This is a change in the iOS audio device to use VoiceProcessingIO API instead of RemoteIO. This way we don't need to use WebRTC EC and NS because it happens on the device hardware.
Review URL: https://webrtc-codereview.appspot.com/1061007
Patch from Gil Osher <gil.osher@vonage.com>.
git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@3437 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/modules/audio_device/ios/audio_device_ios.cc b/modules/audio_device/ios/audio_device_ios.cc
index 264556c..1381859 100644
--- a/modules/audio_device/ios/audio_device_ios.cc
+++ b/modules/audio_device/ios/audio_device_ios.cc
@@ -23,7 +23,7 @@
_captureWorkerThread(NULL),
_captureWorkerThreadId(0),
_id(id),
- _auRemoteIO(NULL),
+ _auVoiceProcessing(NULL),
_initialized(false),
_isShutDown(false),
_recording(false),
@@ -127,17 +127,6 @@
WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice,
_id, "Thread already created");
}
-
- // Set preferred hardware sample rate to 16 kHz
- Float64 sampleRate(16000.0);
- OSStatus result = AudioSessionSetProperty(
- kAudioSessionProperty_PreferredHardwareSampleRate,
- sizeof(sampleRate), &sampleRate);
- if (0 != result) {
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- "Could not set preferred sample rate (result=%d)", result);
- }
-
_playWarning = 0;
_playError = 0;
_recWarning = 0;
@@ -168,7 +157,7 @@
_captureWorkerThread = NULL;
}
- // Shut down AU Remote IO
+ // Shut down Audio Unit
ShutdownPlayOrRecord();
_isShutDown = true;
@@ -941,13 +930,13 @@
_recError = 0;
if (!_playing) {
- // Start AU Remote IO
+ // Start Audio Unit
WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
- " Starting AU Remote IO");
- OSStatus result = AudioOutputUnitStart(_auRemoteIO);
+ " Starting Audio Unit");
+ OSStatus result = AudioOutputUnitStart(_auVoiceProcessing);
if (0 != result) {
WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice, _id,
- " Error starting AU Remote IO (result=%d)", result);
+ " Error starting Audio Unit (result=%d)", result);
return -1;
}
}
@@ -991,7 +980,7 @@
// This lock is (among other things) needed to avoid concurrency issues
// with capture thread
- // shutting down AU Remote IO
+ // shutting down Audio Unit
CriticalSectionScoped lock(&_critSect);
if (!_playIsInitialized) {
@@ -1016,13 +1005,13 @@
_playError = 0;
if (!_recording) {
- // Start AU Remote IO
+ // Start Audio Unit
WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
- " Starting AU Remote IO");
- OSStatus result = AudioOutputUnitStart(_auRemoteIO);
+ " Starting Audio Unit");
+ OSStatus result = AudioOutputUnitStart(_auVoiceProcessing);
if (0 != result) {
WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice, _id,
- " Error starting AU Remote IO (result=%d)", result);
+ " Error starting Audio Unit (result=%d)", result);
return -1;
}
}
@@ -1196,7 +1185,7 @@
OSStatus result = -1;
// Check if already initialized
- if (NULL != _auRemoteIO) {
+ if (NULL != _auVoiceProcessing) {
// We already have initialized before and created any of the audio unit,
// check that all exist
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
@@ -1205,12 +1194,12 @@
return 0;
}
- // Create AU Remote IO
+ // Create Voice Processing Audio Unit
AudioComponentDescription desc;
AudioComponent comp;
desc.componentType = kAudioUnitType_Output;
- desc.componentSubType = kAudioUnitSubType_RemoteIO;
+ desc.componentSubType = kAudioUnitSubType_VoiceProcessingIO;
desc.componentManufacturer = kAudioUnitManufacturer_Apple;
desc.componentFlags = 0;
desc.componentFlagsMask = 0;
@@ -1218,23 +1207,38 @@
comp = AudioComponentFindNext(NULL, &desc);
if (NULL == comp) {
WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " Could not find audio component for AU Remote IO");
+ " Could not find audio component for Audio Unit");
return -1;
}
- result = AudioComponentInstanceNew(comp, &_auRemoteIO);
+ result = AudioComponentInstanceNew(comp, &_auVoiceProcessing);
if (0 != result) {
WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " Could not create AU Remote IO instance (result=%d)",
+ " Could not create Audio Unit instance (result=%d)",
result);
return -1;
}
- //////////////////////
- // Setup AU remote IO
+ // Set preferred hardware sample rate to 16 kHz
+ Float64 sampleRate(16000.0);
+ result = AudioSessionSetProperty(
+ kAudioSessionProperty_PreferredHardwareSampleRate,
+ sizeof(sampleRate), &sampleRate);
+ if (0 != result) {
+ WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
+ "Could not set preferred sample rate (result=%d)", result);
+ }
- // Note: For AU Remote IO element 0 is output bus, element 1 is input bus
- // for global scope element is irrelevant (always use element 0)
+ WebRtc_UWord32 voiceChat = kAudioSessionMode_VoiceChat;
+ AudioSessionSetProperty(kAudioSessionProperty_Mode,
+ sizeof(voiceChat), &voiceChat);
+
+ //////////////////////
+ // Setup Voice Processing Audio Unit
+
+ // Note: For Signal Processing AU element 0 is output bus, element 1 is
+ // input bus for global scope element is irrelevant (always use
+ // element 0)
// Enable IO on both elements
@@ -1243,7 +1247,7 @@
// todo: Log info about setup.
UInt32 enableIO = 1;
- result = AudioUnitSetProperty(_auRemoteIO,
+ result = AudioUnitSetProperty(_auVoiceProcessing,
kAudioOutputUnitProperty_EnableIO,
kAudioUnitScope_Input,
1, // input bus
@@ -1254,7 +1258,7 @@
" Could not enable IO on input (result=%d)", result);
}
- result = AudioUnitSetProperty(_auRemoteIO,
+ result = AudioUnitSetProperty(_auVoiceProcessing,
kAudioOutputUnitProperty_EnableIO,
kAudioUnitScope_Output,
0, // output bus
@@ -1268,7 +1272,7 @@
// Disable AU buffer allocation for the recorder, we allocate our own
UInt32 flag = 0;
result = AudioUnitSetProperty(
- _auRemoteIO, kAudioUnitProperty_ShouldAllocateBuffer,
+ _auVoiceProcessing, kAudioUnitProperty_ShouldAllocateBuffer,
kAudioUnitScope_Output, 1, &flag, sizeof(flag));
if (0 != result) {
WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
@@ -1277,28 +1281,53 @@
// Should work anyway
}
- // Initialize here already to be able to get/set stream properties.
- result = AudioUnitInitialize(_auRemoteIO);
+ // Set recording callback
+ AURenderCallbackStruct auCbS;
+ memset(&auCbS, 0, sizeof(auCbS));
+ auCbS.inputProc = RecordProcess;
+ auCbS.inputProcRefCon = this;
+ result = AudioUnitSetProperty(_auVoiceProcessing,
+ kAudioOutputUnitProperty_SetInputCallback,
+ kAudioUnitScope_Global, 1,
+ &auCbS, sizeof(auCbS));
if (0 != result) {
WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " Could not init AU Remote IO (result=%d)", result);
+ " Could not set record callback for Audio Unit (result=%d)",
+ result);
+ }
+
+ // Set playout callback
+ memset(&auCbS, 0, sizeof(auCbS));
+ auCbS.inputProc = PlayoutProcess;
+ auCbS.inputProcRefCon = this;
+ result = AudioUnitSetProperty(_auVoiceProcessing,
+ kAudioUnitProperty_SetRenderCallback,
+ kAudioUnitScope_Global, 0,
+ &auCbS, sizeof(auCbS));
+ if (0 != result) {
+ WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
+ " Could not set play callback for Audio Unit (result=%d)",
+ result);
}
// Get stream format for out/0
AudioStreamBasicDescription playoutDesc;
UInt32 size = sizeof(playoutDesc);
- result = AudioUnitGetProperty(_auRemoteIO, kAudioUnitProperty_StreamFormat,
+ result = AudioUnitGetProperty(_auVoiceProcessing,
+ kAudioUnitProperty_StreamFormat,
kAudioUnitScope_Output, 0, &playoutDesc,
&size);
if (0 != result) {
WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " Could not get stream format AU Remote IO out/0 (result=%d)",
+ " Could not get stream format Audio Unit out/0 (result=%d)",
result);
}
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- " AU Remote IO playout opened in sampling rate %f",
+ " Audio Unit playout opened in sampling rate %f",
playoutDesc.mSampleRate);
+ playoutDesc.mSampleRate = sampleRate;
+
// Store the sampling frequency to use towards the Audio Device Buffer
// todo: Add 48 kHz (increase buffer sizes). Other fs?
if ((playoutDesc.mSampleRate > 44090.0)
@@ -1313,7 +1342,7 @@
} else {
_adbSampFreq = 0;
WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " AU Remote IO out/0 opened in unknown sampling rate (%f)",
+ " Audio Unit out/0 opened in unknown sampling rate (%f)",
playoutDesc.mSampleRate);
// todo: We should bail out here.
}
@@ -1341,37 +1370,32 @@
playoutDesc.mBytesPerFrame = 2;
playoutDesc.mChannelsPerFrame = 1;
playoutDesc.mBitsPerChannel = 16;
- result = AudioUnitSetProperty(_auRemoteIO, kAudioUnitProperty_StreamFormat,
+ result = AudioUnitSetProperty(_auVoiceProcessing,
+ kAudioUnitProperty_StreamFormat,
kAudioUnitScope_Input, 0, &playoutDesc, size);
if (0 != result) {
WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " Could not set stream format AU Remote IO in/0 (result=%d)",
+ " Could not set stream format Audio Unit in/0 (result=%d)",
result);
}
// Get stream format for in/1
AudioStreamBasicDescription recordingDesc;
size = sizeof(recordingDesc);
- result = AudioUnitGetProperty(_auRemoteIO, kAudioUnitProperty_StreamFormat,
+ result = AudioUnitGetProperty(_auVoiceProcessing,
+ kAudioUnitProperty_StreamFormat,
kAudioUnitScope_Input, 1, &recordingDesc,
&size);
if (0 != result) {
WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " Could not get stream format AU Remote IO in/1 (result=%d)",
+ " Could not get stream format Audio Unit in/1 (result=%d)",
result);
}
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- " AU Remote IO recording opened in sampling rate %f",
+ " Audio Unit recording opened in sampling rate %f",
recordingDesc.mSampleRate);
- if (static_cast<int>(playoutDesc.mSampleRate)
- != static_cast<int>(recordingDesc.mSampleRate)) {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " AU Remote IO recording and playout opened " \
- "in different sampling rates");
- // todo: Bail out if rec and play sampling rates are not the same?
- // Add handling of different sampling rates?
- }
+ recordingDesc.mSampleRate = sampleRate;
// Set stream format for out/1 (use same sampling frequency as for in/1)
recordingDesc.mFormatFlags = kLinearPCMFormatFlagIsSignedInteger
@@ -1383,54 +1407,36 @@
recordingDesc.mBytesPerFrame = 2;
recordingDesc.mChannelsPerFrame = 1;
recordingDesc.mBitsPerChannel = 16;
- result = AudioUnitSetProperty(_auRemoteIO, kAudioUnitProperty_StreamFormat,
+ result = AudioUnitSetProperty(_auVoiceProcessing,
+ kAudioUnitProperty_StreamFormat,
kAudioUnitScope_Output, 1, &recordingDesc,
size);
if (0 != result) {
WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " Could not set stream format AU Remote IO out/1 (result=%d)",
+ " Could not set stream format Audio Unit out/1 (result=%d)",
result);
}
- // Set recording callback
- AURenderCallbackStruct auCbS;
- memset(&auCbS, 0, sizeof(auCbS));
- auCbS.inputProc = RecordProcess;
- auCbS.inputProcRefCon = this;
- result = AudioUnitSetProperty(_auRemoteIO,
- kAudioOutputUnitProperty_SetInputCallback, kAudioUnitScope_Global, 1,
- &auCbS, sizeof(auCbS));
+ // Initialize here already to be able to get/set stream properties.
+ result = AudioUnitInitialize(_auVoiceProcessing);
if (0 != result) {
WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " Could not set record callback for AU Remote IO (result=%d)",
- result);
- }
-
- // Set playout callback
- memset(&auCbS, 0, sizeof(auCbS));
- auCbS.inputProc = PlayoutProcess;
- auCbS.inputProcRefCon = this;
- result = AudioUnitSetProperty(_auRemoteIO,
- kAudioUnitProperty_SetRenderCallback, kAudioUnitScope_Global, 0,
- &auCbS, sizeof(auCbS));
- if (0 != result) {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " Could not set play callback for AU Remote IO (result=%d)",
- result);
+ " Could not init Audio Unit (result=%d)", result);
}
// Get hardware sample rate for logging (see if we get what we asked for)
- Float64 sampleRate(0.0);
- size = sizeof(sampleRate);
+ Float64 hardwareSampleRate = 0.0;
+ size = sizeof(hardwareSampleRate);
result = AudioSessionGetProperty(
- kAudioSessionProperty_CurrentHardwareSampleRate, &size, &sampleRate);
+ kAudioSessionProperty_CurrentHardwareSampleRate, &size,
+ &hardwareSampleRate);
if (0 != result) {
WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
" Could not get current HW sample rate (result=%d)", result);
}
WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
" Current HW sample rate is %f, ADB sample rate is %d",
- sampleRate, _adbSampFreq);
+ hardwareSampleRate, _adbSampFreq);
return 0;
}
@@ -1440,23 +1446,18 @@
// Close and delete AU
OSStatus result = -1;
- if (NULL != _auRemoteIO) {
- result = AudioOutputUnitStop(_auRemoteIO);
+ if (NULL != _auVoiceProcessing) {
+ result = AudioOutputUnitStop(_auVoiceProcessing);
if (0 != result) {
WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " Error stopping AU Remote IO (result=%d)", result);
+ " Error stopping Audio Unit (result=%d)", result);
}
- result = AudioUnitUninitialize(_auRemoteIO);
+ result = AudioComponentInstanceDispose(_auVoiceProcessing);
if (0 != result) {
WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " Error uninitializing AU Remote IO (result=%d)", result);
+ " Error disposing Audio Unit (result=%d)", result);
}
- result = AudioComponentInstanceDispose(_auRemoteIO);
- if (0 != result) {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " Error disposing AU Remote IO (result=%d)", result);
- }
- _auRemoteIO = NULL;
+ _auVoiceProcessing = NULL;
}
return 0;
@@ -1502,7 +1503,8 @@
abList.mBuffers[0].mNumberChannels = 1;
// Get data from mic
- OSStatus res = AudioUnitRender(_auRemoteIO, ioActionFlags, inTimeStamp,
+ OSStatus res = AudioUnitRender(_auVoiceProcessing,
+ ioActionFlags, inTimeStamp,
inBusNumber, inNumberFrames, &abList);
if (res != 0) {
WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
@@ -1740,7 +1742,7 @@
// AU latency
Float64 f64(0);
size = sizeof(f64);
- result = AudioUnitGetProperty(_auRemoteIO,
+ result = AudioUnitGetProperty(_auVoiceProcessing,
kAudioUnitProperty_Latency, kAudioUnitScope_Global, 0, &f64, &size);
if (0 != result) {
WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
@@ -1790,7 +1792,8 @@
// AU latency
Float64 f64(0);
size = sizeof(f64);
- result = AudioUnitGetProperty(_auRemoteIO, kAudioUnitProperty_Latency,
+ result = AudioUnitGetProperty(_auVoiceProcessing,
+ kAudioUnitProperty_Latency,
kAudioUnitScope_Global, 0, &f64, &size);
if (0 != result) {
WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
diff --git a/modules/audio_device/ios/audio_device_ios.h b/modules/audio_device/ios/audio_device_ios.h
index d494603..dc32482 100644
--- a/modules/audio_device/ios/audio_device_ios.h
+++ b/modules/audio_device/ios/audio_device_ios.h
@@ -227,7 +227,7 @@
WebRtc_Word32 _id;
- AudioUnit _auRemoteIO;
+ AudioUnit _auVoiceProcessing;
private:
bool _initialized;
