diff options
Diffstat (limited to 'gr-audio-osx/src/audio_osx_source.cc')
| -rw-r--r-- | gr-audio-osx/src/audio_osx_source.cc | 978 |
1 files changed, 978 insertions, 0 deletions
diff --git a/gr-audio-osx/src/audio_osx_source.cc b/gr-audio-osx/src/audio_osx_source.cc new file mode 100644 index 000000000..2abf1c2a7 --- /dev/null +++ b/gr-audio-osx/src/audio_osx_source.cc @@ -0,0 +1,978 @@ +/* -*- c++ -*- */ +/* + * Copyright 2006 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 2, 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., 59 Temple Place - Suite 330, + * Boston, MA 02111-1307, USA. + */ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#define _USE_OMNI_THREADS_ + +#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) { + fprintf (stderr, "PrintStreamDesc: Can't print a NULL desc!\n"); + return; + } + + fprintf (stderr, " Sample Rate : %g\n", inDesc->mSampleRate); + fprintf (stderr, " Format ID : %4s\n", (char*)&inDesc->mFormatID); + fprintf (stderr, " Format Flags : %lX\n", inDesc->mFormatFlags); + fprintf (stderr, " Bytes per Packet : %ld\n", inDesc->mBytesPerPacket); + fprintf (stderr, " Frames per Packet : %ld\n", inDesc->mFramesPerPacket); + fprintf (stderr, " Bytes per Frame : %ld\n", inDesc->mBytesPerFrame); + fprintf (stderr, " Channels per Frame : %ld\n", inDesc->mChannelsPerFrame); + fprintf (stderr, " Bits per Channel : %ld\n", inDesc->mBitsPerChannel); +} + +// 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) { + fprintf (stderr, "Invalid Sample Rate: %d\n", sample_rate); + throw std::invalid_argument ("audio_osx_source::audio_osx_source"); + } else + d_outputSampleRate = (Float64) sample_rate; + + if (channel_config <= 0 & channel_config != -1) { + fprintf (stderr, "Invalid Channel Config: %d\n", channel_config); + 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) { + fprintf (stderr, "Error Converting Device Name: %d\n", errno); + 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) { + fprintf (stderr, "Invalid Max Sample Count: %d\n", max_sample_count); + throw std::invalid_argument ("audio_osx_source::audio_osx_source"); + } + + d_max_sample_count = max_sample_count; + +#if _OSX_AU_DEBUG_ + fprintf (stderr, "source(): max # samples = %ld", d_max_sample_count); +#endif + + OSStatus err = noErr; + +// create the default AudioUnit for input + +// Open the default input unit + ComponentDescription InputDesc; + + InputDesc.componentType = kAudioUnitType_Output; + InputDesc.componentSubType = kAudioUnitSubType_HALOutput; + InputDesc.componentManufacturer = kAudioUnitManufacturer_Apple; + InputDesc.componentFlags = 0; + InputDesc.componentFlagsMask = 0; + + Component comp = FindNextComponent (NULL, &InputDesc); + if (comp == NULL) { + fprintf (stderr, "FindNextComponent Error\n"); + throw std::runtime_error ("audio_osx_source::audio_osx_source"); + } + + err = OpenAComponent (comp, &d_InputAU); + CheckErrorAndThrow (err, "OpenAComponent", + "audio_osx_source::audio_osx_source"); + + 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) { + fprintf (stderr, "Selected Audio Device does not support Input.\n"); + 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_ + fprintf (stderr, "---- Device Stream Format ----\n" ); + 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_ + fprintf (stderr, "---- Client Stream Format ----\n"); + 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_internal = new mld_mutex (); + if (d_internal == NULL) + CheckErrorAndThrow (errno, "new mld_mutex (internal)", + "audio_osx_source::audio_osx_source"); + + d_cond_data = new mld_condition (); + if (d_cond_data == NULL) + CheckErrorAndThrow (errno, "new mld_condition (data)", + "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_ + fprintf (stderr, "audio_osx_source Parameters:\n"); + fprintf (stderr, " Device Sample Rate is %g\n", d_deviceSampleRate); + fprintf (stderr, " User Sample Rate is %g\n", d_outputSampleRate); + fprintf (stderr, " Max Sample Count is %ld\n", d_max_sample_count); + fprintf (stderr, " # Device Channels is %ld\n", d_n_deviceChannels); + fprintf (stderr, " # Max Channels is %ld\n", d_n_max_channels); + fprintf (stderr, " Device Buffer Size is Frames = %ld\n", + d_deviceBufferSizeFrames); + fprintf (stderr, " Lead Size is Frames = %ld\n", + d_leadSizeFrames); + fprintf (stderr, " Trail Size is Frames = %ld\n", + d_trailSizeFrames); + fprintf (stderr, " Input Buffer Size is Frames = %ld\n", + d_inputBufferSizeFrames); + fprintf (stderr, " Output Buffer Size is Frames = %ld\n", + d_outputBufferSizeFrames); +#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"); + + err = CloseComponent (d_InputAU); + CheckError (err, "~audio_osx_source: CloseComponent"); + +// 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_internal; + delete d_cond_data; +} + +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 audio_osx_source_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) { + fprintf (stderr, "audio_osx_source::check_topology(): " + "number of input streams provided (%d) should be 0.\n", + ninputs); + 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)) { + fprintf (stderr, "audio_osx_source::check_topology(): " + "number of output streams provided (%d) should be in " + "[1,%ld] for the selected audio device.\n", + noutputs, d_n_max_channels); + 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_ + fprintf (stderr, "chk_topo: Actual # user output channels = %d\n", + noutputs); +#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 + d_internal->wait (); + +#if _OSX_AU_DEBUG_ + fprintf (stderr, "work1: SC = %4ld, #OI = %4d, #Chan = %ld\n", + d_queueSampleCount, noutput_items, output_items.size()); +#endif + +// ?: always block until there is something to output from the source +// or return anything that is available, even if it's less than desired? + + 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 + d_internal->post (); +// block until there is data to return + d_cond_data->wait (); +// the condition's signal() was called; acquire control +// to keep thread safe + d_internal->wait (); + } + } else { +// not enough data & not blocking; return nothing + return (0); + } + } + actual_noutput_items = d_queueSampleCount; + } + + int l_counter = (int) output_items.size(); + +// get the items from the circular buffers + while (--l_counter >= 0) { + UInt32 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) { + fprintf (stderr, "audio_osx_source::work(): " + "number of available items changing " + "unexpectedly; expecting %ld, got %ld.\n", + actual_noutput_items, t_n_output_items); + throw std::runtime_error ("audio_osx_source::work()"); + } + } + + d_queueSampleCount -= actual_noutput_items; + +#if _OSX_AU_DEBUG_ + fprintf (stderr, "work2: SC = %4ld, act#OI = %4ld\n", + d_queueSampleCount, actual_noutput_items); +#endif + +// release control to allow for other processing parts to run + d_internal->post (); + + 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_ + fprintf (stderr, "cc1: io#DP = %ld, TIBSB = %ld, #C = %d\n", + *ioNumberDataPackets, totalInputBufferSizeBytes, counter); +#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; + } + + 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); + + This->d_internal->wait (); + +#if _OSX_AU_DEBUG_ + fprintf (stderr, "cb0: in#F = %4ld, inBN = %ld, SC = %4ld\n", + inNumberFrames, inBusNumber, This->d_queueSampleCount); +#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_ + fprintf (stderr, "cb1: avail: #IF = %ld, #OF = %ld\n", + AvailableInputFrames, AvailableOutputFrames); +#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_ + fprintf (stderr, "cb2: actual: #IF = %ld, #OF = %ld\n", + This->d_n_ActualInputFrames, AvailableOutputFrames); + if (This->d_n_ActualInputFrames != AvailableInputFrames) + fprintf (stderr, "cb2.1: avail#IF = %ld, actual#IF = %ld\n", + AvailableInputFrames, This->d_n_ActualInputFrames); +#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; + 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_ + fprintf (stderr, "cb5: #OI = %4ld, #Cnt = %4ld, mSC = %ld, \n", + ActualOutputFrames, This->d_queueSampleCount, + This->d_max_sample_count); +#endif + +// signal that data is available, if appropraite + This->d_cond_data->signal (); + +// release control to allow for other processing parts to run + This->d_internal->post (); + + 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); + + fprintf (stderr, "a_o_s::HardwareListener\n"); + +// 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; + + fprintf (stderr, "a_o_s::UnitListener\n"); + +// 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"); + + fprintf (stderr, "UnitListener: Input Source changed.\n" + "Old Source Output Info:\n"); + 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"); + + fprintf (stderr, "New Source Output Info:\n"); + 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 |