I am trying to get started with PortAudio. I am able to build the bundled example file "paex_sine.c" with no problems. This a sine wave on the left channel and a different frequency sine wave on the right channel. This works properly with no errors.
My setup is a 32-bit PC running Puppy Linux Slacko 5.5. It has a SoundBlaster SB0200 with EMU10k1x chip. The Alsa library is v1.0.26, and driver is v1.0.24. I have tested all 5.1 channels using this command:
% speaker-test -Dplug:surround51 -c6
The test plays sound properly on each of the 6 channels, though it does complain about broken pipes. This is probably because the buffer is not large enough in the test program for all 6 channels.
The issue that I am having is that when I modify "paex_sine.c" to run on 6 channels instead of just 2, it will only play sound through the front-right and front-left channels. There are no errors reported, and the 2 channels sound like they should. I have heard that channels must be unmuted in some cases. In AlsaMixer and Puppy's "Retrovol" (which mirrors AlsaMixer), I have set Master, PCM, and Surround to max volume, unmuted. Could there be a mixer in PortAudio that I also have to unmute? I can go back and forth between running speaker-test properly and running the modified paex_sine example and hearing only 2 channels. Here is paex_sine.c that I have modified:
/** @file paex_sine.c
@ingroup examples_src
@brief Play a sine wave for several seconds.
@author Ross Bencina <[email protected]>
@author Phil Burk <[email protected]>
*/
/*
* $Id: paex_sine.c 1752 2011-09-08 03:21:55Z philburk $
*
* This program uses the PortAudio Portable Audio Library.
* For more information see: http://www.portaudio.com/
* Copyright (c) 1999-2000 Ross Bencina and Phil Burk
*
* Permission is hereby granted, free of charge, to any person obtaining
* a copy of this software and associated documentation files
* (the "Software"), to deal in the Software without restriction,
* including without limitation the rights to use, copy, modify, merge,
* publish, distribute, sublicense, and/or sell copies of the Software,
* and to permit persons to whom the Software is furnished to do so,
* subject to the following conditions:
*
* The above copyright notice and this permission notice shall be
* included in all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
* EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
* MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
* IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR
* ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF
* CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION
* WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
*/
/*
* The text above constitutes the entire PortAudio license; however,
* the PortAudio community also makes the following non-binding requests:
*
* Any person wishing to distribute modifications to the Software is
* requested to send the modifications to the original developer so that
* they can be incorporated into the canonical version. It is also
* requested that these non-binding requests be included along with the
* license above.
*/
#include <stdio.h>
#include <math.h>
#include "portaudio.h"
#define NUM_SECONDS (30)
#define SAMPLE_RATE (44100)
#define FRAMES_PER_BUFFER (192)//(64)
#ifndef M_PI
#define M_PI (3.14159265)
#endif
#define TABLE_SIZE (200)
typedef struct
{
float sine[TABLE_SIZE];
int left_phase;
int right_phase;
int left2_phase;
int right2_phase;
int left3_phase;
int right3_phase;
char message[20];
}
paTestData;
/* This routine will be called by the PortAudio engine when audio is needed.
** It may called at interrupt level on some machines so don't do anything
** that could mess up the system like calling malloc() or free().
*/
static int patestCallback( const void *inputBuffer, void *outputBuffer,
unsigned long framesPerBuffer,
const PaStreamCallbackTimeInfo* timeInfo,
PaStreamCallbackFlags statusFlags,
void *userData )
{
paTestData *data = (paTestData*)userData;
float *out = (float*)outputBuffer;
unsigned long i;
(void) timeInfo; /* Prevent unused variable warnings. */
(void) statusFlags;
(void) inputBuffer;
for( i=0; i<framesPerBuffer; i++ )
{
*out++ = data->sine[data->left_phase]; /* left */
*out++ = data->sine[data->right_phase]; /* right */
*out++ = data->sine[data->left2_phase]; /* left */
*out++ = data->sine[data->right2_phase]; /* right */
*out++ = data->sine[data->left3_phase]; /* left */
*out++ = data->sine[data->right3_phase]; /* right */
data->left_phase += 1;
if( data->left_phase >= TABLE_SIZE ) data->left_phase -= TABLE_SIZE;
data->right_phase += 3; /* higher pitch so we can distinguish left and right. */
if( data->right_phase >= TABLE_SIZE ) data->right_phase -= TABLE_SIZE;
data->left2_phase += 5;
if( data->left2_phase >= TABLE_SIZE ) data->left2_phase -= TABLE_SIZE;
data->right2_phase += 7; /* higher pitch so we can distinguish left and right. */
if( data->right2_phase >= TABLE_SIZE ) data->right2_phase -= TABLE_SIZE;
data->left3_phase += 9;
if( data->left3_phase >= TABLE_SIZE ) data->left3_phase -= TABLE_SIZE;
data->right3_phase += 11; /* higher pitch so we can distinguish left and right. */
if( data->right3_phase >= TABLE_SIZE ) data->right3_phase -= TABLE_SIZE;
}
return paContinue;
}
/*
* This routine is called by portaudio when playback is done.
*/
static void StreamFinished( void* userData )
{
paTestData *data = (paTestData *) userData;
printf( "Stream Completed: %s\n", data->message );
}
/*******************************************************************/
int main(void);
int main(void)
{
PaStreamParameters outputParameters;
PaStream *stream;
PaError err;
paTestData data;
int i;
printf("PortAudio Test: output sine wave. SR = %d, BufSize = %d\n", SAMPLE_RATE, FRAMES_PER_BUFFER);
/* initialise sinusoidal wavetable */
for( i=0; i<TABLE_SIZE; i++ )
{
data.sine[i] = (float) sin( ((double)i/(double)TABLE_SIZE) * M_PI * 2. );
}
data.left_phase = data.right_phase = 0;
data.left2_phase = data.right2_phase = 0;
data.left3_phase = data.right3_phase = 0;
err = Pa_Initialize();
if( err != paNoError ) goto error;
outputParameters.device = Pa_GetDefaultOutputDevice(); /* default output device */
if (outputParameters.device == paNoDevice) {
fprintf(stderr,"Error: No default output device.\n");
goto error;
}
outputParameters.channelCount = 6; /* 5.1 Channel Output */
outputParameters.sampleFormat = paFloat32; /* 32 bit floating point output */
outputParameters.suggestedLatency = Pa_GetDeviceInfo( outputParameters.device )->defaultLowOutputLatency;
outputParameters.hostApiSpecificStreamInfo = NULL;
err = Pa_OpenStream(
&stream,
NULL, /* no input */
&outputParameters,
SAMPLE_RATE,
FRAMES_PER_BUFFER,
paClipOff, /* we won't output out of range samples so don't bother clipping them */
patestCallback,
&data );
if( err != paNoError ) goto error;
sprintf( data.message, "No Message" );
err = Pa_SetStreamFinishedCallback( stream, &StreamFinished );
if( err != paNoError ) goto error;
err = Pa_StartStream( stream );
if( err != paNoError ) goto error;
printf("Play for %d seconds.\n", NUM_SECONDS );
Pa_Sleep( NUM_SECONDS * 1000 );
err = Pa_StopStream( stream );
if( err != paNoError ) goto error;
err = Pa_CloseStream( stream );
if( err != paNoError ) goto error;
Pa_Terminate();
printf("Test finished.\n");
return err;
error:
Pa_Terminate();
fprintf( stderr, "An error occured while using the portaudio stream\n" );
fprintf( stderr, "Error number: %d\n", err );
fprintf( stderr, "Error message: %s\n", Pa_GetErrorText( err ) );
return err;
}
Without the plug:, there will be no automatic resampling.
PortAudio does not allow to set your own device name, so you would have to define you own device in ~/.asoundrc or /etc/asound.conf, like this:
pcm.mydevice = "plug:surround51"
and select that in PortAudio (search for it with Pa_GetDeviceCount/Pa_GetDeviceInfo).
Alternatively, make it the default device with:
pcm.!default = "plug:surround51"
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