/* -*- c++ -*- */
/*
* Copyright 2006-2011 Free Software Foundation, Inc.
*
* This file is part of GNU Radio.
*
* GNU Radio is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation; either version 3, or (at your option)
* any later version.
*
* GNU Radio is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License
* along with GNU Radio; see the file COPYING. If not, write to
* the Free Software Foundation, Inc., 51 Franklin Street,
* Boston, MA 02110-1301, USA.
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include "gr_audio_registry.h"
#include <audio_osx_source.h>
#include <gr_io_signature.h>
#include <stdexcept>
#include <audio_osx.h>
#define _OSX_AU_DEBUG_ 0
#define _OSX_DO_LISTENERS_ 0
AUDIO_REGISTER_SOURCE(REG_PRIO_HIGH, osx)(
int sampling_rate, const std::string &device_name, bool ok_to_block
){
return audio_source::sptr(new audio_osx_source(sampling_rate, device_name, ok_to_block));
}
void PrintStreamDesc (AudioStreamBasicDescription *inDesc)
{
if (inDesc == NULL) {
std::cerr << "PrintStreamDesc: Can't print a NULL desc!" << std::endl;
return;
}
std::cerr << " Sample Rate : " << inDesc->mSampleRate << std::endl;
char format_id[4];
strncpy (format_id, (char*)(&inDesc->mFormatID), 4);
std::cerr << " Format ID : " << format_id << std::endl;
std::cerr << " Format Flags : " << inDesc->mFormatFlags << std::endl;
std::cerr << " Bytes per Packet : " << inDesc->mBytesPerPacket << std::endl;
std::cerr << " Frames per Packet : " << inDesc->mFramesPerPacket << std::endl;
std::cerr << " Bytes per Frame : " << inDesc->mBytesPerFrame << std::endl;
std::cerr << " Channels per Frame : " << inDesc->mChannelsPerFrame << std::endl;
std::cerr << " Bits per Channel : " << inDesc->mBitsPerChannel << std::endl;
}
// FIXME these should query some kind of user preference
audio_osx_source::audio_osx_source (int sample_rate,
const std::string device_name,
bool do_block,
int channel_config,
int max_sample_count)
: gr_sync_block ("audio_osx_source",
gr_make_io_signature (0, 0, 0),
gr_make_io_signature (0, 0, 0)),
d_deviceSampleRate (0.0), d_outputSampleRate (0.0),
d_channel_config (0),
d_inputBufferSizeFrames (0), d_inputBufferSizeBytes (0),
d_outputBufferSizeFrames (0), d_outputBufferSizeBytes (0),
d_deviceBufferSizeFrames (0), d_deviceBufferSizeBytes (0),
d_leadSizeFrames (0), d_leadSizeBytes (0),
d_trailSizeFrames (0), d_trailSizeBytes (0),
d_extraBufferSizeFrames (0), d_extraBufferSizeBytes (0),
d_queueSampleCount (0), d_max_sample_count (0),
d_n_AvailableInputFrames (0), d_n_ActualInputFrames (0),
d_n_user_channels (0), d_n_max_channels (0), d_n_deviceChannels (0),
d_do_block (do_block), d_passThrough (false),
d_internal (0), d_cond_data (0),
d_buffers (0),
d_InputAU (0), d_InputBuffer (0), d_OutputBuffer (0),
d_AudioConverter (0)
{
if (sample_rate <= 0) {
std::cerr << "Invalid Sample Rate: " << sample_rate << std::endl;
throw std::invalid_argument ("audio_osx_source::audio_osx_source");
} else
d_outputSampleRate = (Float64) sample_rate;
if (channel_config <= 0 & channel_config != -1) {
std::cerr << "Invalid Channel Config: " << channel_config << std::endl;
throw std::invalid_argument ("audio_osx_source::audio_osx_source");
} else if (channel_config == -1) {
// no user input; try "device name" instead
int l_n_channels = (int) strtol (device_name.data(), (char **)NULL, 10);
if (l_n_channels == 0 & errno) {
std::cerr << "Error Converting Device Name: " << errno << std::endl;
throw std::invalid_argument ("audio_osx_source::audio_osx_source");
}
if (l_n_channels <= 0)
channel_config = 2;
else
channel_config = l_n_channels;
}
d_channel_config = channel_config;
// check that the max # of samples to store is valid
if (max_sample_count == -1)
max_sample_count = sample_rate;
else if (max_sample_count <= 0) {
std::cerr << "Invalid Max Sample Count: " << max_sample_count << std::endl;
throw std::invalid_argument ("audio_osx_source::audio_osx_source");
}
d_max_sample_count = max_sample_count;
#if _OSX_AU_DEBUG_
std::cerr << "source(): max # samples = " << d_max_sample_count << std::endl;
#endif
OSStatus err = noErr;
// create the default AudioUnit for input
// Open the default input unit
#ifndef GR_USE_OLD_AUDIO_UNIT
AudioComponentDescription InputDesc;
#else
ComponentDescription InputDesc;
#endif
InputDesc.componentType = kAudioUnitType_Output;
InputDesc.componentSubType = kAudioUnitSubType_HALOutput;
InputDesc.componentManufacturer = kAudioUnitManufacturer_Apple;
InputDesc.componentFlags = 0;
InputDesc.componentFlagsMask = 0;
#ifndef GR_USE_OLD_AUDIO_UNIT
AudioComponent comp = AudioComponentFindNext (NULL, &InputDesc);
#else
Component comp = FindNextComponent (NULL, &InputDesc);
#endif
if (comp == NULL) {
#ifndef GR_USE_OLD_AUDIO_UNIT
std::cerr << "AudioComponentFindNext Error" << std::endl;
#else
std::cerr << "FindNextComponent Error" << std::endl;
#endif
throw std::runtime_error ("audio_osx_source::audio_osx_source");
}
#ifndef GR_USE_OLD_AUDIO_UNIT
err = AudioComponentInstanceNew (comp, &d_InputAU);
CheckErrorAndThrow (err, "AudioComponentInstanceNew",
"audio_osx_source::audio_osx_source");
#else
err = OpenAComponent (comp, &d_InputAU);
CheckErrorAndThrow (err, "OpenAComponent",
"audio_osx_source::audio_osx_source");
#endif
UInt32 enableIO;
// must enable the AUHAL for input and disable output
// before setting the AUHAL's current device
// Enable input on the AUHAL
enableIO = 1;
err = AudioUnitSetProperty (d_InputAU,
kAudioOutputUnitProperty_EnableIO,
kAudioUnitScope_Input,
1, // input element
&enableIO,
sizeof (UInt32));
CheckErrorAndThrow (err, "AudioUnitSetProperty Input Enable",
"audio_osx_source::audio_osx_source");
// Disable output on the AUHAL
enableIO = 0;
err = AudioUnitSetProperty (d_InputAU,
kAudioOutputUnitProperty_EnableIO,
kAudioUnitScope_Output,
0, // output element
&enableIO,
sizeof (UInt32));
CheckErrorAndThrow (err, "AudioUnitSetProperty Output Disable",
"audio_osx_source::audio_osx_source");
// set the default input device for our input AU
SetDefaultInputDeviceAsCurrent ();
#if _OSX_DO_LISTENERS_
// set up a listener if default hardware input device changes
err = AudioHardwareAddPropertyListener
(kAudioHardwarePropertyDefaultInputDevice,
(AudioHardwarePropertyListenerProc) HardwareListener,
this);
CheckErrorAndThrow (err, "AudioHardwareAddPropertyListener",
"audio_osx_source::audio_osx_source");
// Add a listener for any changes in the input AU's output stream
// the function "UnitListener" will be called if the stream format
// changes for whatever reason
err = AudioUnitAddPropertyListener
(d_InputAU,
kAudioUnitProperty_StreamFormat,
(AudioUnitPropertyListenerProc) UnitListener,
this);
CheckErrorAndThrow (err, "Adding Unit Property Listener",
"audio_osx_source::audio_osx_source");
#endif
// Now find out if it actually can do input.
UInt32 hasInput = 0;
UInt32 dataSize = sizeof (hasInput);
err = AudioUnitGetProperty (d_InputAU,
kAudioOutputUnitProperty_HasIO,
kAudioUnitScope_Input,
1,
&hasInput,
&dataSize);
CheckErrorAndThrow (err, "AudioUnitGetProperty HasIO",
"audio_osx_source::audio_osx_source");
if (hasInput == 0) {
std::cerr << "Selected Audio Device does not support Input." << std::endl;
throw std::runtime_error ("audio_osx_source::audio_osx_source");
}
// Set up a callback function to retrieve input from the Audio Device
AURenderCallbackStruct AUCallBack;
AUCallBack.inputProc = (AURenderCallback)(audio_osx_source::AUInputCallback);
AUCallBack.inputProcRefCon = this;
err = AudioUnitSetProperty (d_InputAU,
kAudioOutputUnitProperty_SetInputCallback,
kAudioUnitScope_Global,
0,
&AUCallBack,
sizeof (AURenderCallbackStruct));
CheckErrorAndThrow (err, "AudioUnitSetProperty Input Callback",
"audio_osx_source::audio_osx_source");
UInt32 propertySize;
AudioStreamBasicDescription asbd_device, asbd_client, asbd_user;
// asbd_device: ASBD of the device that is creating the input data stream
// asbd_client: ASBD of the client size (output) of the hardware device
// asbd_user: ASBD of the user's arguments
// Get the Stream Format (device side)
propertySize = sizeof (asbd_device);
err = AudioUnitGetProperty (d_InputAU,
kAudioUnitProperty_StreamFormat,
kAudioUnitScope_Input,
1,
&asbd_device,
&propertySize);
CheckErrorAndThrow (err, "AudioUnitGetProperty Device Input Stream Format",
"audio_osx_source::audio_osx_source");
#if _OSX_AU_DEBUG_
std::cerr << std::endl << "---- Device Stream Format ----" << std::endl;
PrintStreamDesc (&asbd_device);
#endif
// Get the Stream Format (client side)
propertySize = sizeof (asbd_client);
err = AudioUnitGetProperty (d_InputAU,
kAudioUnitProperty_StreamFormat,
kAudioUnitScope_Output,
1,
&asbd_client,
&propertySize);
CheckErrorAndThrow (err, "AudioUnitGetProperty Device Ouput Stream Format",
"audio_osx_source::audio_osx_source");
#if _OSX_AU_DEBUG_
std::cerr << std::endl << "---- Client Stream Format ----" << std::endl;
PrintStreamDesc (&asbd_client);
#endif
// Set the format of all the AUs to the input/output devices channel count
// get the max number of input (& thus output) channels supported by
// this device
d_n_max_channels = asbd_device.mChannelsPerFrame;
// create the output io signature;
// no input siganture to set (source is hardware)
set_output_signature (gr_make_io_signature (1,
d_n_max_channels,
sizeof (float)));
// allocate the output circular buffer(s), one per channel
d_buffers = (circular_buffer<float>**) new
circular_buffer<float>* [d_n_max_channels];
UInt32 n_alloc = (UInt32) ceil ((double) d_max_sample_count);
for (UInt32 n = 0; n < d_n_max_channels; n++) {
d_buffers[n] = new circular_buffer<float> (n_alloc, false, false);
}
d_deviceSampleRate = asbd_device.mSampleRate;
d_n_deviceChannels = asbd_device.mChannelsPerFrame;
asbd_client.mSampleRate = asbd_device.mSampleRate;
asbd_client.mFormatID = kAudioFormatLinearPCM;
asbd_client.mFormatFlags = (kAudioFormatFlagIsFloat |
kAudioFormatFlagIsPacked |
kAudioFormatFlagIsNonInterleaved);
if ((asbd_client.mFormatID == kAudioFormatLinearPCM) &&
(d_n_deviceChannels == 1)) {
asbd_client.mFormatFlags &= ~kLinearPCMFormatFlagIsNonInterleaved;
}
asbd_client.mBytesPerFrame = sizeof (float);
asbd_client.mFramesPerPacket = 1;
asbd_client.mBitsPerChannel = asbd_client.mBytesPerFrame * 8;
asbd_client.mChannelsPerFrame = d_n_deviceChannels;
asbd_client.mBytesPerPacket = asbd_client.mBytesPerFrame;
propertySize = sizeof(AudioStreamBasicDescription);
err = AudioUnitSetProperty (d_InputAU,
kAudioUnitProperty_StreamFormat,
kAudioUnitScope_Output,
1,
&asbd_client,
propertySize);
CheckErrorAndThrow (err, "AudioUnitSetProperty Device Ouput Stream Format",
"audio_osx_source::audio_osx_source");
// create an ASBD for the user's wants
asbd_user.mSampleRate = d_outputSampleRate;
asbd_user.mFormatID = kAudioFormatLinearPCM;
asbd_user.mFormatFlags = (kLinearPCMFormatFlagIsFloat |
GR_PCM_ENDIANNESS |
kLinearPCMFormatFlagIsPacked |
kAudioFormatFlagIsNonInterleaved);
asbd_user.mBytesPerPacket = sizeof (float);
asbd_user.mFramesPerPacket = 1;
asbd_user.mBytesPerFrame = asbd_user.mBytesPerPacket;
asbd_user.mChannelsPerFrame = d_n_deviceChannels;
asbd_user.mBitsPerChannel = asbd_user.mBytesPerPacket * 8;
if (d_deviceSampleRate == d_outputSampleRate) {
// no need to do conversion if asbd_client matches user wants
d_passThrough = true;
d_leadSizeFrames = d_trailSizeFrames = 0L;
} else {
d_passThrough = false;
// Create the audio converter
err = AudioConverterNew (&asbd_client, &asbd_user, &d_AudioConverter);
CheckErrorAndThrow (err, "AudioConverterNew",
"audio_osx_source::audio_osx_source");
// Set the audio converter sample rate quality to "max" ...
// requires more samples, but should sound nicer
UInt32 ACQuality = kAudioConverterQuality_Max;
propertySize = sizeof (ACQuality);
err = AudioConverterSetProperty (d_AudioConverter,
kAudioConverterSampleRateConverterQuality,
propertySize,
&ACQuality);
CheckErrorAndThrow (err, "AudioConverterSetProperty "
"SampleRateConverterQuality",
"audio_osx_source::audio_osx_source");
// set the audio converter's prime method to "pre",
// which uses both leading and trailing frames
// from the "current input". All of this is handled
// internally by the AudioConverter; we just supply
// the frames for conversion.
// UInt32 ACPrimeMethod = kConverterPrimeMethod_None;
UInt32 ACPrimeMethod = kConverterPrimeMethod_Pre;
propertySize = sizeof (ACPrimeMethod);
err = AudioConverterSetProperty (d_AudioConverter,
kAudioConverterPrimeMethod,
propertySize,
&ACPrimeMethod);
CheckErrorAndThrow (err, "AudioConverterSetProperty PrimeMethod",
"audio_osx_source::audio_osx_source");
// Get the size of the I/O buffer(s) to allow for pre-allocated buffers
// lead frame info (trail frame info is ignored)
AudioConverterPrimeInfo ACPrimeInfo = {0, 0};
propertySize = sizeof (ACPrimeInfo);
err = AudioConverterGetProperty (d_AudioConverter,
kAudioConverterPrimeInfo,
&propertySize,
&ACPrimeInfo);
CheckErrorAndThrow (err, "AudioConverterGetProperty PrimeInfo",
"audio_osx_source::audio_osx_source");
switch (ACPrimeMethod) {
case (kConverterPrimeMethod_None):
d_leadSizeFrames =
d_trailSizeFrames = 0L;
break;
case (kConverterPrimeMethod_Normal):
d_leadSizeFrames = 0L;
d_trailSizeFrames = ACPrimeInfo.trailingFrames;
break;
default:
d_leadSizeFrames = ACPrimeInfo.leadingFrames;
d_trailSizeFrames = ACPrimeInfo.trailingFrames;
}
}
d_leadSizeBytes = d_leadSizeFrames * sizeof (Float32);
d_trailSizeBytes = d_trailSizeFrames * sizeof (Float32);
propertySize = sizeof (d_deviceBufferSizeFrames);
err = AudioUnitGetProperty (d_InputAU,
kAudioDevicePropertyBufferFrameSize,
kAudioUnitScope_Global,
0,
&d_deviceBufferSizeFrames,
&propertySize);
CheckErrorAndThrow (err, "AudioUnitGetProperty Buffer Frame Size",
"audio_osx_source::audio_osx_source");
d_deviceBufferSizeBytes = d_deviceBufferSizeFrames * sizeof (Float32);
d_inputBufferSizeBytes = d_deviceBufferSizeBytes + d_leadSizeBytes;
d_inputBufferSizeFrames = d_deviceBufferSizeFrames + d_leadSizeFrames;
// outBufSizeBytes = floor (inBufSizeBytes * rate_out / rate_in)
// since this is rarely exact, we need another buffer to hold
// "extra" samples not processed at any given sampling period
// this buffer must be at least 4 floats in size, but generally
// follows the rule that
// extraBufSize = ceil (rate_in / rate_out)*sizeof(float)
d_extraBufferSizeFrames = ((UInt32) ceil (d_deviceSampleRate
/ d_outputSampleRate)
* sizeof (float));
if (d_extraBufferSizeFrames < 4)
d_extraBufferSizeFrames = 4;
d_extraBufferSizeBytes = d_extraBufferSizeFrames * sizeof (float);
d_outputBufferSizeFrames = (UInt32) ceil (((Float64) d_inputBufferSizeFrames)
* d_outputSampleRate
/ d_deviceSampleRate);
d_outputBufferSizeBytes = d_outputBufferSizeFrames * sizeof (float);
d_inputBufferSizeFrames += d_extraBufferSizeFrames;
// pre-alloc all buffers
AllocAudioBufferList (&d_InputBuffer, d_n_deviceChannels,
d_inputBufferSizeBytes);
if (d_passThrough == false) {
AllocAudioBufferList (&d_OutputBuffer, d_n_max_channels,
d_outputBufferSizeBytes);
} else {
d_OutputBuffer = d_InputBuffer;
}
// create the stuff to regulate I/O
d_cond_data = new gruel::condition_variable ();
if (d_cond_data == NULL)
CheckErrorAndThrow (errno, "new condition (data)",
"audio_osx_source::audio_osx_source");
d_internal = new gruel::mutex ();
if (d_internal == NULL)
CheckErrorAndThrow (errno, "new mutex (internal)",
"audio_osx_source::audio_osx_source");
// initialize the AU for input
err = AudioUnitInitialize (d_InputAU);
CheckErrorAndThrow (err, "AudioUnitInitialize",
"audio_osx_source::audio_osx_source");
#if _OSX_AU_DEBUG_
std::cerr << "audio_osx_source Parameters:" << std::endl;
std::cerr << " Device Sample Rate is " << d_deviceSampleRate << std::endl;
std::cerr << " User Sample Rate is " << d_outputSampleRate << std::endl;
std::cerr << " Max Sample Count is " << d_max_sample_count << std::endl;
std::cerr << " # Device Channels is " << d_n_deviceChannels << std::endl;
std::cerr << " # Max Channels is " << d_n_max_channels << std::endl;
std::cerr << " Device Buffer Size is Frames = " << d_deviceBufferSizeFrames << std::endl;
std::cerr << " Lead Size is Frames = " << d_leadSizeFrames << std::endl;
std::cerr << " Trail Size is Frames = " << d_trailSizeFrames << std::endl;
std::cerr << " Input Buffer Size is Frames = " << d_inputBufferSizeFrames << std::endl;
std::cerr << " Output Buffer Size is Frames = " << d_outputBufferSizeFrames << std::endl;
#endif
}
void
audio_osx_source::AllocAudioBufferList (AudioBufferList** t_ABL,
UInt32 n_channels,
UInt32 bufferSizeBytes)
{
FreeAudioBufferList (t_ABL);
UInt32 propertySize = (offsetof (AudioBufferList, mBuffers[0]) +
(sizeof (AudioBuffer) * n_channels));
*t_ABL = (AudioBufferList*) calloc (1, propertySize);
(*t_ABL)->mNumberBuffers = n_channels;
int counter = n_channels;
while (--counter >= 0) {
(*t_ABL)->mBuffers[counter].mNumberChannels = 1;
(*t_ABL)->mBuffers[counter].mDataByteSize = bufferSizeBytes;
(*t_ABL)->mBuffers[counter].mData = calloc (1, bufferSizeBytes);
}
}
void
audio_osx_source::FreeAudioBufferList (AudioBufferList** t_ABL)
{
// free pre-allocated audio buffer, if it exists
if (*t_ABL != NULL) {
int counter = (*t_ABL)->mNumberBuffers;
while (--counter >= 0)
free ((*t_ABL)->mBuffers[counter].mData);
free (*t_ABL);
(*t_ABL) = 0;
}
}
bool audio_osx_source::IsRunning ()
{
UInt32 AURunning = 0, AUSize = sizeof (UInt32);
OSStatus err = AudioUnitGetProperty (d_InputAU,
kAudioOutputUnitProperty_IsRunning,
kAudioUnitScope_Global,
0,
&AURunning,
&AUSize);
CheckErrorAndThrow (err, "AudioUnitGetProperty IsRunning",
"audio_osx_source::IsRunning");
return (AURunning);
}
bool audio_osx_source::start ()
{
if (! IsRunning ()) {
OSStatus err = AudioOutputUnitStart (d_InputAU);
CheckErrorAndThrow (err, "AudioOutputUnitStart",
"audio_osx_source::start");
}
return (true);
}
bool audio_osx_source::stop ()
{
if (IsRunning ()) {
OSStatus err = AudioOutputUnitStop (d_InputAU);
CheckErrorAndThrow (err, "AudioOutputUnitStart",
"audio_osx_source::stop");
for (UInt32 n = 0; n < d_n_user_channels; n++) {
d_buffers[n]->abort ();
}
}
return (true);
}
audio_osx_source::~audio_osx_source ()
{
OSStatus err = noErr;
// stop the AudioUnit
stop();
#if _OSX_DO_LISTENERS_
// remove the listeners
err = AudioUnitRemovePropertyListener
(d_InputAU,
kAudioUnitProperty_StreamFormat,
(AudioUnitPropertyListenerProc) UnitListener);
CheckError (err, "~audio_osx_source: AudioUnitRemovePropertyListener");
err = AudioHardwareRemovePropertyListener
(kAudioHardwarePropertyDefaultInputDevice,
(AudioHardwarePropertyListenerProc) HardwareListener);
CheckError (err, "~audio_osx_source: AudioHardwareRemovePropertyListener");
#endif
// free pre-allocated audio buffers
FreeAudioBufferList (&d_InputBuffer);
if (d_passThrough == false) {
err = AudioConverterDispose (d_AudioConverter);
CheckError (err, "~audio_osx_source: AudioConverterDispose");
FreeAudioBufferList (&d_OutputBuffer);
}
// remove the audio unit
err = AudioUnitUninitialize (d_InputAU);
CheckError (err, "~audio_osx_source: AudioUnitUninitialize");
#ifndef GR_USE_OLD_AUDIO_UNIT
err = AudioComponentInstanceDispose (d_InputAU);
CheckError (err, "~audio_osx_source: AudioComponentInstanceDispose");
#else
err = CloseComponent (d_InputAU);
CheckError (err, "~audio_osx_source: CloseComponent");
#endif
// empty and delete the queues
for (UInt32 n = 0; n < d_n_max_channels; n++) {
delete d_buffers[n];
d_buffers[n] = 0;
}
delete [] d_buffers;
d_buffers = 0;
// close and delete the control stuff
delete d_cond_data;
d_cond_data = 0;
delete d_internal;
d_internal = 0;
}
bool
audio_osx_source::check_topology (int ninputs, int noutputs)
{
// check # inputs to make sure it's valid
if (ninputs != 0) {
std::cerr << "audio_osx_source::check_topology(): number of input "
<< "streams provided (" << ninputs
<< ") should be 0." << std::endl;
throw std::runtime_error ("audio_osx_source::check_topology()");
}
// check # outputs to make sure it's valid
if ((noutputs < 1) | (noutputs > (int) d_n_max_channels)) {
std::cerr << "audio_osx_source::check_topology(): number of output "
<< "streams provided (" << noutputs << ") should be in [1,"
<< d_n_max_channels << "] for the selected audio device."
<< std::endl;
throw std::runtime_error ("audio_osx_source::check_topology()");
}
// save the actual number of output (user) channels
d_n_user_channels = noutputs;
#if _OSX_AU_DEBUG_
std::cerr << "chk_topo: Actual # user output channels = "
<< noutputs << std::endl;
#endif
return (true);
}
int
audio_osx_source::work
(int noutput_items,
gr_vector_const_void_star &input_items,
gr_vector_void_star &output_items)
{
// acquire control to do processing here only
gruel::scoped_lock l (*d_internal);
#if _OSX_AU_DEBUG_
std::cerr << "work1: SC = " << d_queueSampleCount
<< ", #OI = " << noutput_items
<< ", #Chan = " << output_items.size() << std::endl;
#endif
// set the actual # of output items to the 'desired' amount then
// verify that data is available; if not enough data is available,
// either wait until it is (is "do_block" is true), return (0) is no
// data is available and "do_block" is false, or process the actual
// amount of available data.
UInt32 actual_noutput_items = noutput_items;
if (d_queueSampleCount < actual_noutput_items) {
if (d_queueSampleCount == 0) {
// no data; do_block decides what to do
if (d_do_block == true) {
while (d_queueSampleCount == 0) {
// release control so-as to allow data to be retrieved;
// block until there is data to return
d_cond_data->wait (l);
// the condition's 'notify' was called; acquire control to
// keep thread safe
}
} else {
// no data & not blocking; return nothing
return (0);
}
}
// use the actual amount of available data
actual_noutput_items = d_queueSampleCount;
}
// number of channels
int l_counter = (int) output_items.size();
// copy the items from the circular buffer(s) to 'work's output buffers
// verify that the number copied out is as expected.
while (--l_counter >= 0) {
size_t t_n_output_items = actual_noutput_items;
d_buffers[l_counter]->dequeue ((float*) output_items[l_counter],
&t_n_output_items);
if (t_n_output_items != actual_noutput_items) {
std::cerr << "audio_osx_source::work(): ERROR: number of "
<< "available items changing unexpectedly; expecting "
<< actual_noutput_items << ", got "
<< t_n_output_items << "." << std::endl;
throw std::runtime_error ("audio_osx_source::work()");
}
}
// subtract the actual number of items removed from the buffer(s)
// from the local accounting of the number of available samples
d_queueSampleCount -= actual_noutput_items;
#if _OSX_AU_DEBUG_
std::cerr << "work2: SC = " << d_queueSampleCount
<< ", act#OI = " << actual_noutput_items << std::endl
<< "Returning." << std::endl;
#endif
return (actual_noutput_items);
}
OSStatus
audio_osx_source::ConverterCallback
(AudioConverterRef inAudioConverter,
UInt32* ioNumberDataPackets,
AudioBufferList* ioData,
AudioStreamPacketDescription** ioASPD,
void* inUserData)
{
// take current device buffers and copy them to the tail of the
// input buffers the lead buffer is already there in the first
// d_leadSizeFrames slots
audio_osx_source* This = static_cast<audio_osx_source*>(inUserData);
AudioBufferList* l_inputABL = This->d_InputBuffer;
UInt32 totalInputBufferSizeBytes = ((*ioNumberDataPackets) * sizeof (float));
int counter = This->d_n_deviceChannels;
ioData->mNumberBuffers = This->d_n_deviceChannels;
This->d_n_ActualInputFrames = (*ioNumberDataPackets);
#if _OSX_AU_DEBUG_
std::cerr << "cc1: io#DP = " << (*ioNumberDataPackets)
<< ", TIBSB = " << totalInputBufferSizeBytes
<< ", #C = " << counter << std::endl;
#endif
while (--counter >= 0) {
AudioBuffer* l_ioD_AB = &(ioData->mBuffers[counter]);
l_ioD_AB->mNumberChannels = 1;
l_ioD_AB->mData = (float*)(l_inputABL->mBuffers[counter].mData);
l_ioD_AB->mDataByteSize = totalInputBufferSizeBytes;
}
#if _OSX_AU_DEBUG_
std::cerr << "cc2: Returning." << std::endl;
#endif
return (noErr);
}
OSStatus
audio_osx_source::AUInputCallback (void* inRefCon,
AudioUnitRenderActionFlags* ioActionFlags,
const AudioTimeStamp* inTimeStamp,
UInt32 inBusNumber,
UInt32 inNumberFrames,
AudioBufferList* ioData)
{
OSStatus err = noErr;
audio_osx_source* This = static_cast<audio_osx_source*>(inRefCon);
gruel::scoped_lock l (*This->d_internal);
#if _OSX_AU_DEBUG_
std::cerr << "cb0: in#F = " << inNumberFrames
<< ", inBN = " << inBusNumber
<< ", SC = " << This->d_queueSampleCount << std::endl;
#endif
// Get the new audio data from the input device
err = AudioUnitRender (This->d_InputAU,
ioActionFlags,
inTimeStamp,
1, //inBusNumber,
inNumberFrames,
This->d_InputBuffer);
CheckErrorAndThrow (err, "AudioUnitRender",
"audio_osx_source::AUInputCallback");
UInt32 AvailableInputFrames = inNumberFrames;
This->d_n_AvailableInputFrames = inNumberFrames;
// get the number of actual output frames,
// either via converting the buffer or not
UInt32 ActualOutputFrames;
if (This->d_passThrough == true) {
ActualOutputFrames = AvailableInputFrames;
} else {
UInt32 AvailableInputBytes = AvailableInputFrames * sizeof (float);
UInt32 AvailableOutputBytes = AvailableInputBytes;
UInt32 AvailableOutputFrames = AvailableOutputBytes / sizeof (float);
UInt32 propertySize = sizeof (AvailableOutputBytes);
err = AudioConverterGetProperty (This->d_AudioConverter,
kAudioConverterPropertyCalculateOutputBufferSize,
&propertySize,
&AvailableOutputBytes);
CheckErrorAndThrow (err, "AudioConverterGetProperty CalculateOutputBufferSize", "audio_osx_source::audio_osx_source");
AvailableOutputFrames = AvailableOutputBytes / sizeof (float);
#if 0
// when decimating too much, the output sounds warbly due to
// fluctuating # of output frames
// This should not be a surprise, but there's probably some
// clever programming that could lessed the effect ...
// like finding the "ideal" # of output frames, and keeping
// that number constant no matter the # of input frames
UInt32 l_InputBytes = AvailableOutputBytes;
propertySize = sizeof (AvailableOutputBytes);
err = AudioConverterGetProperty (This->d_AudioConverter,
kAudioConverterPropertyCalculateInputBufferSize,
&propertySize,
&l_InputBytes);
CheckErrorAndThrow (err, "AudioConverterGetProperty CalculateInputBufferSize", "audio_osx_source::audio_osx_source");
if (l_InputBytes < AvailableInputBytes) {
// OK to zero pad the input a little
AvailableOutputFrames += 1;
AvailableOutputBytes = AvailableOutputFrames * sizeof (float);
}
#endif
#if _OSX_AU_DEBUG_
std::cerr << "cb1: avail: #IF = " << AvailableInputFrames
<< ", #OF = " << AvailableOutputFrames << std::endl;
#endif
ActualOutputFrames = AvailableOutputFrames;
// convert the data to the correct rate
// on input, ActualOutputFrames is the number of available output frames
err = AudioConverterFillComplexBuffer (This->d_AudioConverter,
(AudioConverterComplexInputDataProc)(This->ConverterCallback),
inRefCon,
&ActualOutputFrames,
This->d_OutputBuffer,
NULL);
CheckErrorAndThrow (err, "AudioConverterFillComplexBuffer",
"audio_osx_source::AUInputCallback");
// on output, ActualOutputFrames is the actual number of output frames
#if _OSX_AU_DEBUG_
std::cerr << "cb2: actual: #IF = " << This->d_n_ActualInputFrames
<< ", #OF = " << AvailableOutputFrames << std::endl;
if (This->d_n_ActualInputFrames != AvailableInputFrames)
std::cerr << "cb2.1: avail#IF = " << AvailableInputFrames
<< ", actual#IF = " << This->d_n_ActualInputFrames << std::endl;
#endif
}
// add the output frames to the buffers' queue, checking for overflow
int l_counter = This->d_n_user_channels;
int res = 0;
while (--l_counter >= 0) {
float* inBuffer = (float*) This->d_OutputBuffer->mBuffers[l_counter].mData;
#if _OSX_AU_DEBUG_
std::cerr << "cb3: enqueuing audio data." << std::endl;
#endif
int l_res = This->d_buffers[l_counter]->enqueue (inBuffer, ActualOutputFrames);
if (l_res == -1)
res = -1;
}
if (res == -1) {
// data coming in too fast
// drop oldest buffer
fputs ("aO", stderr);
fflush (stderr);
// set the local number of samples available to the max
This->d_queueSampleCount = This->d_buffers[0]->buffer_length_items ();
} else {
// keep up the local sample count
This->d_queueSampleCount += ActualOutputFrames;
}
#if _OSX_AU_DEBUG_
std::cerr << "cb4: #OI = " << ActualOutputFrames
<< ", #Cnt = " << This->d_queueSampleCount
<< ", mSC = " << This->d_max_sample_count << std::endl;
#endif
// signal that data is available, if appropraite
This->d_cond_data->notify_one ();
#if _OSX_AU_DEBUG_
std::cerr << "cb5: returning." << std::endl;
#endif
return (err);
}
void
audio_osx_source::SetDefaultInputDeviceAsCurrent
()
{
// set the default input device
AudioDeviceID deviceID = 0;
UInt32 dataSize = sizeof (AudioDeviceID);
OSStatus err = noErr;
#ifndef GR_USE_OLD_AUDIO_UNIT
AudioObjectPropertyAddress theAddress =
{ kAudioHardwarePropertyDefaultInputDevice,
kAudioObjectPropertyScopeGlobal,
kAudioObjectPropertyElementMaster };
err = AudioObjectGetPropertyData
(kAudioObjectSystemObject,
&theAddress,
0,
NULL,
&dataSize,
&deviceID);
#else
err = AudioHardwareGetProperty
(kAudioHardwarePropertyDefaultInputDevice,
&dataSize,
&deviceID);
#endif
CheckErrorAndThrow (err, "Get Audio Unit Property for Current Device",
"audio_osx_source::SetDefaultInputDeviceAsCurrent");
err = AudioUnitSetProperty
(d_InputAU,
kAudioOutputUnitProperty_CurrentDevice,
kAudioUnitScope_Global,
0,
&deviceID,
sizeof (AudioDeviceID));
CheckErrorAndThrow (err, "AudioUnitSetProperty Current Device",
"audio_osx_source::SetDefaultInputDeviceAsCurrent");
}
#if _OSX_DO_LISTENERS_
OSStatus
audio_osx_source::HardwareListener
(AudioHardwarePropertyID inPropertyID,
void *inClientData)
{
OSStatus err = noErr;
audio_osx_source* This = static_cast<audio_osx_source*>(inClientData);
std::cerr << "a_o_s::HardwareListener" << std::endl;
// set the new default hardware input device for use by our AU
This->SetDefaultInputDeviceAsCurrent ();
// reset the converter to tell it that the stream has changed
err = AudioConverterReset (This->d_AudioConverter);
CheckErrorAndThrow (err, "AudioConverterReset",
"audio_osx_source::UnitListener");
return (err);
}
OSStatus
audio_osx_source::UnitListener
(void *inRefCon,
AudioUnit ci,
AudioUnitPropertyID inID,
AudioUnitScope inScope,
AudioUnitElement inElement)
{
OSStatus err = noErr;
audio_osx_source* This = static_cast<audio_osx_source*>(inRefCon);
AudioStreamBasicDescription asbd;
std::cerr << "a_o_s::UnitListener" << std::endl;
// get the converter's input ASBD (for printing)
UInt32 propertySize = sizeof (asbd);
err = AudioConverterGetProperty (This->d_AudioConverter,
kAudioConverterCurrentInputStreamDescription,
&propertySize,
&asbd);
CheckErrorAndThrow (err, "AudioConverterGetProperty "
"CurrentInputStreamDescription",
"audio_osx_source::UnitListener");
std::cerr << "UnitListener: Input Source changed." << std::endl
<< "Old Source Output Info:" << std::endl;
PrintStreamDesc (&asbd);
// get the new input unit's output ASBD
propertySize = sizeof (asbd);
err = AudioUnitGetProperty (This->d_InputAU,
kAudioUnitProperty_StreamFormat,
kAudioUnitScope_Output, 1,
&asbd, &propertySize);
CheckErrorAndThrow (err, "AudioUnitGetProperty StreamFormat",
"audio_osx_source::UnitListener");
std::cerr << "New Source Output Info:" << std::endl;
PrintStreamDesc (&asbd);
// set the converter's input ASBD to this
err = AudioConverterSetProperty (This->d_AudioConverter,
kAudioConverterCurrentInputStreamDescription,
propertySize,
&asbd);
CheckErrorAndThrow (err, "AudioConverterSetProperty "
"CurrentInputStreamDescription",
"audio_osx_source::UnitListener");
// reset the converter to tell it that the stream has changed
err = AudioConverterReset (This->d_AudioConverter);
CheckErrorAndThrow (err, "AudioConverterReset",
"audio_osx_source::UnitListener");
return (err);
}
#endif