forked from Mirrors/freeswitch
39a9e94305
This updates mod_portaudio to use the new v19 api and also contains major behavioural changes. This initial check-in should be tested to find any obscure use cases that lead to crashes etc... All of the old api interface commands are now depricated and any attempt to use them should cause a polite warning asking you to try the new single "pa" command. New Features: *) Mulitiple calls with hold/call switching. *) Inbound calls can play a ring file on specified device. (global and per call) *) Optional hold music for backgrounded calls. (global and per call) Example dialplan usage: <extension name="2000"> <condition field="destination_number" expression="^2000$"> <!--if the next 3 lines are omitted the defaults will be used from portaudio.conf--> <action application="set" data="pa_ring_file=/sounds/myring.wav"/> <action application="set" data="pa_hold_file=/sounds/myhold.wav"/> <action application="set" data="export_vars=pa_ring_file,pa_hold_file"/> <action application="bridge" data="portaudio"/> </condition> </extension> Example API interface usage: call extension 1000 > pa call 1000 call extension 1001 putting the other call on hold > pa call 1001 swap the calls between hold and active > pa switch view the current calls > pa list forground the call with id 1 > pa switch 1 background all calls > pa switch none send a dtmf string (1234) to the current call > pa dtmf 1234 answer the oldest unanswered inbound call > pa answer answer the call with id 1 > pa answer 1 hangup the active call > pa hangup hangup the call with id 1 > pa hangup 1 get device info > pa dump print usage summary > pa help USAGE: -------------------------------------------------------------------------------- pa help pa dump pa call <dest> [<dialplan> <cid_name> <cid_num> <rate>] pa answer [<call_id>] pa hangup [<call_id>] pa list pa switch [<call_id>|none] pa_dtmf <digit string> -------------------------------------------------------------------------------- The source of the portaudio v19 library will also be checked in for the sake of the build system. git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@3981 d0543943-73ff-0310-b7d9-9358b9ac24b2
191 lines
6.9 KiB
C
191 lines
6.9 KiB
C
/*
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* $Id: debug_dual.c 1083 2006-08-23 07:30:49Z rossb $
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* debug_dual.c
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* Try to open TWO streams on separate cards.
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* Play a sine sweep using the Portable Audio api for several seconds.
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* Hacked test for debugging PA.
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*
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* Author: Phil Burk <philburk@softsynth.com>
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*
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* This program uses the PortAudio Portable Audio Library.
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* For more information see: http://www.portaudio.com
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* Copyright (c) 1999-2000 Ross Bencina and Phil Burk
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*
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* Permission is hereby granted, free of charge, to any person obtaining
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* a copy of this software and associated documentation files
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* (the "Software"), to deal in the Software without restriction,
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* including without limitation the rights to use, copy, modify, merge,
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* publish, distribute, sublicense, and/or sell copies of the Software,
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* and to permit persons to whom the Software is furnished to do so,
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* subject to the following conditions:
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*
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* The above copyright notice and this permission notice shall be
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* included in all copies or substantial portions of the Software.
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*
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* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
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* EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
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* MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
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* IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR
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* ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF
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* CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION
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* WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
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*/
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/*
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* The text above constitutes the entire PortAudio license; however,
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* the PortAudio community also makes the following non-binding requests:
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*
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* Any person wishing to distribute modifications to the Software is
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* requested to send the modifications to the original developer so that
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* they can be incorporated into the canonical version. It is also
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* requested that these non-binding requests be included along with the
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* license above.
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*/
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#include <stdio.h>
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#include <math.h>
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#include "portaudio.h"
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#define DEV_ID_1 (13)
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#define DEV_ID_2 (15)
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#define NUM_SECONDS (8)
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#define SLEEP_DUR (800)
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#define SAMPLE_RATE (44100)
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#define FRAMES_PER_BUFFER (256)
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#if 0
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#define MIN_LATENCY_MSEC (200)
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#define NUM_BUFFERS ((MIN_LATENCY_MSEC * SAMPLE_RATE) / (FRAMES_PER_BUFFER * 1000))
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#else
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#define NUM_BUFFERS (0)
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#endif
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#define MIN_FREQ (100.0f)
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#define MAX_FREQ (4000.0f)
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#define FREQ_SCALAR (1.00002f)
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#define CalcPhaseIncrement(freq) (freq/SAMPLE_RATE)
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#ifndef M_PI
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#define M_PI (3.14159265)
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#endif
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#define TABLE_SIZE (400)
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typedef struct
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{
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float sine[TABLE_SIZE + 1]; // add one for guard point for interpolation
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float phase_increment;
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float left_phase;
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float right_phase;
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}
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paTestData;
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/* Convert phase between and 1.0 to sine value
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* using linear interpolation.
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*/
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float LookupSine( paTestData *data, float phase );
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float LookupSine( paTestData *data, float phase )
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{
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float fIndex = phase*TABLE_SIZE;
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int index = (int) fIndex;
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float fract = fIndex - index;
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float lo = data->sine[index];
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float hi = data->sine[index+1];
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float val = lo + fract*(hi-lo);
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return val;
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}
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/* This routine will be called by the PortAudio engine when audio is needed.
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** It may called at interrupt level on some machines so don't do anything
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** that could mess up the system like calling malloc() or free().
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*/
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static int patestCallback( void *inputBuffer, void *outputBuffer,
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unsigned long framesPerBuffer,
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PaTimestamp outTime, void *userData )
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{
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paTestData *data = (paTestData*)userData;
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float *out = (float*)outputBuffer;
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unsigned long i;
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int finished = 0;
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(void) outTime; /* Prevent unused variable warnings. */
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(void) inputBuffer;
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for( i=0; i<framesPerBuffer; i++ )
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{
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*out++ = LookupSine(data, data->left_phase); /* left */
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*out++ = LookupSine(data, data->right_phase); /* right */
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data->left_phase += data->phase_increment;
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if( data->left_phase >= 1.0f ) data->left_phase -= 1.0f;
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data->right_phase += (data->phase_increment * 1.5f); /* fifth above */
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if( data->right_phase >= 1.0f ) data->right_phase -= 1.0f;
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/* sweep frequency then start over. */
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data->phase_increment *= FREQ_SCALAR;
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if( data->phase_increment > CalcPhaseIncrement(MAX_FREQ) ) data->phase_increment = CalcPhaseIncrement(MIN_FREQ);
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}
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return 0;
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}
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PaError TestStart( PortAudioStream **streamPtr, PaDeviceID devID,
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paTestData *data );
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/*******************************************************************/
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int main(void);
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int main(void)
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{
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PortAudioStream *stream1, *stream2;
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PaError err;
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paTestData DATA1, DATA2;
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printf("PortAudio Test: DUAL sine sweep. ask for %d buffers\n", NUM_BUFFERS );
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err = Pa_Initialize();
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if( err != paNoError ) goto error;
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err = TestStart( &stream1, DEV_ID_1, &DATA1 );
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if( err != paNoError ) goto error;
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err = TestStart( &stream2, DEV_ID_2, &DATA2 );
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if( err != paNoError ) goto error;
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printf("Hit ENTER\n");
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getchar();
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err = Pa_StopStream( stream1 );
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if( err != paNoError ) goto error;
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err = Pa_StopStream( stream2 );
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if( err != paNoError ) goto error;
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Pa_Terminate();
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printf("Test finished.\n");
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return err;
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error:
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Pa_Terminate();
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fprintf( stderr, "An error occured while using the portaudio stream\n" );
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fprintf( stderr, "Error number: %d\n", err );
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fprintf( stderr, "Error message: %s\n", Pa_GetErrorText( err ) );
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return err;
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}
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PaError TestStart( PortAudioStream **streamPtr, PaDeviceID devID, paTestData *data )
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{
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PortAudioStream *stream;
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PaError err;
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int i;
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/* initialise sinusoidal wavetable */
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for( i=0; i<TABLE_SIZE; i++ )
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{
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data->sine[i] = (float) sin( ((double)i/(double)TABLE_SIZE) * M_PI * 2. );
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}
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data->sine[TABLE_SIZE] = data->sine[0]; // set guard point
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data->left_phase = data->right_phase = 0.0;
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data->phase_increment = CalcPhaseIncrement(MIN_FREQ);
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printf("PortAudio Test: output device = %d\n", devID );
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err = Pa_OpenStream(
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&stream,
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paNoDevice,
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0, /* no input */
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paFloat32, /* 32 bit floating point input */
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NULL,
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devID,
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2, /* stereo output */
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paFloat32, /* 32 bit floating point output */
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NULL,
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SAMPLE_RATE,
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FRAMES_PER_BUFFER,
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NUM_BUFFERS, /* number of buffers, if zero then use default minimum */
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paClipOff|paDitherOff, /* we won't output out of range samples so don't bother clipping them */
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patestCallback,
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data );
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if( err != paNoError ) goto error;
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err = Pa_StartStream( stream );
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if( err != paNoError ) goto error;
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*streamPtr = stream;
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return 0;
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error:
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return err;
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}
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