From 5748eff26f835bffedb41bd5748ccbaefbe1e02f Mon Sep 17 00:00:00 2001
From: Johnathan Corgan <jcorgan@corganenterprises.com>
Date: Mon, 14 Mar 2011 10:26:56 -0700
Subject: audio: remove obsoleted individual top-level components
---
gr-audio-osx/src/audio_osx_source.cc | 1023 ----------------------------------
1 file changed, 1023 deletions(-)
delete mode 100644 gr-audio-osx/src/audio_osx_source.cc
(limited to 'gr-audio-osx/src/audio_osx_source.cc')
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 757e65a9e2..0000000000
--- 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
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