//
// This file is part of Dire Wolf, an amateur radio packet TNC.
//
// Copyright (C) 2011, 2012, 2013, 2014, 2015 John Langner, WB2OSZ
//
// This program is free software: you can redistribute it and/or modify
// it under the terms of the GNU General Public License as published by
// the Free Software Foundation, either version 2 of the License, or
// (at your option) any later version.
//
// This program is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
// GNU General Public License for more details.
//
// You should have received a copy of the GNU General Public License
// along with this program. If not, see .
//
/*------------------------------------------------------------------
*
* Module: audio_win.c
*
* Purpose: Interface to audio device commonly called a "sound card" for
* historical reasons.
*
* This version uses the native Windows sound interface.
*
* Credits: Fabrice FAURE contributed Linux code for the SDR UDP interface.
*
* Discussion here: http://gqrx.dk/doc/streaming-audio-over-udp
*
* Major revisions:
*
* 1.2 - Add ability to use more than one audio device.
*
*---------------------------------------------------------------*/
#include
#include
#include
#include
#include
#include
#include
#include
#include
#include
#ifndef WAVE_FORMAT_96M16
#define WAVE_FORMAT_96M16 0x40000
#define WAVE_FORMAT_96S16 0x80000
#endif
#include
#define _WIN32_WINNT 0x0501
#include
#include "direwolf.h"
#include "audio.h"
#include "audio_stats.h"
#include "textcolor.h"
#include "ptt.h"
#include "demod.h" /* for alevel_t & demod_get_audio_level() */
/* Audio configuration. */
static struct audio_s *save_audio_config_p;
/*
* Allocate enough buffers for 1 second each direction.
* Each buffer size is a trade off between being responsive
* to activity on the channel vs. overhead of having too
* many little transfers.
*/
/*
* Originally, we had an abitrary buf time of 40 mS.
*
* For mono, the buffer size was rounded up from 3528 to 4k so
* it was really about 50 mS per buffer or about 20 per second.
* For stereo, the buffer size was rounded up from 7056 to 7k so
* it was really about 43.7 mS per buffer or about 23 per second.
*
* In version 1.2, let's try changing it to 10 to reduce the latency.
* For mono, the buffer size was rounded up from 882 to 1k so it
* was really about 12.5 mS per buffer or about 80 per second.
*/
#define TOTAL_BUF_TIME 1000
#define ONE_BUF_TIME 10
#define NUM_IN_BUF ((TOTAL_BUF_TIME)/(ONE_BUF_TIME))
#define NUM_OUT_BUF ((TOTAL_BUF_TIME)/(ONE_BUF_TIME))
#define roundup1k(n) (((n) + 0x3ff) & ~0x3ff)
static int calcbufsize(int rate, int chans, int bits)
{
int size1 = (rate * chans * bits / 8 * ONE_BUF_TIME) / 1000;
int size2 = roundup1k(size1);
#if DEBUG
text_color_set(DW_COLOR_DEBUG);
dw_printf ("audio_open: calcbufsize (rate=%d, chans=%d, bits=%d) calc size=%d, round up to %d\n",
rate, chans, bits, size1, size2);
#endif
/* Version 1.3 - add a sanity check. */
if (size2 < 256 || size2 > 32768) {
text_color_set(DW_COLOR_ERROR);
dw_printf ("Audio buffer has unexpected extreme size of %d bytes.\n", size2);
dw_printf ("Detected at %s, line %d.\n", __FILE__, __LINE__);
dw_printf ("This might be caused by unusual audio device configuration values.\n");
size2 = 2048;
dw_printf ("Using %d to attempt recovery.\n", size2);
}
return (size2);
}
/* Information for each audio stream (soundcard, stdin, or UDP) */
static struct adev_s {
enum audio_in_type_e g_audio_in_type;
/*
* UDP socket for receiving audio stream.
* Buffer, length, and pointer for UDP or stdin.
*/
SOCKET udp_sock;
char stream_data[SDR_UDP_BUF_MAXLEN];
int stream_len;
int stream_next;
/* For sound output. */
/* out_wavehdr.dwUser is used to keep track of output buffer state. */
#define DWU_FILLING 1 /* Ready to use or in process of being filled. */
#define DWU_PLAYING 2 /* Was given to sound system for playing. */
#define DWU_DONE 3 /* Sound system is done with it. */
HWAVEOUT audio_out_handle;
volatile WAVEHDR out_wavehdr[NUM_OUT_BUF];
int out_current; /* index to above. */
int outbuf_size;
/* For sound input. */
/* In this case dwUser is index of next available byte to remove. */
HWAVEIN audio_in_handle;
WAVEHDR in_wavehdr[NUM_IN_BUF];
volatile WAVEHDR *in_headp; /* head of queue to process. */
CRITICAL_SECTION in_cs;
} adev[MAX_ADEVS];
/*------------------------------------------------------------------
*
* Name: audio_open
*
* Purpose: Open the digital audio device.
*
* New in version 1.0, we recognize "udp:" optionally
* followed by a port number.
*
* Inputs: pa - Address of structure of type audio_s.
*
* Using a structure, rather than separate arguments
* seemed to make sense because we often pass around
* the same set of parameters various places.
*
* The fields that we care about are:
* num_channels
* samples_per_sec
* bits_per_sample
* If zero, reasonable defaults will be provided.
*
* Outputs: pa - The ACTUAL values are returned here.
*
* The Linux version adjusts strange values to the
* nearest valid value. Don't know, yet, if Windows
* does the same or just fails. Or performs some
* expensive resampling from a rate supported by
* hardware.
*
* These might not be exactly the same as what was requested.
*
* Example: ask for stereo, 16 bits, 22050 per second.
* An ordinary desktop/laptop PC should be able to handle this.
* However, some other sort of smaller device might be
* more restrictive in its capabilities.
* It might say, the best I can do is mono, 8 bit, 8000/sec.
*
* The sofware modem must use this ACTUAL information
* that the device is supplying, that could be different
* than what the user specified.
*
* Returns: 0 for success, -1 for failure.
*
* References: Multimedia Reference
*
* http://msdn.microsoft.com/en-us/library/windows/desktop/dd743606%28v=vs.85%29.aspx
*
*----------------------------------------------------------------*/
static void CALLBACK in_callback (HWAVEIN handle, UINT msg, DWORD instance, DWORD param1, DWORD param2);
static void CALLBACK out_callback (HWAVEOUT handle, UINT msg, DWORD instance, DWORD param1, DWORD param2);
int audio_open (struct audio_s *pa)
{
int a;
int err;
int chan;
int n;
int in_dev_no[MAX_ADEVS];
int out_dev_no[MAX_ADEVS];
int num_devices;
WAVEINCAPS wic;
WAVEOUTCAPS woc;
save_audio_config_p = pa;
for (a=0; aadev[a].defined) {
struct adev_s *A = &(adev[a]);
assert (A->audio_in_handle == 0);
assert (A->audio_out_handle == 0);
//text_color_set(DW_COLOR_DEBUG);
//dw_printf ("pa->adev[a].adevice_in = '%s'\n", pa->adev[a].adevice_in);
//dw_printf ("pa->adev[a].adevice_out = '%s'\n", pa->adev[a].adevice_out);
/*
* Fill in defaults for any missing values.
*/
if (pa -> adev[a].num_channels == 0)
pa -> adev[a].num_channels = DEFAULT_NUM_CHANNELS;
if (pa -> adev[a].samples_per_sec == 0)
pa -> adev[a].samples_per_sec = DEFAULT_SAMPLES_PER_SEC;
if (pa -> adev[a].bits_per_sample == 0)
pa -> adev[a].bits_per_sample = DEFAULT_BITS_PER_SAMPLE;
A->g_audio_in_type = AUDIO_IN_TYPE_SOUNDCARD;
for (chan=0; chan achan[chan].mark_freq == 0)
pa -> achan[chan].mark_freq = DEFAULT_MARK_FREQ;
if (pa -> achan[chan].space_freq == 0)
pa -> achan[chan].space_freq = DEFAULT_SPACE_FREQ;
if (pa -> achan[chan].baud == 0)
pa -> achan[chan].baud = DEFAULT_BAUD;
if (pa->achan[chan].num_subchan == 0)
pa->achan[chan].num_subchan = 1;
}
A->udp_sock = INVALID_SOCKET;
in_dev_no[a] = WAVE_MAPPER; /* = -1 */
out_dev_no[a] = WAVE_MAPPER;
/*
* Determine the type of audio input and select device.
* This can be soundcard, UDP stream, or stdin.
*/
if (strcasecmp(pa->adev[a].adevice_in, "stdin") == 0 || strcmp(pa->adev[a].adevice_in, "-") == 0) {
A->g_audio_in_type = AUDIO_IN_TYPE_STDIN;
/* Change - to stdin for readability. */
strlcpy (pa->adev[a].adevice_in, "stdin", sizeof(pa->adev[a].adevice_in));
}
else if (strncasecmp(pa->adev[a].adevice_in, "udp:", 4) == 0) {
A->g_audio_in_type = AUDIO_IN_TYPE_SDR_UDP;
/* Supply default port if none specified. */
if (strcasecmp(pa->adev[a].adevice_in,"udp") == 0 ||
strcasecmp(pa->adev[a].adevice_in,"udp:") == 0) {
snprintf (pa->adev[a].adevice_in, sizeof(pa->adev[a].adevice_in), "udp:%d", DEFAULT_UDP_AUDIO_PORT);
}
}
else {
A->g_audio_in_type = AUDIO_IN_TYPE_SOUNDCARD;
/* Does config file have a number? */
/* If so, it is an index into list of devices. */
if (strlen(pa->adev[a].adevice_in) == 1 && isdigit(pa->adev[a].adevice_in[0])) {
in_dev_no[a] = atoi(pa->adev[a].adevice_in);
}
/* Otherwise, does it have search string? */
if (in_dev_no[a] == WAVE_MAPPER && strlen(pa->adev[a].adevice_in) >= 1) {
num_devices = waveInGetNumDevs();
for (n=0 ; nadev[a].adevice_in) != NULL) {
in_dev_no[a] = n;
}
}
}
if (in_dev_no[a] == WAVE_MAPPER) {
text_color_set(DW_COLOR_ERROR);
dw_printf ("\"%s\" doesn't match any of the input devices.\n", pa->adev[a].adevice_in);
}
}
}
/*
* Select output device.
* Only soundcard at this point.
* Purhaps we'd like to add UDP for an SDR transmitter.
*/
if (strlen(pa->adev[a].adevice_out) == 1 && isdigit(pa->adev[a].adevice_out[0])) {
out_dev_no[a] = atoi(pa->adev[a].adevice_out);
}
if (out_dev_no[a] == WAVE_MAPPER && strlen(pa->adev[a].adevice_out) >= 1) {
num_devices = waveOutGetNumDevs();
for (n=0 ; nadev[a].adevice_out) != NULL) {
out_dev_no[a] = n;
}
}
}
if (out_dev_no[a] == WAVE_MAPPER) {
text_color_set(DW_COLOR_ERROR);
dw_printf ("\"%s\" doesn't match any of the output devices.\n", pa->adev[a].adevice_out);
}
}
} /* if defined */
} /* for each device */
/*
* Display the input devices (soundcards) available and what is selected.
*/
text_color_set(DW_COLOR_INFO);
dw_printf ("Available audio input devices for receive (*=selected):\n");
num_devices = waveInGetNumDevs();
for (a=0; aadev[a].defined) {
if (in_dev_no[a] < -1 || in_dev_no[a] >= num_devices) {
text_color_set(DW_COLOR_ERROR);
dw_printf ("Invalid input (receive) audio device number %d.\n", in_dev_no[a]);
in_dev_no[a] = WAVE_MAPPER;
}
}
}
text_color_set(DW_COLOR_INFO);
for (n=0; nadev[a].defined) {
dw_printf (" %c", n==in_dev_no[a] ? '*' : ' ');
}
}
dw_printf (" %d: %s", n, wic.szPname);
for (a=0; aadev[a].defined && n==in_dev_no[a]) {
if (pa->adev[a].num_channels == 2) {
dw_printf (" (channels %d & %d)", ADEVFIRSTCHAN(a), ADEVFIRSTCHAN(a)+1);
}
else {
dw_printf (" (channel %d)", ADEVFIRSTCHAN(a));
}
}
}
dw_printf ("\n");
}
}
// Add UDP or stdin to end of device list if used.
for (a=0; aadev[a].defined) {
struct adev_s *A = &(adev[a]);
/* Display stdin or udp:port if appropriate. */
if (A->g_audio_in_type != AUDIO_IN_TYPE_SOUNDCARD) {
int aaa;
for (aaa=0; aaaadev[aaa].defined) {
dw_printf (" %c", a == aaa ? '*' : ' ');
}
}
dw_printf (" %s ", pa->adev[a].adevice_in); /* should be UDP:nnnn or stdin */
if (pa->adev[a].num_channels == 2) {
dw_printf (" (channels %d & %d)", ADEVFIRSTCHAN(a), ADEVFIRSTCHAN(a)+1);
}
else {
dw_printf (" (channel %d)", ADEVFIRSTCHAN(a));
}
dw_printf ("\n");
}
}
}
/*
* Display the output devices (soundcards) available and what is selected.
*/
dw_printf ("Available audio output devices for transmit (*=selected):\n");
/* TODO? */
/* No "*" is currently displayed when using the default device. */
/* Should we put "*" next to the default device when using it? */
/* Which is the default? The first one? */
num_devices = waveOutGetNumDevs();
for (a=0; aadev[a].defined) {
if (out_dev_no[a] < -1 || out_dev_no[a] >= num_devices) {
text_color_set(DW_COLOR_ERROR);
dw_printf ("Invalid output (transmit) audio device number %d.\n", out_dev_no[a]);
out_dev_no[a] = WAVE_MAPPER;
}
}
}
text_color_set(DW_COLOR_INFO);
for (n=0; nadev[a].defined) {
dw_printf (" %c", n==out_dev_no[a] ? '*' : ' ');
}
}
dw_printf (" %d: %s", n, woc.szPname);
for (a=0; aadev[a].defined && n==out_dev_no[a]) {
if (pa->adev[a].num_channels == 2) {
dw_printf (" (channels %d & %d)", ADEVFIRSTCHAN(a), ADEVFIRSTCHAN(a)+1);
}
else {
dw_printf (" (channel %d)", ADEVFIRSTCHAN(a));
}
}
}
dw_printf ("\n");
}
}
/*
* Open for each audio device input/output pair.
*/
for (a=0; aadev[a].defined) {
struct adev_s *A = &(adev[a]);
WAVEFORMATEX wf;
wf.wFormatTag = WAVE_FORMAT_PCM;
wf.nChannels = pa -> adev[a].num_channels;
wf.nSamplesPerSec = pa -> adev[a].samples_per_sec;
wf.wBitsPerSample = pa -> adev[a].bits_per_sample;
wf.nBlockAlign = (wf.wBitsPerSample / 8) * wf.nChannels;
wf.nAvgBytesPerSec = wf.nBlockAlign * wf.nSamplesPerSec;
wf.cbSize = 0;
A->outbuf_size = calcbufsize(wf.nSamplesPerSec,wf.nChannels,wf.wBitsPerSample);
/*
* Open the audio output device.
* Soundcard is only possibility at this time.
*/
err = waveOutOpen (&(A->audio_out_handle), out_dev_no[a], &wf, (DWORD_PTR)out_callback, a, CALLBACK_FUNCTION);
if (err != MMSYSERR_NOERROR) {
text_color_set(DW_COLOR_ERROR);
dw_printf ("Could not open audio device for output.\n");
return (-1);
}
/*
* Set up the output buffers.
* We use dwUser to indicate it is available for filling.
*/
memset ((void*)(A->out_wavehdr), 0, sizeof(A->out_wavehdr));
for (n = 0; n < NUM_OUT_BUF; n++) {
A->out_wavehdr[n].lpData = malloc(A->outbuf_size);
A->out_wavehdr[n].dwUser = DWU_FILLING;
A->out_wavehdr[n].dwBufferLength = 0;
}
A->out_current = 0;
/*
* Open audio input device.
* More possibilities here: soundcard, UDP port, stdin.
*/
switch (A->g_audio_in_type) {
/*
* Soundcard.
*/
case AUDIO_IN_TYPE_SOUNDCARD:
InitializeCriticalSection (&(A->in_cs));
err = waveInOpen (&(A->audio_in_handle), in_dev_no[a], &wf, (DWORD_PTR)in_callback, a, CALLBACK_FUNCTION);
if (err != MMSYSERR_NOERROR) {
text_color_set(DW_COLOR_ERROR);
dw_printf ("Could not open audio device for input.\n");
return (-1);
}
/*
* Set up the input buffers.
*/
memset ((void*)(A->in_wavehdr), 0, sizeof(A->in_wavehdr));
for (n = 0; n < NUM_OUT_BUF; n++) {
A->in_wavehdr[n].dwBufferLength = A->outbuf_size; /* all the same size */
A->in_wavehdr[n].lpData = malloc(A->outbuf_size);
}
A->in_headp = NULL;
/*
* Give them to the sound input system.
*/
for (n = 0; n < NUM_OUT_BUF; n++) {
waveInPrepareHeader(A->audio_in_handle, &(A->in_wavehdr[n]), sizeof(WAVEHDR));
waveInAddBuffer(A->audio_in_handle, &(A->in_wavehdr[n]), sizeof(WAVEHDR));
}
/*
* Start it up.
* The callback function is called when one is filled.
*/
waveInStart (A->audio_in_handle);
break;
/*
* UDP.
*/
case AUDIO_IN_TYPE_SDR_UDP:
{
WSADATA wsadata;
struct sockaddr_in si_me;
//int slen=sizeof(si_me);
//int data_size = 0;
int err;
err = WSAStartup (MAKEWORD(2,2), &wsadata);
if (err != 0) {
text_color_set(DW_COLOR_ERROR);
dw_printf("WSAStartup failed: %d\n", err);
return (-1);
}
if (LOBYTE(wsadata.wVersion) != 2 || HIBYTE(wsadata.wVersion) != 2) {
text_color_set(DW_COLOR_ERROR);
dw_printf("Could not find a usable version of Winsock.dll\n");
WSACleanup();
return (-1);
}
// Create UDP Socket
A->udp_sock = socket(AF_INET, SOCK_DGRAM, IPPROTO_UDP);
if (A->udp_sock == INVALID_SOCKET) {
text_color_set(DW_COLOR_ERROR);
dw_printf ("Couldn't create socket, errno %d\n", WSAGetLastError());
return -1;
}
memset((char *) &si_me, 0, sizeof(si_me));
si_me.sin_family = AF_INET;
si_me.sin_port = htons((short)atoi(pa->adev[a].adevice_in + 4));
si_me.sin_addr.s_addr = htonl(INADDR_ANY);
// Bind to the socket
if (bind(A->udp_sock, (SOCKADDR *) &si_me, sizeof(si_me)) != 0) {
text_color_set(DW_COLOR_ERROR);
dw_printf ("Couldn't bind socket, errno %d\n", WSAGetLastError());
return -1;
}
A->stream_next= 0;
A->stream_len = 0;
}
break;
/*
* stdin.
*/
case AUDIO_IN_TYPE_STDIN:
setmode (STDIN_FILENO, _O_BINARY);
A->stream_next= 0;
A->stream_len = 0;
break;
default:
text_color_set(DW_COLOR_ERROR);
dw_printf ("Internal error, invalid audio_in_type\n");
return (-1);
}
}
}
return (0);
} /* end audio_open */
/*
* Called when input audio block is ready.
*/
static void CALLBACK in_callback (HWAVEIN handle, UINT msg, DWORD instance, DWORD param1, DWORD param2)
{
int a = instance;
//dw_printf ("in_callback, handle = %d, a = %d\n", (int)handle, a);
assert (a >= 0 && a < MAX_ADEVS);
struct adev_s *A = &(adev[a]);
if (msg == WIM_DATA) {
WAVEHDR *p = (WAVEHDR*)param1;
p->dwUser = -1; /* needs to be unprepared. */
p->lpNext = NULL;
EnterCriticalSection (&(A->in_cs));
if (A->in_headp == NULL) {
A->in_headp = p; /* first one in list */
}
else {
WAVEHDR *last = (WAVEHDR*)(A->in_headp);
while (last->lpNext != NULL) {
last = last->lpNext;
}
last->lpNext = p; /* append to last one */
}
LeaveCriticalSection (&(A->in_cs));
}
}
/*
* Called when output system is done with a block and it
* is again available for us to fill.
*/
static void CALLBACK out_callback (HWAVEOUT handle, UINT msg, DWORD instance, DWORD param1, DWORD param2)
{
if (msg == WOM_DONE) {
WAVEHDR *p = (WAVEHDR*)param1;
p->dwBufferLength = 0;
p->dwUser = DWU_DONE;
}
}
/*------------------------------------------------------------------
*
* Name: audio_get
*
* Purpose: Get one byte from the audio device.
*
*
* Inputs: a - Audio soundcard number.
*
* Returns: 0 - 255 for a valid sample.
* -1 for any type of error.
*
* Description: The caller must deal with the details of mono/stereo
* and number of bytes per sample.
*
* This will wait if no data is currently available.
*
*----------------------------------------------------------------*/
// Use hot attribute for all functions called for every audio sample.
__attribute__((hot))
int audio_get (int a)
{
struct adev_s *A;
WAVEHDR *p;
int n;
int sample;
A = &(adev[a]);
switch (A->g_audio_in_type) {
/*
* Soundcard.
*/
case AUDIO_IN_TYPE_SOUNDCARD:
while (1) {
/*
* Wait if nothing available.
* Could use an event to wake up but this is adequate.
*/
int timeout = 25;
while (A->in_headp == NULL) {
//SLEEP_MS (ONE_BUF_TIME / 5);
SLEEP_MS (ONE_BUF_TIME);
timeout--;
if (timeout <= 0) {
text_color_set(DW_COLOR_ERROR);
// TODO1.2: Need more details. Can we keep going?
dw_printf ("Timeout waiting for input from audio device %d.\n", a);
audio_stats (a,
save_audio_config_p->adev[a].num_channels,
0,
save_audio_config_p->statistics_interval);
return (-1);
}
}
p = (WAVEHDR*)(A->in_headp); /* no need to be volatile at this point */
if (p->dwUser == -1) {
waveInUnprepareHeader(A->audio_in_handle, p, sizeof(WAVEHDR));
p->dwUser = 0; /* Index for next byte. */
audio_stats (a,
save_audio_config_p->adev[a].num_channels,
p->dwBytesRecorded / (save_audio_config_p->adev[a].num_channels * save_audio_config_p->adev[a].bits_per_sample / 8),
save_audio_config_p->statistics_interval);
}
if (p->dwUser < p->dwBytesRecorded) {
n = ((unsigned char*)(p->lpData))[p->dwUser++];
#if DEBUGx
text_color_set(DW_COLOR_DEBUG);
dw_printf ("audio_get(): returns %d\n", n);
#endif
return (n);
}
/*
* Buffer is all used up. Give it back to sound input system.
*/
EnterCriticalSection (&(A->in_cs));
A->in_headp = p->lpNext;
LeaveCriticalSection (&(A->in_cs));
p->dwFlags = 0;
waveInPrepareHeader(A->audio_in_handle, p, sizeof(WAVEHDR));
waveInAddBuffer(A->audio_in_handle, p, sizeof(WAVEHDR));
}
break;
/*
* UDP.
*/
case AUDIO_IN_TYPE_SDR_UDP:
while (A->stream_next >= A->stream_len) {
int res;
assert (A->udp_sock > 0);
res = recv (A->udp_sock, A->stream_data, SDR_UDP_BUF_MAXLEN, 0);
if (res <= 0) {
text_color_set(DW_COLOR_ERROR);
dw_printf ("Can't read from udp socket, errno %d", WSAGetLastError());
A->stream_len = 0;
A->stream_next = 0;
audio_stats (a,
save_audio_config_p->adev[a].num_channels,
0,
save_audio_config_p->statistics_interval);
return (-1);
}
audio_stats (a,
save_audio_config_p->adev[a].num_channels,
res / (save_audio_config_p->adev[a].num_channels * save_audio_config_p->adev[a].bits_per_sample / 8),
save_audio_config_p->statistics_interval);
A->stream_len = res;
A->stream_next = 0;
}
sample = A->stream_data[A->stream_next] & 0xff;
A->stream_next++;
return (sample);
break;
/*
* stdin.
*/
case AUDIO_IN_TYPE_STDIN:
while (A->stream_next >= A->stream_len) {
int res;
res = read(STDIN_FILENO, A->stream_data, 1024);
if (res <= 0) {
text_color_set(DW_COLOR_INFO);
dw_printf ("\nEnd of file on stdin. Exiting.\n");
exit (0);
}
audio_stats (a,
save_audio_config_p->adev[a].num_channels,
res / (save_audio_config_p->adev[a].num_channels * save_audio_config_p->adev[a].bits_per_sample / 8),
save_audio_config_p->statistics_interval);
A->stream_len = res;
A->stream_next = 0;
}
return (A->stream_data[A->stream_next++] & 0xff);
break;
}
return (-1);
} /* end audio_get */
/*------------------------------------------------------------------
*
* Name: audio_put
*
* Purpose: Send one byte to the audio device.
*
* Inputs: a - Index for audio device.
*
* c - One byte in range of 0 - 255.
*
*
* Global In: out_current - index of output buffer currenly being filled.
*
* Returns: Normally non-negative.
* -1 for any type of error.
*
* Description: The caller must deal with the details of mono/stereo
* and number of bytes per sample.
*
* See Also: audio_flush
* audio_wait
*
*----------------------------------------------------------------*/
int audio_put (int a, int c)
{
WAVEHDR *p;
struct adev_s *A;
A = &(adev[a]);
/*
* Wait if no buffers are available.
* Don't use p yet because compiler might might consider dwFlags a loop invariant.
*/
int timeout = 10;
while ( A->out_wavehdr[A->out_current].dwUser == DWU_PLAYING) {
SLEEP_MS (ONE_BUF_TIME);
timeout--;
if (timeout <= 0) {
text_color_set(DW_COLOR_ERROR);
dw_printf ("Audio output failure waiting for buffer.\n");
ptt_term ();
return (-1);
}
}
p = (LPWAVEHDR)(&(A->out_wavehdr[A->out_current]));
if (p->dwUser == DWU_DONE) {
waveOutUnprepareHeader (A->audio_out_handle, p, sizeof(WAVEHDR));
p->dwBufferLength = 0;
p->dwUser = DWU_FILLING;
}
/* Should never be full at this point. */
assert (p->dwBufferLength >= 0);
assert (p->dwBufferLength < A->outbuf_size);
p->lpData[p->dwBufferLength++] = c;
if (p->dwBufferLength == A->outbuf_size) {
return (audio_flush(a));
}
return (0);
} /* end audio_put */
/*------------------------------------------------------------------
*
* Name: audio_flush
*
* Purpose: Send current buffer to the audio output system.
*
* Inputs: a - Index for audio device.
*
* Returns: Normally non-negative.
* -1 for any type of error.
*
* See Also: audio_flush
* audio_wait
*
*----------------------------------------------------------------*/
int audio_flush (int a)
{
WAVEHDR *p;
MMRESULT e;
struct adev_s *A;
A = &(adev[a]);
p = (LPWAVEHDR)(&(A->out_wavehdr[A->out_current]));
if (p->dwUser == DWU_FILLING && p->dwBufferLength > 0) {
p->dwUser = DWU_PLAYING;
waveOutPrepareHeader(A->audio_out_handle, p, sizeof(WAVEHDR));
e = waveOutWrite(A->audio_out_handle, p, sizeof(WAVEHDR));
if (e != MMSYSERR_NOERROR) {
text_color_set (DW_COLOR_ERROR);
dw_printf ("audio out write error %d\n", e);
/* I don't expect this to ever happen but if it */
/* does, make the buffer available for filling. */
p->dwUser = DWU_DONE;
return (-1);
}
A->out_current = (A->out_current + 1) % NUM_OUT_BUF;
}
return (0);
} /* end audio_flush */
/*------------------------------------------------------------------
*
* Name: audio_wait
*
* Purpose: Finish up audio output before turning PTT off.
*
* Inputs: a - Index for audio device (not channel!)
*
* Returns: None.
*
* Description: Flush out any partially filled audio output buffer.
* Wait until all the queued up audio out has been played.
* Take any other necessary actions to stop audio output.
*
* In an ideal world:
*
* We would like to ask the hardware when all the queued
* up sound has actually come out the speaker.
*
* In reality:
*
* This has been found to be less than reliable in practice.
*
* Caller does the following:
*
* (1) Make note of when PTT is turned on.
* (2) Calculate how long it will take to transmit the
* frame including TXDELAY, frame (including
* "flags", data, FCS and bit stuffing), and TXTAIL.
* (3) Call this function, which might or might not wait long enough.
* (4) Add (1) and (2) resulting in when PTT should be turned off.
* (5) Take difference between current time and desired PPT off time
* and wait for additoinal time if required.
*
*----------------------------------------------------------------*/
void audio_wait (int a)
{
audio_flush (a);
} /* end audio_wait */
/*------------------------------------------------------------------
*
* Name: audio_close
*
*
* Purpose: Close all of the audio devices.
*
* Returns: Normally non-negative.
* -1 for any type of error.
*
*
*----------------------------------------------------------------*/
int audio_close (void)
{
int err = 0;
int n;
int a;
for (a=0; aadev[a].defined) {
struct adev_s *A = &(adev[a]);
assert (A->audio_in_handle != 0);
assert (A->audio_out_handle != 0);
audio_wait (a);
/* Shutdown audio input. */
waveInReset(A->audio_in_handle);
waveInStop(A->audio_in_handle);
waveInClose(A->audio_in_handle);
A->audio_in_handle = 0;
for (n = 0; n < NUM_IN_BUF; n++) {
waveInUnprepareHeader (A->audio_in_handle, (LPWAVEHDR)(&(A->in_wavehdr[n])), sizeof(WAVEHDR));
A->in_wavehdr[n].dwFlags = 0;
free (A->in_wavehdr[n].lpData);
A->in_wavehdr[n].lpData = NULL;
}
DeleteCriticalSection (&(A->in_cs));
/* Make sure all output buffers have been played then free them. */
for (n = 0; n < NUM_OUT_BUF; n++) {
if (A->out_wavehdr[n].dwUser == DWU_PLAYING) {
int timeout = 2 * NUM_OUT_BUF;
while (A->out_wavehdr[n].dwUser == DWU_PLAYING) {
SLEEP_MS (ONE_BUF_TIME);
timeout--;
if (timeout <= 0) {
text_color_set(DW_COLOR_ERROR);
dw_printf ("Audio output failure on close.\n");
}
}
waveOutUnprepareHeader (A->audio_out_handle, (LPWAVEHDR)(&(A->out_wavehdr[n])), sizeof(WAVEHDR));
A->out_wavehdr[n].dwUser = DWU_DONE;
}
free (A->out_wavehdr[n].lpData);
A->out_wavehdr[n].lpData = NULL;
}
waveOutClose (A->audio_out_handle);
A->audio_out_handle = 0;
} /* if device configured */
} /* for each device. */
/* Not right. always returns 0 but at this point, doesn't matter. */
return (err);
} /* end audio_close */
/* end audio_win.c */