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-rw-r--r--gr-audio/lib/osx/audio_osx_sink.cc404
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diff --git a/gr-audio/lib/osx/audio_osx_sink.cc b/gr-audio/lib/osx/audio_osx_sink.cc
deleted file mode 100644
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--- a/gr-audio/lib/osx/audio_osx_sink.cc
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@@ -1,404 +0,0 @@
-/* -*- c++ -*- */
-/*
- * Copyright 2006-2011 Free Software Foundation, Inc.
- *
- * This file is part of GNU Radio.
- *
- * GNU Radio is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 3, or (at your option)
- * any later version.
- *
- * GNU Radio is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
- * GNU General Public License for more details.
- *
- * You should have received a copy of the GNU General Public License
- * along with GNU Radio; see the file COPYING. If not, write to
- * the Free Software Foundation, Inc., 51 Franklin Street,
- * Boston, MA 02110-1301, USA.
- */
-
-#ifdef HAVE_CONFIG_H
-#include "config.h"
-#endif
-
-#include "gr_audio_registry.h"
-#include <audio_osx_sink.h>
-#include <gr_io_signature.h>
-#include <stdexcept>
-#include <audio_osx.h>
-
-#define _OSX_AU_DEBUG_ 0
-
-AUDIO_REGISTER_SINK(REG_PRIO_HIGH, osx)(
- int sampling_rate, const std::string &device_name, bool ok_to_block
-){
- return audio_sink::sptr(new audio_osx_sink(sampling_rate, device_name, ok_to_block));
-}
-
-audio_osx_sink::audio_osx_sink (int sample_rate,
- const std::string device_name,
- bool do_block,
- int channel_config,
- int max_sample_count)
- : gr_sync_block ("audio_osx_sink",
- gr_make_io_signature (0, 0, 0),
- gr_make_io_signature (0, 0, 0)),
- d_sample_rate (0.0), d_channel_config (0), d_n_channels (0),
- d_queueSampleCount (0), d_max_sample_count (0),
- d_do_block (do_block), d_internal (0), d_cond_data (0),
- d_OutputAU (0)
-{
- if (sample_rate <= 0) {
- std::cerr << "Invalid Sample Rate: " << sample_rate << std::endl;
- throw std::invalid_argument ("audio_osx_sink::audio_osx_sink");
- } else
- d_sample_rate = (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_sink::audio_osx_sink");
- } 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_sink::audio_osx_sink");
- }
- if (l_n_channels <= 0)
- channel_config = 2;
- else
- channel_config = l_n_channels;
- }
-
- d_n_channels = d_channel_config = channel_config;
-
-// set the input signature
-
- set_input_signature (gr_make_io_signature (1, d_n_channels, sizeof (float)));
-
-// 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_sink::audio_osx_sink");
- }
-
- d_max_sample_count = max_sample_count;
-
-// allocate the output circular buffer(s), one per channel
-
- d_buffers = (circular_buffer<float>**) new
- circular_buffer<float>* [d_n_channels];
- UInt32 n_alloc = (UInt32) ceil ((double) d_max_sample_count);
- for (UInt32 n = 0; n < d_n_channels; n++) {
- d_buffers[n] = new circular_buffer<float> (n_alloc, false, false);
- }
-
-// create the default AudioUnit for output
- OSStatus err = noErr;
-
-// Open the default output unit
-#ifndef GR_USE_OLD_AUDIO_UNIT
- AudioComponentDescription desc;
-#else
- ComponentDescription desc;
-#endif
-
- desc.componentType = kAudioUnitType_Output;
- desc.componentSubType = kAudioUnitSubType_DefaultOutput;
- desc.componentManufacturer = kAudioUnitManufacturer_Apple;
- desc.componentFlags = 0;
- desc.componentFlagsMask = 0;
-
-#ifndef GR_USE_OLD_AUDIO_UNIT
- AudioComponent comp = AudioComponentFindNext(NULL, &desc);
- if (comp == NULL) {
- std::cerr << "AudioComponentFindNext Error" << std::endl;
- throw std::runtime_error ("audio_osx_sink::audio_osx_sink");
- }
-#else
- Component comp = FindNextComponent (NULL, &desc);
- if (comp == NULL) {
- std::cerr << "FindNextComponent Error" << std::endl;
- throw std::runtime_error ("audio_osx_sink::audio_osx_sink");
- }
-#endif
-
-#ifndef GR_USE_OLD_AUDIO_UNIT
- err = AudioComponentInstanceNew (comp, &d_OutputAU);
- CheckErrorAndThrow (err, "AudioComponentInstanceNew", "audio_osx_sink::audio_osx_sink");
-#else
- err = OpenAComponent (comp, &d_OutputAU);
- CheckErrorAndThrow (err, "OpenAComponent", "audio_osx_sink::audio_osx_sink");
-#endif
-
-// Set up a callback function to generate output to the output unit
-
- AURenderCallbackStruct input;
- input.inputProc = (AURenderCallback)(audio_osx_sink::AUOutputCallback);
- input.inputProcRefCon = this;
-
- err = AudioUnitSetProperty (d_OutputAU,
- kAudioUnitProperty_SetRenderCallback,
- kAudioUnitScope_Input,
- 0,
- &input,
- sizeof (input));
- CheckErrorAndThrow (err, "AudioUnitSetProperty Render Callback", "audio_osx_sink::audio_osx_sink");
-
-// tell the Output Unit what format data will be supplied to it
-// so that it handles any format conversions
-
- AudioStreamBasicDescription streamFormat;
- streamFormat.mSampleRate = (Float64)(sample_rate);
- streamFormat.mFormatID = kAudioFormatLinearPCM;
- streamFormat.mFormatFlags = (kLinearPCMFormatFlagIsFloat |
- GR_PCM_ENDIANNESS |
- kLinearPCMFormatFlagIsPacked |
- kAudioFormatFlagIsNonInterleaved);
- streamFormat.mBytesPerPacket = 4;
- streamFormat.mFramesPerPacket = 1;
- streamFormat.mBytesPerFrame = 4;
- streamFormat.mChannelsPerFrame = d_n_channels;
- streamFormat.mBitsPerChannel = 32;
-
- err = AudioUnitSetProperty (d_OutputAU,
- kAudioUnitProperty_StreamFormat,
- kAudioUnitScope_Input,
- 0,
- &streamFormat,
- sizeof (AudioStreamBasicDescription));
- CheckErrorAndThrow (err, "AudioUnitSetProperty StreamFormat", "audio_osx_sink::audio_osx_sink");
-
-// 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_sink::audio_osx_sink");
-
- d_internal = new gruel::mutex ();
- if (d_internal == NULL)
- CheckErrorAndThrow (errno, "new mutex (internal)",
- "audio_osx_sink::audio_osx_sink");
-
-// initialize the AU for output
-
- err = AudioUnitInitialize (d_OutputAU);
- CheckErrorAndThrow (err, "AudioUnitInitialize",
- "audio_osx_sink::audio_osx_sink");
-
-#if _OSX_AU_DEBUG_
- std::cerr << "audio_osx_sink Parameters:" << std::endl;
- std::cerr << " Sample Rate is " << d_sample_rate << std::endl;
- std::cerr << " Number of Channels is " << d_n_channels << std::endl;
- std::cerr << " Max # samples to store per channel is " << d_max_sample_count << std::endl;
-#endif
-}
-
-bool audio_osx_sink::IsRunning ()
-{
- UInt32 AURunning = 0, AUSize = sizeof (UInt32);
-
- OSStatus err = AudioUnitGetProperty (d_OutputAU,
- kAudioOutputUnitProperty_IsRunning,
- kAudioUnitScope_Global,
- 0,
- &AURunning,
- &AUSize);
- CheckErrorAndThrow (err, "AudioUnitGetProperty IsRunning",
- "audio_osx_sink::IsRunning");
-
- return (AURunning);
-}
-
-bool audio_osx_sink::start ()
-{
- if (! IsRunning ()) {
- OSStatus err = AudioOutputUnitStart (d_OutputAU);
- CheckErrorAndThrow (err, "AudioOutputUnitStart", "audio_osx_sink::start");
- }
-
- return (true);
-}
-
-bool audio_osx_sink::stop ()
-{
- if (IsRunning ()) {
- OSStatus err = AudioOutputUnitStop (d_OutputAU);
- CheckErrorAndThrow (err, "AudioOutputUnitStop", "audio_osx_sink::stop");
-
- for (UInt32 n = 0; n < d_n_channels; n++) {
- d_buffers[n]->abort ();
- }
- }
-
- return (true);
-}
-
-audio_osx_sink::~audio_osx_sink ()
-{
-// stop and close the AudioUnit
- stop ();
- AudioUnitUninitialize (d_OutputAU);
-#ifndef GR_USE_OLD_AUDIO_UNIT
- AudioComponentInstanceDispose (d_OutputAU);
-#else
- CloseComponent (d_OutputAU);
-#endif
-
-// empty and delete the queues
- for (UInt32 n = 0; n < d_n_channels; n++) {
- delete d_buffers[n];
- d_buffers[n] = 0;
- }
- delete [] d_buffers;
- d_buffers = 0;
-
-// close and delete control stuff
- delete d_cond_data;
- d_cond_data = 0;
- delete d_internal;
- d_internal = 0;
-}
-
-int
-audio_osx_sink::work (int noutput_items,
- gr_vector_const_void_star &input_items,
- gr_vector_void_star &output_items)
-{
- gruel::scoped_lock l (*d_internal);
-
- /* take the input data, copy it, and push it to the bottom of the queue
- mono input are pushed onto queue[0];
- stereo input are pushed onto queue[1].
- Start the AudioUnit if necessary. */
-
- UInt32 l_max_count;
- int diff_count = d_max_sample_count - noutput_items;
- if (diff_count < 0)
- l_max_count = 0;
- else
- l_max_count = (UInt32) diff_count;
-
-#if 0
- if (l_max_count < d_queueItemLength->back()) {
-// allow 2 buffers at a time, regardless of length
- l_max_count = d_queueItemLength->back();
- }
-#endif
-
-#if _OSX_AU_DEBUG_
- std::cerr << "work1: qSC = " << d_queueSampleCount << ", lMC = "<< l_max_count
- << ", dmSC = " << d_max_sample_count << ", nOI = " << noutput_items << std::endl;
-#endif
-
- if (d_queueSampleCount > l_max_count) {
-// data coming in too fast; do_block decides what to do
- if (d_do_block == true) {
-// block until there is data to return
- while (d_queueSampleCount > l_max_count) {
-// 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
- }
- }
- }
-// not blocking case and overflow is handled by the circular buffer
-
-// add the input frames to the buffers' queue, checking for overflow
-
- UInt32 l_counter;
- int res = 0;
- float* inBuffer = (float*) input_items[0];
- const UInt32 l_size = input_items.size();
- for (l_counter = 0; l_counter < l_size; l_counter++) {
- inBuffer = (float*) input_items[l_counter];
- int l_res = d_buffers[l_counter]->enqueue (inBuffer,
- noutput_items);
- if (l_res == -1)
- res = -1;
- }
- while (l_counter < d_n_channels) {
-// for extra channels, copy the last input's data
- int l_res = d_buffers[l_counter++]->enqueue (inBuffer,
- noutput_items);
- 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
- d_queueSampleCount = d_buffers[0]->buffer_length_items ();
- } else {
-// keep up the local sample count
- d_queueSampleCount += noutput_items;
- }
-
-#if _OSX_AU_DEBUG_
- std::cerr << "work2: #OI = " << noutput_items << ", #Cnt = "
- << d_queueSampleCount << ", mSC = " << d_max_sample_count << std::endl;
-#endif
-
- return (noutput_items);
-}
-
-OSStatus audio_osx_sink::AUOutputCallback
-(void *inRefCon,
- AudioUnitRenderActionFlags *ioActionFlags,
- const AudioTimeStamp *inTimeStamp,
- UInt32 inBusNumber,
- UInt32 inNumberFrames,
- AudioBufferList *ioData)
-{
- audio_osx_sink* This = (audio_osx_sink*) inRefCon;
- OSStatus err = noErr;
-
- gruel::scoped_lock l (*This->d_internal);
-
-#if _OSX_AU_DEBUG_
- std::cerr << "cb_in: SC = " << This->d_queueSampleCount
- << ", in#F = " << inNumberFrames << std::endl;
-#endif
-
- if (This->d_queueSampleCount < inNumberFrames) {
-// not enough data to fill request
- err = -1;
- } else {
-// enough data; remove data from our buffers into the AU's buffers
- int l_counter = This->d_n_channels;
-
- while (--l_counter >= 0) {
- size_t t_n_output_items = inNumberFrames;
- float* outBuffer = (float*) ioData->mBuffers[l_counter].mData;
- This->d_buffers[l_counter]->dequeue (outBuffer, &t_n_output_items);
- if (t_n_output_items != inNumberFrames) {
- throw std::runtime_error ("audio_osx_sink::AUOutputCallback(): "
- "number of available items changing "
- "unexpectedly.\n");
- }
- }
-
- This->d_queueSampleCount -= inNumberFrames;
- }
-
-#if _OSX_AU_DEBUG_
- std::cerr << "cb_out: SC = " << This->d_queueSampleCount << std::endl;
-#endif
-
-// signal that data is available
- This->d_cond_data->notify_one ();
-
- return (err);
-}