for (UInt32 i = 0; i < numberDevices; i++)
{
// Check for input channels
WEBRTC_CA_LOG_ERR(AudioObjectGetPropertyDataSize(deviceIds[i],
&propertyAddress, 0, NULL, &size));
if (err == kAudioHardwareBadDeviceError)
{
// This device doesn't actually exist; continue iterating.
continue;
} else if (err != noErr)
{
listOK = false;
break;
}
bufferList = (AudioBufferList*) malloc(size);
WEBRTC_CA_LOG_ERR(AudioObjectGetPropertyData(deviceIds[i],
&propertyAddress, 0, NULL, &size, bufferList));
if (err != noErr)
{
listOK = false;
break;
}
if (bufferList->mNumberBuffers > 0)
{
if (numberScopedDevices >= deviceListLength)
{
WEBRTC_TRACE(kTraceError,
kTraceAudioDevice, _id,
"Device list is not long enough");
listOK = false;
break;
}
scopedDeviceIds[numberScopedDevices] = deviceIds[i];
numberScopedDevices++;
}
free(bufferList);
bufferList = NULL;
} // for
}
if (!listOK)
{
if (deviceIds)
{
free(deviceIds);
deviceIds = NULL;
}
if (bufferList)
{
free(bufferList);
bufferList = NULL;
}
return -1;
}
// Happy ending
if (deviceIds)
{
free(deviceIds);
deviceIds = NULL;
}
return numberScopedDevices;
}
int32_t
AudioDeviceMac::GetDeviceName(const AudioObjectPropertyScope scope,
const uint16_t index,
char* name)
{
OSStatus err = noErr;
UInt32 len = kAdmMaxDeviceNameSize;
AudioDeviceID deviceIds[MaxNumberDevices];
int numberDevices = GetNumberDevices(scope, deviceIds, MaxNumberDevices);
if (numberDevices < 0)
{
return -1;
} else if (numberDevices == 0)
{
WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
"No devices");
return -1;
}
// If the number is below the number of devices, assume it's "WEBRTC ID"
// otherwise assume it's a CoreAudio ID
AudioDeviceID usedID;
// Check if there is a default device
bool isDefaultDevice = false;
if (index == 0)
{
UInt32 hardwareProperty = 0;
if (scope == kAudioDevicePropertyScopeOutput)
{
hardwareProperty = kAudioHardwarePropertyDefaultOutputDevice;
} else
{
hardwareProperty = kAudioHardwarePropertyDefaultInputDevice;
}
AudioObjectPropertyAddress propertyAddress = { hardwareProperty,
kAudioObjectPropertyScopeGlobal,
kAudioObjectPropertyElementMaster };
UInt32 size = sizeof(UInt32);
WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(kAudioObjectSystemObject,
&propertyAddress, 0, NULL, &size, &usedID));
if (usedID == kAudioDeviceUnknown)
{
WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
"GetDeviceName(): Default device unknown");
} else
{
isDefaultDevice = true;
}
}
AudioObjectPropertyAddress propertyAddress = {
kAudioDevicePropertyDeviceName, scope, 0 };
if (isDefaultDevice)
{
char devName[len];
WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(usedID,
&propertyAddress, 0, NULL, &len, devName));
sprintf(name, "default (%s)", devName);
} else
{
if (index < numberDevices)
{
usedID = deviceIds[index];
} else
{
usedID = index;
}
WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(usedID,
&propertyAddress, 0, NULL, &len, name));
}
return 0;
}
int32_t AudioDeviceMac::InitDevice(const uint16_t userDeviceIndex,
AudioDeviceID& deviceId,
const bool isInput)
{
OSStatus err = noErr;
UInt32 size = 0;
AudioObjectPropertyScope deviceScope;
AudioObjectPropertySelector defaultDeviceSelector;
AudioDeviceID deviceIds[MaxNumberDevices];
if (isInput)
{
deviceScope = kAudioDevicePropertyScopeInput;
defaultDeviceSelector = kAudioHardwarePropertyDefaultInputDevice;
} else
{
deviceScope = kAudioDevicePropertyScopeOutput;
defaultDeviceSelector = kAudioHardwarePropertyDefaultOutputDevice;
}
AudioObjectPropertyAddress
propertyAddress = { defaultDeviceSelector,
kAudioObjectPropertyScopeGlobal,
kAudioObjectPropertyElementMaster };
// Get the actual device IDs
int numberDevices = GetNumberDevices(deviceScope, deviceIds,
MaxNumberDevices);
if (numberDevices < 0)
{
return -1;
} else if (numberDevices == 0)
{
WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
"InitDevice(): No devices");
return -1;
}
bool isDefaultDevice = false;
deviceId = kAudioDeviceUnknown;
if (userDeviceIndex == 0)
{
// Try to use default system device
size = sizeof(AudioDeviceID);
WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(kAudioObjectSystemObject,
&propertyAddress, 0, NULL, &size, &deviceId));
if (deviceId == kAudioDeviceUnknown)
{
WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
" No default device exists");
} else
{
isDefaultDevice = true;
}
}
if (!isDefaultDevice)
{
deviceId = deviceIds[userDeviceIndex];
}
// Obtain device name and manufacturer for logging.
// Also use this as a test to ensure a user-set device ID is valid.
char devName[128];
char devManf[128];
memset(devName, 0, sizeof(devName));
memset(devManf, 0, sizeof(devManf));
propertyAddress.mSelector = kAudioDevicePropertyDeviceName;
propertyAddress.mScope = deviceScope;
propertyAddress.mElement = 0;
size = sizeof(devName);
WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(deviceId,
&propertyAddress, 0, NULL, &size, devName));
propertyAddress.mSelector = kAudioDevicePropertyDeviceManufacturer;
size = sizeof(devManf);
WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(deviceId,
&propertyAddress, 0, NULL, &size, devManf));
if (isInput)
{
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
" Input device: %s %s", devManf, devName);
} else
{
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
" Output device: %s %s", devManf, devName);
}
return 0;
}
OSStatus AudioDeviceMac::SetDesiredPlayoutFormat()
{
// Our preferred format to work with.
_outDesiredFormat.mSampleRate = N_PLAY_SAMPLES_PER_SEC;
_outDesiredFormat.mChannelsPerFrame = _playChannels;
if (_ptrAudioBuffer)
{
// Update audio buffer with the selected parameters.
_ptrAudioBuffer->SetPlayoutSampleRate(N_PLAY_SAMPLES_PER_SEC);
_ptrAudioBuffer->SetPlayoutChannels((uint8_t) _playChannels);
}
_renderDelayOffsetSamples = _renderBufSizeSamples - N_BUFFERS_OUT *
ENGINE_PLAY_BUF_SIZE_IN_SAMPLES * _outDesiredFormat.mChannelsPerFrame;
_outDesiredFormat.mBytesPerPacket = _outDesiredFormat.mChannelsPerFrame *
sizeof(SInt16);
// In uncompressed audio, a packet is one frame.
_outDesiredFormat.mFramesPerPacket = 1;
_outDesiredFormat.mBytesPerFrame = _outDesiredFormat.mChannelsPerFrame *
sizeof(SInt16);
_outDesiredFormat.mBitsPerChannel = sizeof(SInt16) * 8;
_outDesiredFormat.mFormatFlags = kLinearPCMFormatFlagIsSignedInteger |
kLinearPCMFormatFlagIsPacked;
#ifdef WEBRTC_ARCH_BIG_ENDIAN
_outDesiredFormat.mFormatFlags |= kLinearPCMFormatFlagIsBigEndian;
#endif
_outDesiredFormat.mFormatID = kAudioFormatLinearPCM;
OSStatus err = noErr;
WEBRTC_CA_RETURN_ON_ERR(AudioConverterNew(&_outDesiredFormat,
&_outStreamFormat,
&_renderConverter));
// Try to set buffer size to desired value (_playBufDelayFixed).
UInt32 bufByteCount = static_cast<UInt32> ((_outStreamFormat.mSampleRate /
1000.0) *
_playBufDelayFixed *
_outStreamFormat.mChannelsPerFrame *
sizeof(Float32));
if (_outStreamFormat.mFramesPerPacket != 0)
{
if (bufByteCount % _outStreamFormat.mFramesPerPacket != 0)
{
bufByteCount = (static_cast<UInt32> (bufByteCount /
_outStreamFormat.mFramesPerPacket) + 1) *
_outStreamFormat.mFramesPerPacket;
}
}
// Ensure the buffer size is within the range provided by the device.
AudioObjectPropertyAddress propertyAddress =
{kAudioDevicePropertyDataSource,
kAudioDevicePropertyScopeOutput,
0};
propertyAddress.mSelector = kAudioDevicePropertyBufferSizeRange;
AudioValueRange range;
UInt32 size = sizeof(range);
WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(_outputDeviceID,
&propertyAddress,
0,
NULL,
&size,
&range));
if (range.mMinimum > bufByteCount)
{
bufByteCount = range.mMinimum;
} else if (range.mMaximum < bufByteCount)
{
bufByteCount = range.mMaximum;
}
propertyAddress.mSelector = kAudioDevicePropertyBufferSize;
size = sizeof(bufByteCount);
WEBRTC_CA_RETURN_ON_ERR(AudioObjectSetPropertyData(_outputDeviceID,
&propertyAddress,
0,
NULL,
size,
&bufByteCount));
// Get render device latency.
propertyAddress.mSelector = kAudioDevicePropertyLatency;
UInt32 latency = 0;
size = sizeof(UInt32);
WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(_outputDeviceID,
&propertyAddress,
0,
NULL,
&size,
&latency));
_renderLatencyUs = static_cast<uint32_t> ((1.0e6 * latency) /
_outStreamFormat.mSampleRate);
// Get render stream latency.
propertyAddress.mSelector = kAudioDevicePropertyStreams;
AudioStreamID stream = 0;
size = sizeof(AudioStreamID);
WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(_outputDeviceID,
&propertyAddress,
0,
NULL,
&size,
&stream));
propertyAddress.mSelector = kAudioStreamPropertyLatency;
size = sizeof(UInt32);
latency = 0;
WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(_outputDeviceID,
&propertyAddress,
0,
NULL,
&size,
&latency));
_renderLatencyUs += static_cast<uint32_t> ((1.0e6 * latency) /
_outStreamFormat.mSampleRate);
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
" initial playout status: _renderDelayOffsetSamples=%d,"
" _renderDelayUs=%d, _renderLatencyUs=%d",
_renderDelayOffsetSamples, _renderDelayUs, _renderLatencyUs);
return 0;
}
OSStatus AudioDeviceMac::objectListenerProc(
AudioObjectID objectId,
UInt32 numberAddresses,
const AudioObjectPropertyAddress addresses[],
void* clientData)
{
AudioDeviceMac *ptrThis = (AudioDeviceMac *) clientData;
DCHECK(ptrThis != NULL);
ptrThis->implObjectListenerProc(objectId, numberAddresses, addresses);
// AudioObjectPropertyListenerProc functions are supposed to return 0
return 0;
}
OSStatus AudioDeviceMac::implObjectListenerProc(
const AudioObjectID objectId,
const UInt32 numberAddresses,
const AudioObjectPropertyAddress addresses[])
{
WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
"AudioDeviceMac::implObjectListenerProc()");
for (UInt32 i = 0; i < numberAddresses; i++)
{
if (addresses[i].mSelector == kAudioHardwarePropertyDevices)
{
HandleDeviceChange();
} else if (addresses[i].mSelector == kAudioDevicePropertyStreamFormat)
{
HandleStreamFormatChange(objectId, addresses[i]);
} else if (addresses[i].mSelector == kAudioDevicePropertyDataSource)
{
HandleDataSourceChange(objectId, addresses[i]);
} else if (addresses[i].mSelector == kAudioDeviceProcessorOverload)
{
HandleProcessorOverload(addresses[i]);
}
}
return 0;
}
int32_t AudioDeviceMac::HandleDeviceChange()
{
OSStatus err = noErr;
WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
"kAudioHardwarePropertyDevices");
// A device has changed. Check if our registered devices have been removed.
// Ensure the devices have been initialized, meaning the IDs are valid.
if (MicrophoneIsInitialized())
{
AudioObjectPropertyAddress propertyAddress = {
kAudioDevicePropertyDeviceIsAlive,
kAudioDevicePropertyScopeInput, 0 };
UInt32 deviceIsAlive = 1;
UInt32 size = sizeof(UInt32);
err = AudioObjectGetPropertyData(_inputDeviceID, &propertyAddress, 0,
NULL, &size, &deviceIsAlive);
if (err == kAudioHardwareBadDeviceError || deviceIsAlive == 0)
{
WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
"Capture device is not alive (probably removed)");
AtomicSet32(&_captureDeviceIsAlive, 0);
_mixerManager.CloseMicrophone();
if (_recError == 1)
{
WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice,
_id, " pending recording error exists");
}
_recError = 1; // triggers callback from module process thread
} else if (err != noErr)
{
logCAMsg(kTraceError, kTraceAudioDevice, _id,
"Error in AudioDeviceGetProperty()", (const char*) &err);
return -1;
}
}
if (SpeakerIsInitialized())
{
AudioObjectPropertyAddress propertyAddress = {
kAudioDevicePropertyDeviceIsAlive,
kAudioDevicePropertyScopeOutput, 0 };
UInt32 deviceIsAlive = 1;
UInt32 size = sizeof(UInt32);
err = AudioObjectGetPropertyData(_outputDeviceID, &propertyAddress, 0,
NULL, &size, &deviceIsAlive);
if (err == kAudioHardwareBadDeviceError || deviceIsAlive == 0)
{
WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
"Render device is not alive (probably removed)");
AtomicSet32(&_renderDeviceIsAlive, 0);
_mixerManager.CloseSpeaker();
if (_playError == 1)
{
WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice,
_id, " pending playout error exists");
}
_playError = 1; // triggers callback from module process thread
} else if (err != noErr)
{
logCAMsg(kTraceError, kTraceAudioDevice, _id,
"Error in AudioDeviceGetProperty()", (const char*) &err);
return -1;
}
}
return 0;
}
int32_t AudioDeviceMac::HandleStreamFormatChange(
const AudioObjectID objectId,
const AudioObjectPropertyAddress propertyAddress)
{
OSStatus err = noErr;
WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
"Stream format changed");
if (objectId != _inputDeviceID && objectId != _outputDeviceID)
{
return 0;
}
// Get the new device format
AudioStreamBasicDescription streamFormat;
UInt32 size = sizeof(streamFormat);
WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(objectId,
&propertyAddress, 0, NULL, &size, &streamFormat));
if (streamFormat.mFormatID != kAudioFormatLinearPCM)
{
logCAMsg(kTraceError, kTraceAudioDevice, _id,
"Unacceptable input stream format -> mFormatID",
(const char *) &streamFormat.mFormatID);
return -1;
}
if (streamFormat.mChannelsPerFrame > N_DEVICE_CHANNELS)
{
WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
"Too many channels on device (mChannelsPerFrame = %d)",
streamFormat.mChannelsPerFrame);
return -1;
}
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
"Stream format:");
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
"mSampleRate = %f, mChannelsPerFrame = %u",
streamFormat.mSampleRate, streamFormat.mChannelsPerFrame);
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
"mBytesPerPacket = %u, mFramesPerPacket = %u",
streamFormat.mBytesPerPacket, streamFormat.mFramesPerPacket);
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
"mBytesPerFrame = %u, mBitsPerChannel = %u",
streamFormat.mBytesPerFrame, streamFormat.mBitsPerChannel);
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
"mFormatFlags = %u",
streamFormat.mFormatFlags);
logCAMsg(kTraceInfo, kTraceAudioDevice, _id, "mFormatID",
(const char *) &streamFormat.mFormatID);
if (propertyAddress.mScope == kAudioDevicePropertyScopeInput)
{
const int io_block_size_samples = streamFormat.mChannelsPerFrame *
streamFormat.mSampleRate / 100 * N_BLOCKS_IO;
if (io_block_size_samples > _captureBufSizeSamples)
{
WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
"Input IO block size (%d) is larger than ring buffer (%u)",
io_block_size_samples, _captureBufSizeSamples);
return -1;
}
memcpy(&_inStreamFormat, &streamFormat, sizeof(streamFormat));
if (_inStreamFormat.mChannelsPerFrame >= 2 && (_recChannels == 2))
{
_inDesiredFormat.mChannelsPerFrame = 2;
} else
{
// Disable stereo recording when we only have one channel on the device.
_inDesiredFormat.mChannelsPerFrame = 1;
_recChannels = 1;
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
"Stereo recording unavailable on this device");
}
if (_ptrAudioBuffer)
{
// Update audio buffer with the selected parameters
_ptrAudioBuffer->SetRecordingSampleRate(N_REC_SAMPLES_PER_SEC);
_ptrAudioBuffer->SetRecordingChannels((uint8_t) _recChannels);
}
// Recreate the converter with the new format
// TODO(xians): make this thread safe
WEBRTC_CA_RETURN_ON_ERR(AudioConverterDispose(_captureConverter));
WEBRTC_CA_RETURN_ON_ERR(AudioConverterNew(&streamFormat, &_inDesiredFormat,
&_captureConverter));
} else
{
memcpy(&_outStreamFormat, &streamFormat, sizeof(streamFormat));
// Our preferred format to work with
if (_outStreamFormat.mChannelsPerFrame < 2)
{
_playChannels = 1;
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
"Stereo playout unavailable on this device");
}
WEBRTC_CA_RETURN_ON_ERR(SetDesiredPlayoutFormat());
}
return 0;
}
int32_t AudioDeviceMac::HandleDataSourceChange(
const AudioObjectID objectId,
const AudioObjectPropertyAddress propertyAddress)
{
OSStatus err = noErr;
if (_macBookPro && propertyAddress.mScope
== kAudioDevicePropertyScopeOutput)
{
WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
"Data source changed");
_macBookProPanRight = false;
UInt32 dataSource = 0;
UInt32 size = sizeof(UInt32);
WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(objectId,
&propertyAddress, 0, NULL, &size, &dataSource));
if (dataSource == 'ispk')
{
_macBookProPanRight = true;
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
"MacBook Pro using internal speakers; stereo panning right");
} else
{
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
"MacBook Pro not using internal speakers");
}
}
return 0;
}
int32_t AudioDeviceMac::HandleProcessorOverload(
const AudioObjectPropertyAddress propertyAddress)
{
// TODO(xians): we probably want to notify the user in some way of the
// overload. However, the Windows interpretations of these errors seem to
// be more severe than what ProcessorOverload is thrown for.
//
// We don't log the notification, as it's sent from the HAL's IO thread. We
// don't want to slow it down even further.
if (propertyAddress.mScope == kAudioDevicePropertyScopeInput)
{
//WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "Capture processor
// overload");
//_callback->ProblemIsReported(
// SndCardStreamObserver::ERecordingProblem);
} else
{
//WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
// "Render processor overload");
//_callback->ProblemIsReported(
// SndCardStreamObserver::EPlaybackProblem);
}
return 0;
}
// ============================================================================
// Thread Methods
// ============================================================================
OSStatus AudioDeviceMac::deviceIOProc(AudioDeviceID, const AudioTimeStamp*,
const AudioBufferList* inputData,
const AudioTimeStamp* inputTime,
AudioBufferList* outputData,
const AudioTimeStamp* outputTime,
void *clientData)
{
AudioDeviceMac *ptrThis = (AudioDeviceMac *) clientData;
DCHECK(ptrThis != NULL);
ptrThis->implDeviceIOProc(inputData, inputTime, outputData, outputTime);
// AudioDeviceIOProc functions are supposed to return 0
return 0;
}
OSStatus AudioDeviceMac::outConverterProc(AudioConverterRef,
UInt32 *numberDataPackets,
AudioBufferList *data,
AudioStreamPacketDescription **,
void *userData)
{
AudioDeviceMac *ptrThis = (AudioDeviceMac *) userData;
DCHECK(ptrThis != NULL);
return ptrThis->implOutConverterProc(numberDataPackets, data);
}
OSStatus AudioDeviceMac::inDeviceIOProc(AudioDeviceID, const AudioTimeStamp*,
const AudioBufferList* inputData,
const AudioTimeStamp* inputTime,
AudioBufferList*,
const AudioTimeStamp*, void* clientData)
{
AudioDeviceMac *ptrThis = (AudioDeviceMac *) clientData;
DCHECK(ptrThis != NULL);
ptrThis->implInDeviceIOProc(inputData, inputTime);
// AudioDeviceIOProc functions are supposed to return 0
return 0;
}
OSStatus AudioDeviceMac::inConverterProc(
AudioConverterRef,
UInt32 *numberDataPackets,
AudioBufferList *data,
AudioStreamPacketDescription ** /*dataPacketDescription*/,
void *userData)
{
AudioDeviceMac *ptrThis = static_cast<AudioDeviceMac*> (userData);
DCHECK(ptrThis != NULL);
return ptrThis->implInConverterProc(numberDataPackets, data);
}
OSStatus AudioDeviceMac::implDeviceIOProc(const AudioBufferList *inputData,
const AudioTimeStamp *inputTime,
AudioBufferList *outputData,
const AudioTimeStamp *outputTime)
{
OSStatus err = noErr;
UInt64 outputTimeNs = AudioConvertHostTimeToNanos(outputTime->mHostTime);
UInt64 nowNs = AudioConvertHostTimeToNanos(AudioGetCurrentHostTime());
if (!_twoDevices && _recording)
{
implInDeviceIOProc(inputData, inputTime);
}
// Check if we should close down audio device
// Double-checked locking optimization to remove locking overhead
if (_doStop)
{
_critSect.Enter();
if (_doStop)
{
if (_twoDevices || (!_recording && !_playing))
{
// In the case of a shared device, the single driving ioProc
// is stopped here
WEBRTC_CA_LOG_ERR(AudioDeviceStop(_outputDeviceID,
_deviceIOProcID));
WEBRTC_CA_LOG_WARN(AudioDeviceDestroyIOProcID(_outputDeviceID,
_deviceIOProcID));
if (err == noErr)
{
WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice,
_id, " Playout or shared device stopped");
}
}
_doStop = false;
_stopEvent.Set();
_critSect.Leave();
return 0;
}
_critSect.Leave();
}
if (!_playing)
{
// This can be the case when a shared device is capturing but not
// rendering. We allow the checks above before returning to avoid a
// timeout when capturing is stopped.
return 0;
}
DCHECK(_outStreamFormat.mBytesPerFrame != 0);
UInt32 size = outputData->mBuffers->mDataByteSize
/ _outStreamFormat.mBytesPerFrame;
// TODO(xians): signal an error somehow?
err = AudioConverterFillComplexBuffer(_renderConverter, outConverterProc,
this, &size, outputData, NULL);
if (err != noErr)
{
if (err == 1)
{
// This is our own error.
WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
" Error in AudioConverterFillComplexBuffer()");
return 1;
} else
{
logCAMsg(kTraceError, kTraceAudioDevice, _id,
"Error in AudioConverterFillComplexBuffer()",
(const char *) &err);
return 1;
}
}
PaRingBufferSize bufSizeSamples =
PaUtil_GetRingBufferReadAvailable(_paRenderBuffer);
int32_t renderDelayUs = static_cast<int32_t> (1e-3 * (outputTimeNs - nowNs)
+ 0.5);
renderDelayUs += static_cast<int32_t> ((1.0e6 * bufSizeSamples)
/ _outDesiredFormat.mChannelsPerFrame / _outDesiredFormat.mSampleRate
+ 0.5);
AtomicSet32(&_renderDelayUs, renderDelayUs);
return 0;
}
OSStatus AudioDeviceMac::implOutConverterProc(UInt32 *numberDataPackets,
AudioBufferList *data)
{
DCHECK(data->mNumberBuffers == 1);
PaRingBufferSize numSamples = *numberDataPackets
* _outDesiredFormat.mChannelsPerFrame;
data->mBuffers->mNumberChannels = _outDesiredFormat.mChannelsPerFrame;
// Always give the converter as much as it wants, zero padding as required.
data->mBuffers->mDataByteSize = *numberDataPackets
* _outDesiredFormat.mBytesPerPacket;
data->mBuffers->mData = _renderConvertData;
memset(_renderConvertData, 0, sizeof(_renderConvertData));
PaUtil_ReadRingBuffer(_paRenderBuffer, _renderConvertData, numSamples);
kern_return_t kernErr = semaphore_signal_all(_renderSemaphore);
if (kernErr != KERN_SUCCESS)
{
WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
" semaphore_signal_all() error: %d", kernErr);
return 1;
}
return 0;
}
OSStatus AudioDeviceMac::implInDeviceIOProc(const AudioBufferList *inputData,
const AudioTimeStamp *inputTime)
{
OSStatus err = noErr;
UInt64 inputTimeNs = AudioConvertHostTimeToNanos(inputTime->mHostTime);
UInt64 nowNs = AudioConvertHostTimeToNanos(AudioGetCurrentHostTime());
// Check if we should close down audio device
// Double-checked locking optimization to remove locking overhead
if (_doStopRec)
{
_critSect.Enter();
if (_doStopRec)
{
// This will be signalled only when a shared device is not in use.
WEBRTC_CA_LOG_ERR(AudioDeviceStop(_inputDeviceID, _inDeviceIOProcID));
WEBRTC_CA_LOG_WARN(AudioDeviceDestroyIOProcID(_inputDeviceID,
_inDeviceIOProcID));
if (err == noErr)
{
WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice,
_id, " Recording device stopped");
}
_doStopRec = false;
_stopEventRec.Set();
_critSect.Leave();
return 0;
}
_critSect.Leave();
}
if (!_recording)
{
// Allow above checks to avoid a timeout on stopping capture.
return 0;
}
PaRingBufferSize bufSizeSamples =
PaUtil_GetRingBufferReadAvailable(_paCaptureBuffer);
int32_t captureDelayUs = static_cast<int32_t> (1e-3 * (nowNs - inputTimeNs)
+ 0.5);
captureDelayUs
+= static_cast<int32_t> ((1.0e6 * bufSizeSamples)
/ _inStreamFormat.mChannelsPerFrame / _inStreamFormat.mSampleRate
+ 0.5);
AtomicSet32(&_captureDelayUs, captureDelayUs);
DCHECK(inputData->mNumberBuffers == 1);
PaRingBufferSize numSamples = inputData->mBuffers->mDataByteSize
* _inStreamFormat.mChannelsPerFrame / _inStreamFormat.mBytesPerPacket;
PaUtil_WriteRingBuffer(_paCaptureBuffer, inputData->mBuffers->mData,
numSamples);
kern_return_t kernErr = semaphore_signal_all(_captureSemaphore);
if (kernErr != KERN_SUCCESS)
{
WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
" semaphore_signal_all() error: %d", kernErr);
}
return err;
}
OSStatus AudioDeviceMac::implInConverterProc(UInt32 *numberDataPackets,
AudioBufferList *data)
{
DCHECK(data->mNumberBuffers == 1);
PaRingBufferSize numSamples = *numberDataPackets
* _inStreamFormat.mChannelsPerFrame;
while (PaUtil_GetRingBufferReadAvailable(_paCaptureBuffer) < numSamples)
{
mach_timespec_t timeout;
timeout.tv_sec = 0;
timeout.tv_nsec = TIMER_PERIOD_MS;
kern_return_t kernErr = semaphore_timedwait(_captureSemaphore, timeout);
if (kernErr == KERN_OPERATION_TIMED_OUT)
{
int32_t signal = AtomicGet32(&_captureDeviceIsAlive);
if (signal == 0)
{
// The capture device is no longer alive; stop the worker thread.
*numberDataPackets = 0;
return 1;
}
} else if (kernErr != KERN_SUCCESS)
{
WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
" semaphore_wait() error: %d", kernErr);
}
}
// Pass the read pointer directly to the converter to avoid a memcpy.
void* dummyPtr;
PaRingBufferSize dummySize;
PaUtil_GetRingBufferReadRegions(_paCaptureBuffer, numSamples,
&data->mBuffers->mData, &numSamples,
&dummyPtr, &dummySize);
PaUtil_AdvanceRingBufferReadIndex(_paCaptureBuffer, numSamples);
data->mBuffers->mNumberChannels = _inStreamFormat.mChannelsPerFrame;
*numberDataPackets = numSamples / _inStreamFormat.mChannelsPerFrame;
data->mBuffers->mDataByteSize = *numberDataPackets
* _inStreamFormat.mBytesPerPacket;
return 0;
}
bool AudioDeviceMac::RunRender(void* ptrThis)
{
return static_cast<AudioDeviceMac*> (ptrThis)->RenderWorkerThread();
}
bool AudioDeviceMac::RenderWorkerThread()
{
PaRingBufferSize numSamples = ENGINE_PLAY_BUF_SIZE_IN_SAMPLES
* _outDesiredFormat.mChannelsPerFrame;
while (PaUtil_GetRingBufferWriteAvailable(_paRenderBuffer)
- _renderDelayOffsetSamples < numSamples)
{
mach_timespec_t timeout;
timeout.tv_sec = 0;
timeout.tv_nsec = TIMER_PERIOD_MS;
kern_return_t kernErr = semaphore_timedwait(_renderSemaphore, timeout);
if (kernErr == KERN_OPERATION_TIMED_OUT)
{
int32_t signal = AtomicGet32(&_renderDeviceIsAlive);
if (signal == 0)
{
// The render device is no longer alive; stop the worker thread.
return false;
}
} else if (kernErr != KERN_SUCCESS)
{
WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
" semaphore_timedwait() error: %d", kernErr);
}
}
int8_t playBuffer[4 * ENGINE_PLAY_BUF_SIZE_IN_SAMPLES];
if (!_ptrAudioBuffer)
{
WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
" capture AudioBuffer is invalid");
return false;
}
// Ask for new PCM data to be played out using the AudioDeviceBuffer.
uint32_t nSamples =
_ptrAudioBuffer->RequestPlayoutData(ENGINE_PLAY_BUF_SIZE_IN_SAMPLES);
nSamples = _ptrAudioBuffer->GetPlayoutData(playBuffer);
if (nSamples != ENGINE_PLAY_BUF_SIZE_IN_SAMPLES)
{
WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
" invalid number of output samples(%d)", nSamples);
}
uint32_t nOutSamples = nSamples * _outDesiredFormat.mChannelsPerFrame;
SInt16 *pPlayBuffer = (SInt16 *) &playBuffer;
if (_macBookProPanRight && (_playChannels == 2))
{
// Mix entirely into the right channel and zero the left channel.
SInt32 sampleInt32 = 0;
for (uint32_t sampleIdx = 0; sampleIdx < nOutSamples; sampleIdx
+= 2)
{
sampleInt32 = pPlayBuffer[sampleIdx];
sampleInt32 += pPlayBuffer[sampleIdx + 1];
sampleInt32 /= 2;
if (sampleInt32 > 32767)
{
sampleInt32 = 32767;
} else if (sampleInt32 < -32768)
{
sampleInt32 = -32768;
}
pPlayBuffer[sampleIdx] = 0;
pPlayBuffer[sampleIdx + 1] = static_cast<SInt16> (sampleInt32);
}
}
PaUtil_WriteRingBuffer(_paRenderBuffer, pPlayBuffer, nOutSamples);
return true;
}
bool AudioDeviceMac::RunCapture(void* ptrThis)
{
return static_cast<AudioDeviceMac*> (ptrThis)->CaptureWorkerThread();
}
bool AudioDeviceMac::CaptureWorkerThread()
{
OSStatus err = noErr;
UInt32 noRecSamples = ENGINE_REC_BUF_SIZE_IN_SAMPLES
* _inDesiredFormat.mChannelsPerFrame;
SInt16 recordBuffer[noRecSamples];
UInt32 size = ENGINE_REC_BUF_SIZE_IN_SAMPLES;
AudioBufferList engineBuffer;
engineBuffer.mNumberBuffers = 1; // Interleaved channels.
engineBuffer.mBuffers->mNumberChannels = _inDesiredFormat.mChannelsPerFrame;
engineBuffer.mBuffers->mDataByteSize = _inDesiredFormat.mBytesPerPacket
* noRecSamples;
engineBuffer.mBuffers->mData = recordBuffer;
err = AudioConverterFillComplexBuffer(_captureConverter, inConverterProc,
this, &size, &engineBuffer, NULL);
if (err != noErr)
{
if (err == 1)
{
// This is our own error.
return false;
} else
{
logCAMsg(kTraceError, kTraceAudioDevice, _id,
"Error in AudioConverterFillComplexBuffer()",
(const char *) &err);
return false;
}
}
// TODO(xians): what if the returned size is incorrect?
if (size == ENGINE_REC_BUF_SIZE_IN_SAMPLES)
{
uint32_t currentMicLevel(0);
uint32_t newMicLevel(0);
int32_t msecOnPlaySide;
int32_t msecOnRecordSide;
int32_t captureDelayUs = AtomicGet32(&_captureDelayUs);
int32_t renderDelayUs = AtomicGet32(&_renderDelayUs);
msecOnPlaySide = static_cast<int32_t> (1e-3 * (renderDelayUs +
_renderLatencyUs) + 0.5);
msecOnRecordSide = static_cast<int32_t> (1e-3 * (captureDelayUs +
_captureLatencyUs) +
0.5);
if (!_ptrAudioBuffer)
{
WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
" capture AudioBuffer is invalid");
return false;
}
// store the recorded buffer (no action will be taken if the
// #recorded samples is not a full buffer)
_ptrAudioBuffer->SetRecordedBuffer((int8_t*) &recordBuffer,
(uint32_t) size);
if (AGC())
{
// store current mic level in the audio buffer if AGC is enabled
if (MicrophoneVolume(currentMicLevel) == 0)
{
// this call does not affect the actual microphone volume
_ptrAudioBuffer->SetCurrentMicLevel(currentMicLevel);
}
}
_ptrAudioBuffer->SetVQEData(msecOnPlaySide, msecOnRecordSide, 0);
_ptrAudioBuffer->SetTypingStatus(KeyPressed());
// deliver recorded samples at specified sample rate, mic level etc.
// to the observer using callback
_ptrAudioBuffer->DeliverRecordedData();
if (AGC())
{
newMicLevel = _ptrAudioBuffer->NewMicLevel();
if (newMicLevel != 0)
{
// The VQE will only deliver non-zero microphone levels when
// a change is needed.
// Set this new mic level (received from the observer as return
// value in the callback).
WEBRTC_TRACE(kTraceStream, kTraceAudioDevice,
_id, " AGC change of volume: old=%u => new=%u",
currentMicLevel, newMicLevel);
if (SetMicrophoneVolume(newMicLevel) == -1)
{
WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
" the required modification of the microphone "
"volume failed");
}
}
}
}
return true;
}
bool AudioDeviceMac::KeyPressed() {
bool key_down = false;
// Loop through all Mac virtual key constant values.
for (unsigned int key_index = 0;
key_index < arraysize(prev_key_state_);
++key_index) {
bool keyState = CGEventSourceKeyState(
kCGEventSourceStateHIDSystemState,
key_index);
// A false -> true change in keymap means a key is pressed.
key_down |= (keyState && !prev_key_state_[key_index]);
// Save current state.
prev_key_state_[key_index] = keyState;
}
return key_down;
}
} // namespace webrtc
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