diff options
Diffstat (limited to 'gr-audio-osx/src/audio_osx_source.cc')
| -rw-r--r-- | gr-audio-osx/src/audio_osx_source.cc | 1023 |
1 files changed, 0 insertions, 1023 deletions
diff --git a/gr-audio-osx/src/audio_osx_source.cc b/gr-audio-osx/src/audio_osx_source.cc deleted file mode 100644 index 757e65a9e..000000000 --- a/gr-audio-osx/src/audio_osx_source.cc +++ /dev/null @@ -1,1023 +0,0 @@ -/* -*- c++ -*- */ -/* - * Copyright 2006,2010 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 <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 - -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_client.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; - -// 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 = 4; - asbd_user.mFramesPerPacket = 1; - asbd_user.mBytesPerFrame = 4; - asbd_user.mChannelsPerFrame = d_n_max_channels; - asbd_user.mBitsPerChannel = 32; - - 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; -} - -audio_osx_source_sptr -audio_osx_make_source (int sampling_freq, - const std::string device_name, - bool do_block, - int channel_config, - int max_sample_count) -{ - return gnuradio::get_initial_sptr(new audio_osx_source (sampling_freq, - device_name, - do_block, - channel_config, - max_sample_count)); -} - -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; - UInt32 dataSize = sizeof (AudioDeviceID); - AudioHardwareGetProperty (kAudioHardwarePropertyDefaultInputDevice, - &dataSize, - &deviceID); - OSStatus 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 |