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|
/*
* $Id$
* patest_record.c
* Record input into an array.
* Save array to a file.
* Playback recorded data.
*
* Author: Phil Burk <a href="http://www.softsynth.com" target="_blank">http://www.softsynth.com</a>
*
* This program uses the PortAudio Portable Audio Library.
* For more information see: <a href="http://www.portaudio.com" target="_blank">http://www.portaudio.com</a>
* 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.
*
* 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.
*
* 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 <stdio.h>
#include <stdlib.h>
#include "portaudio.h"
/* #define SAMPLE_RATE (17932) /* Test failure to open with this value. */
#define SAMPLE_RATE 44100
#define NUM_SECONDS 5
#define NUM_CHANNELS 2
/* #define DITHER_FLAG (paDitherOff) /**/
#define DITHER_FLAG (0) /**/
/* Select sample format. */
#define PA_SAMPLE_TYPE paFloat32
typedef float SAMPLE;
#define SAMPLE_SILENCE (0.0f)
typedef struct
{
int frameIndex; /* Index into sample array. */
int maxFrameIndex;
SAMPLE *recordedSamples;
}
paTestData;
/* This routine will be called by the PortAudio engine when audio is needed.
** 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 recordCallback( const void *inputBuffer,
void *outputBuffer,
unsigned long framesPerBuffer,
const PaStreamCallbackTimeInfo* timeInfo,
PaStreamCallbackFlags statusFlags,
void *userData)
{
paTestData *data = (paTestData*)userData;
SAMPLE *rptr = (SAMPLE*)inputBuffer;
SAMPLE *wptr = &data->recordedSamples[data->frameIndex * NUM_CHANNELS];
long framesToCalc;
long i;
int finished;
unsigned long framesLeft = data->maxFrameIndex - data->frameIndex;
(void) outputBuffer; /* Prevent unused variable warnings. */
(void) timeInfo;
if( framesLeft < framesPerBuffer )
{
framesToCalc = framesLeft;
finished = 1;
}
else
{
framesToCalc = framesPerBuffer;
finished = 0;
}
if( inputBuffer == NULL )
{
for( i=0; i<framesToCalc; i++ )
{
*wptr++ = SAMPLE_SILENCE; /* left */
if( NUM_CHANNELS == 2 ) *wptr++ = SAMPLE_SILENCE; /* right */
}
}
else
{
for( i=0; i<framesToCalc; i++ )
{
*wptr++ = *rptr++; /* left */
if( NUM_CHANNELS == 2 ) *wptr++ = *rptr++; /* right */
}
}
data->frameIndex += framesToCalc;
return finished;
}
static int liveCallback( const void *inputBuffer,
void *outputBuffer,
unsigned long framesPerBuffer,
const PaStreamCallbackTimeInfo* timeInfo,
PaStreamCallbackFlags statusFlags,
void *userData)
{
SAMPLE *rptr = (SAMPLE*)inputBuffer;
SAMPLE *wptr = (SAMPLE*)outputBuffer;
(void) outputBuffer; /* Prevent unused variable warnings. */
(void) timeInfo;
for( int i=0; i<framesPerBuffer; i++ )
{
*wptr++ = *rptr++; /* left */
if( NUM_CHANNELS == 2 ) *wptr++ = *rptr++; /* right */
}
return 0;
}
/* This routine will be called by the PortAudio engine when audio is needed.
** 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 playCallback( const void *inputBuffer,
void *outputBuffer,
unsigned long framesPerBuffer,
const PaStreamCallbackTimeInfo* timeInfo,
PaStreamCallbackFlags statusFlags,
void *userData )
{
paTestData *data = (paTestData*)userData;
SAMPLE *rptr = &data->recordedSamples[data->frameIndex * NUM_CHANNELS];
SAMPLE *wptr = (SAMPLE*)outputBuffer;
unsigned int i;
int finished;
unsigned int framesLeft = data->maxFrameIndex - data->frameIndex;
(void) inputBuffer; /* Prevent unused variable warnings. */
(void) timeInfo;
if( framesLeft < framesPerBuffer )
{
/* final buffer... */
for( i=0; i<framesLeft; i++ )
{
*wptr++ = *rptr++; /* left */
if( NUM_CHANNELS == 2 ) *wptr++ = *rptr++; /* right */
}
for( ; i<framesPerBuffer; i++ )
{
*wptr++ = 0; /* left */
if( NUM_CHANNELS == 2 ) *wptr++ = 0; /* right */
}
data->frameIndex += framesLeft;
finished = 1;
}
else
{
for( i=0; i<framesPerBuffer; i++ )
{
*wptr++ = *rptr++; /* left */
if( NUM_CHANNELS == 2 ) *wptr++ = *rptr++; /* right */
}
data->frameIndex += framesPerBuffer;
finished = 0;
}
return finished;
}
/*******************************************************************/
int main(void)
{
PaStream *stream;
PaError err;
paTestData data;
int i;
int totalFrames;
int numSamples;
int numBytes;
data.maxFrameIndex = totalFrames = NUM_SECONDS * SAMPLE_RATE; /* Record for a
few seconds. */
data.frameIndex = 0;
numSamples = totalFrames * NUM_CHANNELS;
numBytes = numSamples * sizeof(SAMPLE);
data.recordedSamples = (SAMPLE *) malloc( numBytes );
if( data.recordedSamples == NULL )
{
printf("Could not allocate record array.\n");
exit(1);
}
for( i=0; i<numSamples; i++ ) data.recordedSamples[i] = 0;
err = Pa_Initialize();
if( err != paNoError ) goto error;
/* Record some audio. -------------------------------------------- */
err = Pa_OpenDefaultStream( &stream,
2, /* no input channels */
2, /* stereo output */
PA_SAMPLE_TYPE, /* 32 bit floating point output
*/
SAMPLE_RATE,
paFramesPerBufferUnspecified, /* frames
per buffer, i.e. the number
of sample frames that
PortAudio will
request from the callback.
Many apps
may want to use
paFramesPerBufferUnspecified,
which
tells PortAudio to pick the
best,
possibly changing, buffer
size.*/
liveCallback, /* this is your callback function */
0 ); /*This is a pointer that will be passed to
//&data your callback*/
if( err != paNoError ) goto error;
err = Pa_StartStream( stream );
if( err != paNoError ) goto error;
printf("Now recording!!\n"); fflush(stdout);
while (1);
//Pa_Sleep(1000 * NUM_SECONDS);
err = Pa_CloseStream( stream );
if( err != paNoError ) goto error;
/* Write recorded data to a file. */
#if 0
{
FILE *fid;
fid = fopen("recorded.raw", "wb");
if( fid == NULL )
{
printf("Could not open file.");
}
else
{
fwrite( data.recordedSamples, NUM_CHANNELS * sizeof(SAMPLE),
totalFrames, fid );
fclose( fid );
printf("Wrote data to 'recorded.raw'\n");
}
}
#endif
/* Playback recorded data. -------------------------------------------- */
data.frameIndex = 0;
printf("Begin playback.\n"); fflush(stdout);
err =Pa_OpenDefaultStream( &stream,
0, /* no input channels */
2, /* stereo output */
PA_SAMPLE_TYPE, /* 32 bit floating point output
*/
SAMPLE_RATE,
paFramesPerBufferUnspecified, /* frames
per buffer, i.e. the number
of sample frames that
PortAudio will
request from the callback.
Many apps
may want to use
paFramesPerBufferUnspecified,
which
tells PortAudio to pick the
best,
possibly changing, buffer
size.*/
playCallback, /* this is your callback function */
&data ); /*This is a pointer that will be passed
to
your callback*/
if( err != paNoError ) goto error;
if( stream )
{
err = Pa_StartStream( stream );
if( err != paNoError ) goto error;
printf("Waiting for playback to finish.\n"); fflush(stdout);
Pa_Sleep(1000 * NUM_SECONDS);
err = Pa_CloseStream( stream );
if( err != paNoError ) goto error;
printf("Done.\n"); fflush(stdout);
}
free( data.recordedSamples );
Pa_Terminate();
return 0;
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 -1;
}
**********************************************************************
Pour l'encodage(fichier sampleenc.c qui se trouve dans la doc de speex.org),
faites en ligne de commande: "gcc -lspeex -o encode sampleenc.c":
#include <speex/speex.h>
#include <stdio.h>
/*The frame size in hardcoded for this sample code but it doesn't have to be*/
#define FRAME_SIZE 160
int main(int argc, char **argv)
{
char *inFile;
FILE *fin;
FILE *fin2;
short in[FRAME_SIZE];
float input[FRAME_SIZE];
printf("FRAME_SIZE: %i\n", FRAME_SIZE);
char cbits[4096];
int nbBytes;
/*Holds the state of the encoder*/
void *state;
/*Holds bits so they can be read and written to by the Speex routines*/
SpeexBits bits;
int i, tmp;
if (argc != 2)
{
printf("Warning: 2 arguments needed!\n");
return 0;
}
/*Create a new encoder state in narrowband mode*/
state = speex_encoder_init(&speex_nb_mode);
/*Set the quality to 8 (15 kbps)*/
tmp=8;
speex_encoder_ctl(state, SPEEX_SET_QUALITY, &tmp);
inFile = argv[1];
//fin = fopen(inFile, "r");
if ((fin = fopen(inFile, "r")) == NULL)
{
printf("Warning: Cannot open file!\n");
return 0;
}
//fin2 = fopen("test.flux", "w");
if ((fin2 = fopen("test.flux", "w")) == NULL)
{
printf("Warning: Cannot open file!\n");
return 0;
}
/*Initialization of the structure that holds the bits*/
speex_bits_init(&bits);
while (1)
{
/*Read a 16 bits/sample audio frame*/
fread(in, sizeof(short), FRAME_SIZE, fin);
if (feof(fin))
break;
/*Copy the 16 bits values to float so Speex can work on them*/
for (i=0;i<FRAME_SIZE;i++)
input[i]=in[i];
/*Flush all the bits in the struct so we can encode a new frame*/
speex_bits_reset(&bits);
/*Encode the frame*/
speex_encode(state, input, &bits);
/*Copy the bits to an array of char that can be written*/
nbBytes = speex_bits_write(&bits, cbits, 200);
/*Write the size of the frame first. This is what sampledec expects but
it's likely to be different in your own application*/
fwrite(&nbBytes, sizeof(int), 1, fin2);
/*Write the compressed data*/
fwrite(cbits, 1, nbBytes, fin2);
}
/*Destroy the encoder state*/
speex_encoder_destroy(state);
/*Destroy the bit-packing struct*/
speex_bits_destroy(&bits);
fclose(fin);
return 0;
}
*********************************************************************************
Pour le decodage(fichier sampledec.c qui se trouve dans la doc de speex.org),
faites en ligne de commande: "gcc -lspeex -o decode sampledec.c":
#include <speex/speex.h>
#include <stdio.h>
/*The frame size in hardcoded for this sample code but it doesn't have to be*/
#define FRAME_SIZE 160
int main(int argc, char **argv)
{
char *outFile;
FILE *fout;
FILE *fout2;
/*Holds the audio that will be written to file (16 bits per sample)*/
short out[FRAME_SIZE];
/*Speex handle samples as float, so we need an array of floats*/
float output[FRAME_SIZE];
char cbits[4096];
int nbBytes;
/*Holds the state of the decoder*/
void *state;
/*Holds bits so they can be read and written to by the Speex routines*/
SpeexBits bits;
int i, tmp;
if (argc != 2)
{
printf("Warning: 2 arguments needed!\n");
return 0;
}
/*Create a new decoder state in narrowband mode*/
state = speex_decoder_init(&speex_nb_mode);
/*Set the perceptual enhancement on*/
tmp=8;
speex_decoder_ctl(state, SPEEX_SET_ENH, &tmp);
outFile = argv[1];
if ((fout = fopen(outFile, "r+")) == NULL)
{
printf("Warning: Cannot open file!\n");
return 0;
}
if ((fout2 = fopen("newFile.mp3", "w+")) == NULL)
{
printf("Warning: Cannot open file!\n");
return 0;
}
/*Initialization of the structure that holds the bits*/
speex_bits_init(&bits);
//printf("FRAME SIZE: %i\n", FRAME_SIZE);
while (1)
{
/*Read the size encoded by sampleenc, this part will likely be
different in your application*/
fread(&nbBytes, sizeof(int), 1, fout);
fprintf (stderr, "nbBytes: %d\n", nbBytes);
if (feof(fout))
break;
/*Read the "packet" encoded by sampleenc*/
fread(cbits, 1, nbBytes, fout);
/*Copy the data into the bit-stream struct*/
speex_bits_read_from(&bits, cbits, nbBytes);
/*Decode the data*/
speex_decode(state, &bits, output);
/*Copy from float to short (16 bits) for output*/
for (i=0;i<FRAME_SIZE;i++)
out[i]=output[i];
/*Write the decoded audio to file*/
fwrite(out, sizeof(short), FRAME_SIZE, fout2);
}
/*Destroy the decoder state*/
speex_decoder_destroy(state);
/*Destroy the bit-stream truct*/
speex_bits_destroy(&bits);
fclose(fout);
fclose(fout2);
return 0;
}
****************************************************
**************************************************** |
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