/* pa_recplay.c David Rowe July 8 2012 Echos audio from sound card input to sound card output. Samples at 48 kHz, converts to 8 kHz, converts back to 48kHz, and plays using the default sound device. Used as an intermediate step in Portaudio integration. Modified from paex_record.c Portaudio example. Original author author Phil Burk http://www.softsynth.com */ /* * $Id: paex_record.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. */ #include #include #include #include "portaudio.h" #include "fdmdv.h" #include "fifo.h" #define SAMPLE_RATE 48000 /* 48 kHz sampling rate rec. as we can trust accuracy of sound card */ #define N8 160 /* processing buffer size at 8 kHz */ #define N48 (N8*FDMDV_OS) /* processing buffer size at 48 kHz */ #define MEM8 (FDMDV_OS_TAPS/FDMDV_OS) #define NUM_CHANNELS 2 /* I think most sound cards prefer stereo, we will convert to mono as we sample */ #define MAX_FPB 2048 /* maximum value of framesPerBuffer */ /* state information passed to call back */ typedef struct { float in48k[FDMDV_OS_TAPS + N48]; float in8k[MEM8 + N8]; struct FIFO *infifo; struct FIFO *outfifo; } paTestData; /* This routine will be called by the PortAudio engine when audio is required. It may be 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 callback( const void *inputBuffer, void *outputBuffer, unsigned long framesPerBuffer, const PaStreamCallbackTimeInfo* timeInfo, PaStreamCallbackFlags statusFlags, void *userData ) { paTestData *data = (paTestData*)userData; int i; short *rptr = (short*)inputBuffer; short *wptr = (short*)outputBuffer; float *in8k = data->in8k; float *in48k = data->in48k; float out8k[N8]; float out48k[N48]; short out48k_short[N48]; short in48k_short[N48]; short indata[MAX_FPB]; short outdata[MAX_FPB]; (void) timeInfo; (void) statusFlags; assert(inputBuffer != NULL); assert(outputBuffer != NULL); /* framesPerBuffer is portaudio-speak for number of samples we actually get from the record side and need to provide to the play side. On Linux (at least) it was found that framesPerBuffer may not always be what we ask for in the framesPerBuffer field of Pa_OpenStream. For example a request for 960 sample buffers lead to framesPerBuffer = 1024. To perform the 48 to 8 kHz conversion we need an integer multiple of FDMDV_OS samples to support the interpolation and decimation. As we can't guarantee the size of framesPerBuffer we do a little FIFO buffering. */ //printf("framesPerBuffer: %d N48 %d\n", framesPerBuffer, N48); /* assemble a mono buffer (just use left channel) and write to FIFO */ assert(framesPerBuffer < MAX_FPB); for(i=0; iinfifo, indata, framesPerBuffer); /* while we have enough samples available ... */ //printf("infifo before: %d\n", fifo_n(data->infifo)); while (fifo_read(data->infifo, in48k_short, N48) == 0) { /* convert to float */ for(i=0; ioutfifo, out48k_short, N48); } //printf("infifo after: %d\n", fifo_n(data->infifo)); //printf("outfifo : %d\n", fifo_n(data->outfifo)); /* OK now set up output samples */ if (fifo_read(data->outfifo, outdata, framesPerBuffer) == 0) { /* write signal to both channels */ for(i=0; idefaultLowInputLatency; inputParameters.hostApiSpecificStreamInfo = NULL; outputParameters.device = Pa_GetDefaultOutputDevice(); /* default output device */ if (outputParameters.device == paNoDevice) { fprintf(stderr,"Error: No default output device.\n"); goto done; } outputParameters.channelCount = NUM_CHANNELS; /* stereo output */ outputParameters.sampleFormat = paInt16; outputParameters.suggestedLatency = Pa_GetDeviceInfo( outputParameters.device )->defaultLowOutputLatency; outputParameters.hostApiSpecificStreamInfo = NULL; /* Play some audio --------------------------------------------- */ err = Pa_OpenStream( &stream, &inputParameters, &outputParameters, SAMPLE_RATE, 512, paClipOff, callback, &data ); if( err != paNoError ) goto done; err = Pa_StartStream( stream ); if( err != paNoError ) goto done; while( ( err = Pa_IsStreamActive( stream ) ) == 1 ) { Pa_Sleep(100); } if( err < 0 ) goto done; err = Pa_CloseStream( stream ); if( err != paNoError ) goto done; done: Pa_Terminate(); if( err != paNoError ) { 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 ) ); err = 1; /* Always return 0 or 1, but no other return codes. */ } fifo_destroy(data.infifo); fifo_destroy(data.outfifo); return err; }