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Allow audio devices with up to 64 channels on Mac.

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Does not increase memory requirements. Adds an additional check to ensure
configurations requiring more memory per IO block than the input ring buffer
contains are rejected.

BUG=1904
TESTED=Using Soundflower (64 channels) at 48 kHz as input gives good quality.
Selecting a higher sample rate (96 kHz), which would otherwise give choppy
audio, instead results in an error.

R=henrika@webrtc.org, xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1628004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4198 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
andrew@webrtc.org 2013-06-07 17:56:50 +00:00
parent 1064cf06b0
commit 6f69eb78dd
2 changed files with 66 additions and 56 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

79
webrtc/modules/audio_device/mac/audio_device_mac.cc
View File

@ -373,10 +373,10 @@ int32_t AudioDeviceMac::Init()
return -1; return -1;
} }
// Setting RunLoop to NULL here instructs HAL to manage its own thread for // 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 // notifications. This was the default behaviour on OS X 10.5 and earlier,
// must be explicitly specified. HAL would otherwise try to use the main thread to // but now must be explicitly specified. HAL would otherwise try to use the
// issue notifications. // main thread to issue notifications.
AudioObjectPropertyAddress propertyAddress = { AudioObjectPropertyAddress propertyAddress = {
kAudioHardwarePropertyRunLoop, kAudioHardwarePropertyRunLoop,
kAudioObjectPropertyScopeGlobal, kAudioObjectPropertyScopeGlobal,
@ -495,8 +495,8 @@ int32_t AudioDeviceMac::SpeakerIsAvailable(bool& available)
return 0; return 0;
} }
// Given that InitSpeaker was successful, we know that a valid speaker exists // Given that InitSpeaker was successful, we know that a valid speaker
// // exists.
available = true; available = true;
// Close the initialized output mixer // Close the initialized output mixer
@ -554,8 +554,8 @@ int32_t AudioDeviceMac::MicrophoneIsAvailable(bool& available)
return 0; return 0;
} }
// Given that InitMicrophone was successful, we know that a valid microphone exists // Given that InitMicrophone was successful, we know that a valid microphone
// // exists.
available = true; available = true;
// Close the initialized input mixer // Close the initialized input mixer
@ -1363,13 +1363,13 @@ int32_t AudioDeviceMac::InitPlayout()
_id, "MacBook Pro not using internal speakers"); _id, "MacBook Pro not using internal speakers");
} }
// Add a listener to determine if the status changes. // Add a listener to determine if the status changes.
WEBRTC_CA_LOG_WARN(AudioObjectAddPropertyListener(_outputDeviceID, WEBRTC_CA_LOG_WARN(AudioObjectAddPropertyListener(_outputDeviceID,
&propertyAddress, &objectListenerProc, this)); &propertyAddress, &objectListenerProc, this));
} }
} }
// Get current stream description // Get current stream description
propertyAddress.mSelector = kAudioDevicePropertyStreamFormat; propertyAddress.mSelector = kAudioDevicePropertyStreamFormat;
memset(&_outStreamFormat, 0, sizeof(_outStreamFormat)); memset(&_outStreamFormat, 0, sizeof(_outStreamFormat));
size = sizeof(_outStreamFormat); size = sizeof(_outStreamFormat);
@ -1387,8 +1387,8 @@ int32_t AudioDeviceMac::InitPlayout()
if (_outStreamFormat.mChannelsPerFrame > N_DEVICE_CHANNELS) if (_outStreamFormat.mChannelsPerFrame > N_DEVICE_CHANNELS)
{ {
WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
"Too many channels on device -> mChannelsPerFrame = %d", "Too many channels on output device (mChannelsPerFrame = %d)",
_outStreamFormat.mChannelsPerFrame); _outStreamFormat.mChannelsPerFrame);
return -1; return -1;
} }
@ -1415,13 +1415,12 @@ int32_t AudioDeviceMac::InitPlayout()
_outStreamFormat.mBytesPerFrame, _outStreamFormat.mBytesPerFrame,
_outStreamFormat.mBitsPerChannel); _outStreamFormat.mBitsPerChannel);
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
"mFormatFlags = %u, mChannelsPerFrame = %u", "mFormatFlags = %u",
_outStreamFormat.mFormatFlags, _outStreamFormat.mFormatFlags);
_outStreamFormat.mChannelsPerFrame);
logCAMsg(kTraceInfo, kTraceAudioDevice, _id, "mFormatID", logCAMsg(kTraceInfo, kTraceAudioDevice, _id, "mFormatID",
(const char *) &_outStreamFormat.mFormatID); (const char *) &_outStreamFormat.mFormatID);
// Our preferred format to work with // Our preferred format to work with
_outDesiredFormat.mSampleRate = N_PLAY_SAMPLES_PER_SEC; _outDesiredFormat.mSampleRate = N_PLAY_SAMPLES_PER_SEC;
if (_outStreamFormat.mChannelsPerFrame >= 2 && (_playChannels == 2)) if (_outStreamFormat.mChannelsPerFrame >= 2 && (_playChannels == 2))
{ {
@ -1447,7 +1446,7 @@ int32_t AudioDeviceMac::InitPlayout()
_outDesiredFormat.mBytesPerPacket = _outDesiredFormat.mChannelsPerFrame _outDesiredFormat.mBytesPerPacket = _outDesiredFormat.mChannelsPerFrame
* sizeof(SInt16); * sizeof(SInt16);
_outDesiredFormat.mFramesPerPacket = 1; // In uncompressed audio, _outDesiredFormat.mFramesPerPacket = 1; // In uncompressed audio,
// a packet is one frame. // a packet is one frame.
_outDesiredFormat.mBytesPerFrame = _outDesiredFormat.mChannelsPerFrame _outDesiredFormat.mBytesPerFrame = _outDesiredFormat.mChannelsPerFrame
* sizeof(SInt16); * sizeof(SInt16);
@ -1595,7 +1594,7 @@ int32_t AudioDeviceMac::InitRecording()
_captureDeviceIsAlive = 1; _captureDeviceIsAlive = 1;
_doStopRec = false; _doStopRec = false;
// Get current stream description // Get current stream description
AudioObjectPropertyAddress AudioObjectPropertyAddress
propertyAddress = { kAudioDevicePropertyStreamFormat, propertyAddress = { kAudioDevicePropertyStreamFormat,
kAudioDevicePropertyScopeInput, 0 }; kAudioDevicePropertyScopeInput, 0 };
@ -1615,8 +1614,18 @@ int32_t AudioDeviceMac::InitRecording()
if (_inStreamFormat.mChannelsPerFrame > N_DEVICE_CHANNELS) if (_inStreamFormat.mChannelsPerFrame > N_DEVICE_CHANNELS)
{ {
WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
", Too many channels on device (mChannelsPerFrame = %d)", "Too many channels on input device (mChannelsPerFrame = %d)",
_inStreamFormat.mChannelsPerFrame); _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; return -1;
} }
@ -1634,9 +1643,8 @@ int32_t AudioDeviceMac::InitRecording()
_inStreamFormat.mBytesPerFrame, _inStreamFormat.mBytesPerFrame,
_inStreamFormat.mBitsPerChannel); _inStreamFormat.mBitsPerChannel);
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
" mFormatFlags = %u, mChannelsPerFrame = %u", " mFormatFlags = %u",
_inStreamFormat.mFormatFlags, _inStreamFormat.mFormatFlags);
_inStreamFormat.mChannelsPerFrame);
logCAMsg(kTraceInfo, kTraceAudioDevice, _id, "mFormatID", logCAMsg(kTraceInfo, kTraceAudioDevice, _id, "mFormatID",
(const char *) &_inStreamFormat.mFormatID); (const char *) &_inStreamFormat.mFormatID);
@ -1931,7 +1939,7 @@ bool AudioDeviceMac::PlayoutIsInitialized() const
int32_t AudioDeviceMac::StartPlayout() int32_t AudioDeviceMac::StartPlayout()
{ {
CriticalSectionScoped lock(&_critSect); CriticalSectionScoped lock(&_critSect);
if (!_playIsInitialized) if (!_playIsInitialized)
@ -2283,7 +2291,7 @@ AudioDeviceMac::GetNumberDevices(const AudioObjectPropertyScope scope,
return -1; return -1;
} }
// Happy ending // Happy ending
if (deviceIds) if (deviceIds)
{ {
free(deviceIds); free(deviceIds);
@ -2435,7 +2443,7 @@ int32_t AudioDeviceMac::InitDevice(const uint16_t userDeviceIndex,
} }
// Obtain device name and manufacturer for logging. // Obtain device name and manufacturer for logging.
// Also use this as a test to ensure a user-set device ID is valid. // Also use this as a test to ensure a user-set device ID is valid.
char devName[128]; char devName[128];
char devManf[128]; char devManf[128];
memset(devName, 0, sizeof(devName)); memset(devName, 0, sizeof(devName));
@ -2488,7 +2496,7 @@ OSStatus AudioDeviceMac::implObjectListenerProc(
{ {
WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id, WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
"AudioDeviceMac::implObjectListenerProc()"); "AudioDeviceMac::implObjectListenerProc()");
for (UInt32 i = 0; i < numberAddresses; i++) for (UInt32 i = 0; i < numberAddresses; i++)
{ {
if (addresses[i].mSelector == kAudioHardwarePropertyDevices) if (addresses[i].mSelector == kAudioHardwarePropertyDevices)
@ -2545,7 +2553,7 @@ int32_t AudioDeviceMac::HandleDeviceChange()
logCAMsg(kTraceError, kTraceAudioDevice, _id, logCAMsg(kTraceError, kTraceAudioDevice, _id,
"Error in AudioDeviceGetProperty()", (const char*) &err); "Error in AudioDeviceGetProperty()", (const char*) &err);
return -1; return -1;
} }
} }
if (SpeakerIsInitialized()) if (SpeakerIsInitialized())
@ -2629,13 +2637,24 @@ int32_t AudioDeviceMac::HandleStreamFormatChange(
"mBytesPerFrame = %u, mBitsPerChannel = %u", "mBytesPerFrame = %u, mBitsPerChannel = %u",
streamFormat.mBytesPerFrame, streamFormat.mBitsPerChannel); streamFormat.mBytesPerFrame, streamFormat.mBitsPerChannel);
WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
"mFormatFlags = %u, mChannelsPerFrame = %u", "mFormatFlags = %u",
streamFormat.mFormatFlags, streamFormat.mChannelsPerFrame); streamFormat.mFormatFlags);
logCAMsg(kTraceInfo, kTraceAudioDevice, _id, "mFormatID", logCAMsg(kTraceInfo, kTraceAudioDevice, _id, "mFormatID",
(const char *) &streamFormat.mFormatID); (const char *) &streamFormat.mFormatID);
if (propertyAddress.mScope == kAudioDevicePropertyScopeInput) 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)); memcpy(&_inStreamFormat, &streamFormat, sizeof(streamFormat));
if (_inStreamFormat.mChannelsPerFrame >= 2 && (_recChannels == 2)) if (_inStreamFormat.mChannelsPerFrame >= 2 && (_recChannels == 2))

43
webrtc/modules/audio_device/mac/audio_device_mac.h
View File

@ -26,35 +26,26 @@ namespace webrtc
class EventWrapper; class EventWrapper;
class ThreadWrapper; class ThreadWrapper;
const uint32_t N_REC_SAMPLES_PER_SEC = 48000; const int N_REC_SAMPLES_PER_SEC = 48000;
const uint32_t N_PLAY_SAMPLES_PER_SEC = 48000; const int N_PLAY_SAMPLES_PER_SEC = 48000;
const uint32_t N_REC_CHANNELS = 1; // default is mono recording const int N_REC_CHANNELS = 1; // default is mono recording
const uint32_t N_PLAY_CHANNELS = 2; // default is stereo playout const int N_PLAY_CHANNELS = 2; // default is stereo playout
const uint32_t N_DEVICE_CHANNELS = 8; const int N_DEVICE_CHANNELS = 64;
const uint32_t ENGINE_REC_BUF_SIZE_IN_SAMPLES = (N_REC_SAMPLES_PER_SEC / 100); const int ENGINE_REC_BUF_SIZE_IN_SAMPLES = (N_REC_SAMPLES_PER_SEC / 100);
const uint32_t ENGINE_PLAY_BUF_SIZE_IN_SAMPLES = (N_PLAY_SAMPLES_PER_SEC / 100); const int ENGINE_PLAY_BUF_SIZE_IN_SAMPLES = (N_PLAY_SAMPLES_PER_SEC / 100);
enum const int N_BLOCKS_IO = 2;
{ const int N_BUFFERS_IN = 2; // Must be at least N_BLOCKS_IO.
N_BLOCKS_IO = 2 const int N_BUFFERS_OUT = 3; // Must be at least N_BLOCKS_IO.
};
enum
{
N_BUFFERS_IN = 10
};
enum
{
N_BUFFERS_OUT = 3
}; // Must be at least N_BLOCKS_IO
const uint32_t TIMER_PERIOD_MS = (2 * 10 * N_BLOCKS_IO * 1000000); const int TIMER_PERIOD_MS = (2 * 10 * N_BLOCKS_IO * 1000000);
const uint32_t REC_BUF_SIZE_IN_SAMPLES = (ENGINE_REC_BUF_SIZE_IN_SAMPLES const int REC_BUF_SIZE_IN_SAMPLES =
* N_DEVICE_CHANNELS * N_BUFFERS_IN); ENGINE_REC_BUF_SIZE_IN_SAMPLES * N_DEVICE_CHANNELS * N_BUFFERS_IN;
const uint32_t PLAY_BUF_SIZE_IN_SAMPLES = const int PLAY_BUF_SIZE_IN_SAMPLES =
(ENGINE_PLAY_BUF_SIZE_IN_SAMPLES * N_PLAY_CHANNELS * N_BUFFERS_OUT); ENGINE_PLAY_BUF_SIZE_IN_SAMPLES * N_PLAY_CHANNELS * N_BUFFERS_OUT;
class AudioDeviceMac: public AudioDeviceGeneric class AudioDeviceMac: public AudioDeviceGeneric
{ {
@ -384,8 +375,8 @@ private:
semaphore_t _renderSemaphore; semaphore_t _renderSemaphore;
semaphore_t _captureSemaphore; semaphore_t _captureSemaphore;
uint32_t _captureBufSizeSamples; int _captureBufSizeSamples;
uint32_t _renderBufSizeSamples; int _renderBufSizeSamples;
}; };
} // namespace webrtc } // namespace webrtc