From: drowe67 <drowe67@01035d8c-6547-0410-b346-abe4f91aad63>
Date: Sun, 8 Jul 2012 05:30:51 +0000 (+0000)
Subject: moved portaudio test programs so they dont break main automake system when portaudio... 
X-Git-Url: http://git.whiteaudio.com/gitweb/?a=commitdiff_plain;h=d8e80376cb385289031366d0735ff637b9b5beb3;p=freetel-svn-tracking.git

moved portaudio test programs so they dont break main automake system when portaudio library is absent

git-svn-id: https://svn.code.sf.net/p/freetel/code@586 01035d8c-6547-0410-b346-abe4f91aad63
---

diff --git a/codec2-dev/portaudio/Makefile b/codec2-dev/portaudio/Makefile
new file mode 100644
index 00000000..6c1e2c12
--- /dev/null
+++ b/codec2-dev/portaudio/Makefile
@@ -0,0 +1,20 @@
+# Makefile for Portaudio test programs
+# Requires Portaudio V19
+
+CFLAGS =  -g -Wall -I../src
+LIBS = -lm -lportaudio -pthread
+SRC  = ../src/fdmdv.c ../src/kiss_fft.c
+
+all: pa_rec pa_play pa_recplay
+
+pa_rec: Makefile pa_rec.c $(SRC)
+	gcc $(CFLAGS) pa_rec.c $(SRC) -o pa_rec $(LIBS)
+
+pa_play: Makefile pa_play.c $(SRC)
+	gcc $(CFLAGS) pa_play.c $(SRC) -o pa_play $(LIBS)
+
+pa_recplay: Makefile pa_recplay.c $(SRC)
+	gcc $(CFLAGS) pa_recplay.c $(SRC) -o pa_recplay $(LIBS)
+
+clean:
+	rm -f pa_rec pa_play pa_recplay
diff --git a/codec2-dev/portaudio/pa_play.c b/codec2-dev/portaudio/pa_play.c
new file mode 100644
index 00000000..00ea0868
--- /dev/null
+++ b/codec2-dev/portaudio/pa_play.c
@@ -0,0 +1,199 @@
+/* 
+   pa_play.c
+   David Rowe
+   July 8 2012
+
+   Converts samples from a 16 bit short 8000 Hz rawfile to 480000Hz
+   sample rate and plays them 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 <assert.h>
+#include <stdio.h>
+#include <stdlib.h>
+#include "portaudio.h"
+#include "fdmdv.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 */
+
+/* state information passed to call back */
+
+typedef struct {
+    FILE               *fin;
+    float               in8k[MEM8 + N8];
+} 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 playCallback( const void *inputBuffer, void *outputBuffer,
+                           unsigned long framesPerBuffer,
+                           const PaStreamCallbackTimeInfo* timeInfo,
+                           PaStreamCallbackFlags statusFlags,
+                           void *userData )
+{
+    paTestData *data = (paTestData*)userData;
+    FILE       *fin = data->fin;
+    int         i, nread;
+    int         finished;
+    short      *wptr = (short*)outputBuffer;
+    float      *in8k = data->in8k;
+    float       out48k[N48];
+    short       out48k_short[N48];
+    short       in8k_short[N8];
+
+    (void) outputBuffer; /* Prevent unused variable warnings. */
+    (void) timeInfo;
+    (void) statusFlags;
+    (void) userData;
+
+    /* note Portaudio docs recs. against making systems calls like
+       fwrite() in this callback but seems to work OK */
+    
+    nread = fread(in8k_short, sizeof(short), N8, fin);
+    if (nread == N8)
+	finished = paContinue;
+    else
+	finished = paComplete;
+
+    for(i=0; i<N8; i++)
+	in8k[MEM8+i] = in8k_short[i];
+
+    /* upsample and update filter memory */
+
+    fdmdv_8_to_48(out48k, &in8k[MEM8], N8);
+    for(i=0; i<MEM8; i++)
+	in8k[i] = in8k[i+N8];
+
+    assert(outputBuffer != NULL);
+
+    /* write signal to both channels */
+
+    for(i=0; i<N48; i++)
+	out48k_short[i] = (short)out48k[i];
+    for(i=0; i<framesPerBuffer; i++,wptr+=2) {
+	wptr[0] = out48k_short[i]; 
+	wptr[1] = out48k_short[i]; 
+    }
+
+    return finished;
+}
+
+int main(int argc, char *argv[])
+{
+    PaStreamParameters  outputParameters;
+    PaStream*           stream;
+    PaError             err = paNoError;
+    paTestData          data;
+    int                 i;
+
+    if (argc != 2) {
+	printf("usage: %s rawFile\n", argv[0]);
+	exit(0);
+    }
+
+    data.fin = fopen(argv[1], "rt");
+    if (data.fin == NULL) {
+	printf("Error opening input raw file %s\n", argv[1]);
+	exit(1);
+    }
+
+    for(i=0; i<MEM8; i++)
+	data.in8k[i] = 0.0;
+
+    err = Pa_Initialize();
+    if( err != paNoError ) goto done;
+
+    outputParameters.device = Pa_GetDefaultOutputDevice(); /* default input device */
+    if (outputParameters.device == paNoDevice) {
+        fprintf(stderr,"Error: No default output device.\n");
+        goto done;
+    }
+    outputParameters.channelCount = NUM_CHANNELS;         /* stereo input */
+    outputParameters.sampleFormat = paInt16;
+    outputParameters.suggestedLatency = Pa_GetDeviceInfo( outputParameters.device )->defaultLowOutputLatency;
+    outputParameters.hostApiSpecificStreamInfo = NULL;
+
+    /* Play some audio --------------------------------------------- */
+
+    err = Pa_OpenStream(
+              &stream,
+	      NULL,
+              &outputParameters,
+              SAMPLE_RATE,
+              N48,
+              paClipOff,      
+              playCallback,
+              &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;
+
+    fclose(data.fin);
+
+
+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. */
+    }
+    return err;
+}
+
diff --git a/codec2-dev/portaudio/pa_rec.c b/codec2-dev/portaudio/pa_rec.c
new file mode 100644
index 00000000..79366621
--- /dev/null
+++ b/codec2-dev/portaudio/pa_rec.c
@@ -0,0 +1,210 @@
+/* 
+   pa_rec.c
+   David Rowe
+   July 6 2012
+
+   Records at 48000 Hz from default sound device, convertes to 8 kHz,
+   and saves to raw file.  Used to get experience with Portaudio.
+
+   Modified from paex_record.c Portaudio example. Original author
+   author Phil Burk http://www.softsynth.com
+
+   To Build:
+
+     gcc paex_rec.c -o paex_rec -lm -lrt -lportaudio -pthread
+*/
+
+/*
+ * $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 <assert.h>
+#include <stdio.h>
+#include <stdlib.h>
+#include "portaudio.h"
+#include "fdmdv.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 NUM_CHANNELS 2             /* I think most sound cards prefer
+				      stereo, we will convert to mono
+				      as we sample */
+
+/* state information passed to call back */
+
+typedef struct {
+    FILE               *fout;
+    int                 framesLeft;
+    float               in48k[FDMDV_OS_TAPS + N48];
+} paTestData;
+
+
+/* 
+   This routine will be called by the PortAudio engine when audio is
+   available.  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;
+    FILE       *fout = data->fout;
+    int         framesToCopy;
+    int         i, n8;
+    int         finished;
+    short      *rptr = (short*)inputBuffer;
+    float       out8k[N8];
+    short       out8k_short[N8];
+
+    (void) outputBuffer; /* Prevent unused variable warnings. */
+    (void) timeInfo;
+    (void) statusFlags;
+    (void) userData;
+
+    if (data->framesLeft < framesPerBuffer) {
+        framesToCopy = data->framesLeft;
+        finished = paComplete;
+    } 
+    else {
+        framesToCopy = framesPerBuffer;
+        finished = paContinue;
+    }
+    data->framesLeft -= framesToCopy;
+
+    assert(inputBuffer != NULL);
+
+    /* just use left channel */
+
+    for(i=0; i<framesToCopy; i++,rptr+=2)
+	data->in48k[i+FDMDV_OS_TAPS] = *rptr; 
+
+    /* downsample and update filter memory */
+
+    fdmdv_48_to_8(out8k, &data->in48k[FDMDV_OS_TAPS], N8);
+    for(i=0; i<FDMDV_OS_TAPS; i++)
+	data->in48k[i] = data->in48k[i+framesToCopy];
+
+    /* save 8k to disk  */
+
+    for(i=0; i<N8; i++)
+	out8k_short[i] = (short)out8k[i];
+
+    /* note Portaudio docs recs. against making systems calls like
+       fwrite() in this callback but seems to work OK */
+    
+    fwrite(out8k_short, sizeof(short), N8, fout);
+
+    return finished;
+}
+
+int main(int argc, char *argv[])
+{
+    PaStreamParameters  inputParameters,
+                        outputParameters;
+    PaStream*           stream;
+    PaError             err = paNoError;
+    paTestData          data;
+    int                 i;
+    int                 numSecs;
+
+    if (argc != 3) {
+	printf("usage: %s rawFile time(s)\n", argv[0]);
+	exit(0);
+    }
+
+    data.fout = fopen(argv[1], "wt");
+    if (data.fout == NULL) {
+	printf("Error opening output raw file %s\n", argv[1]);
+	exit(1);
+    }
+
+    numSecs = atoi(argv[2]);
+    data.framesLeft = numSecs * SAMPLE_RATE;
+
+    for(i=0; i<FDMDV_OS_TAPS; i++)
+	data.in48k[i] = 0.0;
+
+    err = Pa_Initialize();
+    if( err != paNoError ) goto done;
+
+    inputParameters.device = Pa_GetDefaultInputDevice(); /* default input device */
+    if (inputParameters.device == paNoDevice) {
+        fprintf(stderr,"Error: No default input device.\n");
+        goto done;
+    }
+    inputParameters.channelCount = NUM_CHANNELS;         /* stereo input */
+    inputParameters.sampleFormat = paInt16;
+    inputParameters.suggestedLatency = Pa_GetDeviceInfo( inputParameters.device )->defaultLowInputLatency;
+    inputParameters.hostApiSpecificStreamInfo = NULL;
+
+    /* Record some audio --------------------------------------------- */
+
+    err = Pa_OpenStream(
+              &stream,
+              &inputParameters,
+              NULL,                  /* &outputParameters, */
+              SAMPLE_RATE,
+              N48,
+              paClipOff,      /* we won't output out of range samples so don't bother clipping them */
+              recordCallback,
+              &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;
+
+    fclose(data.fout);
+
+
+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. */
+    }
+    return err;
+}
+
diff --git a/codec2-dev/portaudio/pa_recplay.c b/codec2-dev/portaudio/pa_recplay.c
new file mode 100644
index 00000000..f745396f
--- /dev/null
+++ b/codec2-dev/portaudio/pa_recplay.c
@@ -0,0 +1,203 @@
+/* 
+   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 <assert.h>
+#include <stdio.h>
+#include <stdlib.h>
+#include "portaudio.h"
+#include "fdmdv.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 */
+
+/* state information passed to call back */
+
+typedef struct {
+    float               in48k[FDMDV_OS_TAPS + N48];
+    float               in8k[MEM8 + N8];
+} 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];
+
+    (void) timeInfo;
+    (void) statusFlags;
+
+    assert(inputBuffer != NULL);
+
+    /* just use left channel */
+
+    for(i=0; i<framesPerBuffer; i++,rptr+=2)
+	data->in48k[i+FDMDV_OS_TAPS] = *rptr; 
+
+    /* downsample and update filter memory */
+
+    fdmdv_48_to_8(out8k, &in48k[FDMDV_OS_TAPS], N8);
+    for(i=0; i<FDMDV_OS_TAPS; i++)
+	in48k[i] = in48k[i+framesPerBuffer];
+
+    /* play side, back up to 8k */
+
+    for(i=0; i<N8; i++)
+	in8k[MEM8+i] = out8k[i];
+
+    /* upsample and update filter memory */
+
+    fdmdv_8_to_48(out48k, &in8k[MEM8], N8);
+    for(i=0; i<MEM8; i++)
+	in8k[i] = in8k[i+N8];
+
+    assert(outputBuffer != NULL);
+
+    /* write signal to both channels */
+
+    for(i=0; i<N48; i++)
+	out48k_short[i] = (short)out48k[i];
+    for(i=0; i<framesPerBuffer; i++,wptr+=2) {
+	wptr[0] = out48k_short[i]; 
+	wptr[1] = out48k_short[i]; 
+    }
+
+    return paContinue;
+}
+
+int main(int argc, char *argv[])
+{
+    PaStreamParameters  inputParameters, outputParameters;
+    PaStream*           stream;
+    PaError             err = paNoError;
+    paTestData          data;
+    int                 i;
+
+    for(i=0; i<MEM8; i++)
+	data.in8k[i] = 0.0;
+    for(i=0; i<FDMDV_OS_TAPS; i++)
+	data.in48k[i] = 0.0;
+
+    err = Pa_Initialize();
+    if( err != paNoError ) goto done;
+
+    inputParameters.device = Pa_GetDefaultInputDevice(); /* default input device */
+    if (inputParameters.device == paNoDevice) {
+        fprintf(stderr,"Error: No default input device.\n");
+        goto done;
+    }
+    inputParameters.channelCount = NUM_CHANNELS;         /* stereo input */
+    inputParameters.sampleFormat = paInt16;
+    inputParameters.suggestedLatency = Pa_GetDeviceInfo( inputParameters.device )->defaultLowInputLatency;
+    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,
+              N48,
+              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. */
+    }
+    return err;
+}
+