mirror of
https://github.com/saitohirga/WSJT-X.git
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184 lines
6.6 KiB
C
184 lines
6.6 KiB
C
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/*
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* $Id$
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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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* 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.
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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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#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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