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-rw-r--r--gr-audio/lib/osx/audio_osx_sink.cc404
1 files changed, 404 insertions, 0 deletions
diff --git a/gr-audio/lib/osx/audio_osx_sink.cc b/gr-audio/lib/osx/audio_osx_sink.cc
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+++ b/gr-audio/lib/osx/audio_osx_sink.cc
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+/* -*- 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)
+ : audio_sink ("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);
+}