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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.

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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
This commit is contained in:
braveyao@webrtc.org
2015-03-21 01:05:56 +00:00
parent 4553941d32
commit 346a64b9b5
2 changed files with 146 additions and 127 deletions
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269
webrtc/modules/audio_device/mac/audio_device_mac.cc
View File

@ -1381,128 +1381,40 @@ int32_t AudioDeviceMac::InitPlayout()
logCAMsg(kTraceInfo, kTraceAudioDevice, _id, "mFormatID",
(const char *) &_outStreamFormat.mFormatID);
// Our preferred format to work with
_outDesiredFormat.mSampleRate = N_PLAY_SAMPLES_PER_SEC;
if (_outStreamFormat.mChannelsPerFrame >= 2 && (_playChannels == 2))
{
_outDesiredFormat.mChannelsPerFrame = 2;
} else
// Our preferred format to work with.
if (_outStreamFormat.mChannelsPerFrame < 2)
{
// Disable stereo playout when we only have one channel on the device.
_outDesiredFormat.mChannelsPerFrame = 1;
_playChannels = 1;
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
"Stereo playout unavailable on this device");
}
WEBRTC_CA_RETURN_ON_ERR(SetDesiredPlayoutFormat());
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);
_outDesiredFormat.mFramesPerPacket = 1; // In uncompressed audio,
// a packet is one frame.
_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;
WEBRTC_CA_RETURN_ON_ERR(AudioConverterNew(&_outDesiredFormat, &_outStreamFormat,
&_renderConverter));
// First try to set buffer size to desired value (_playBufDelayFixed)
UInt32 bufByteCount = (UInt32)((_outStreamFormat.mSampleRate / 1000.0)
* _playBufDelayFixed * _outStreamFormat.mChannelsPerFrame
* sizeof(Float32));
if (_outStreamFormat.mFramesPerPacket != 0)
{
if (bufByteCount % _outStreamFormat.mFramesPerPacket != 0)
{
bufByteCount = ((UInt32)(bufByteCount
/ _outStreamFormat.mFramesPerPacket) + 1)
* _outStreamFormat.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(_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 = (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 += (uint32_t) ((1.0e6 * latency)
/ _outStreamFormat.mSampleRate);
// Listen for format changes
// Listen for format changes.
propertyAddress.mSelector = kAudioDevicePropertyStreamFormat;
WEBRTC_CA_RETURN_ON_ERR(AudioObjectAddPropertyListener(_outputDeviceID,
&propertyAddress, &objectListenerProc, this));
&propertyAddress,
&objectListenerProc,
this));
// Listen for processor overloads
// Listen for processor overloads.
propertyAddress.mSelector = kAudioDeviceProcessorOverload;
WEBRTC_CA_LOG_WARN(AudioObjectAddPropertyListener(_outputDeviceID,
&propertyAddress, &objectListenerProc, this));
&propertyAddress,
&objectListenerProc,
this));
if (_twoDevices || !_recIsInitialized)
{
WEBRTC_CA_RETURN_ON_ERR(AudioDeviceCreateIOProcID(_outputDeviceID,
deviceIOProc, this, &_deviceIOProcID));
deviceIOProc,
this,
&_deviceIOProcID));
}
// Mark playout side as initialized
_playIsInitialized = true;
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
" initial playout status: _renderDelayOffsetSamples=%d,"
" _renderDelayUs=%d, _renderLatencyUs=%d",
_renderDelayOffsetSamples, _renderDelayUs, _renderLatencyUs);
return 0;
}
@ -2423,6 +2335,131 @@ int32_t AudioDeviceMac::InitDevice(const uint16_t userDeviceIndex,
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,
@ -2635,37 +2672,15 @@ int32_t AudioDeviceMac::HandleStreamFormatChange(
{
memcpy(&_outStreamFormat, &streamFormat, sizeof(streamFormat));
if (_outStreamFormat.mChannelsPerFrame >= 2 && (_playChannels == 2))
// Our preferred format to work with
if (_outStreamFormat.mChannelsPerFrame < 2)
{
_outDesiredFormat.mChannelsPerFrame = 2;
} else
{
// Disable stereo playout when we only have one channel on the device.
_outDesiredFormat.mChannelsPerFrame = 1;
_playChannels = 1;
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
"Stereo playout unavailable on this device");
}
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;
// Recreate the converter with the new format
// TODO(xians): make this thread safe
WEBRTC_CA_RETURN_ON_ERR(AudioConverterDispose(_renderConverter));
WEBRTC_CA_RETURN_ON_ERR(AudioConverterNew(&_outDesiredFormat, &streamFormat,
&_renderConverter));
WEBRTC_CA_RETURN_ON_ERR(SetDesiredPlayoutFormat());
}
return 0;
}

4
webrtc/modules/audio_device/mac/audio_device_mac.h
View File

@ -200,6 +200,10 @@ private:
int32_t InitDevice(uint16_t userDeviceIndex,
AudioDeviceID& deviceId, bool isInput);
// Always work with our preferred playout format inside VoE.
// Then convert the output to the OS setting using an AudioConverter.
OSStatus SetDesiredPlayoutFormat();
static OSStatus
objectListenerProc(AudioObjectID objectId, UInt32 numberAddresses,
const AudioObjectPropertyAddress addresses[],