ioq3/code/sdl/sdl_snd.c
Zack Middleton eacb83a244 Allow using pulseaudio for SDL audio capture
Pulseaudio audio capture didn't stop when paused on Debian 8 but works
on Debian 9 when using the same manual SDL build. So it seems to have
been an issue in pulseaudio, not SDL.
2021-02-26 23:55:12 -05:00

440 lines
10 KiB
C

/*
===========================================================================
Copyright (C) 1999-2005 Id Software, Inc.
This file is part of Quake III Arena source code.
Quake III Arena source code 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.
Quake III Arena source code 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 Quake III Arena source code; if not, write to the Free Software
Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA
===========================================================================
*/
#include <stdlib.h>
#include <stdio.h>
#ifdef USE_LOCAL_HEADERS
# include "SDL.h"
#else
# include <SDL.h>
#endif
#include "../qcommon/q_shared.h"
#include "../client/snd_local.h"
#include "../client/client.h"
qboolean snd_inited = qfalse;
cvar_t *s_sdlBits;
cvar_t *s_sdlSpeed;
cvar_t *s_sdlChannels;
cvar_t *s_sdlDevSamps;
cvar_t *s_sdlMixSamps;
/* The audio callback. All the magic happens here. */
static int dmapos = 0;
static int dmasize = 0;
static SDL_AudioDeviceID sdlPlaybackDevice;
#if defined USE_VOIP && SDL_VERSION_ATLEAST( 2, 0, 5 )
#define USE_SDL_AUDIO_CAPTURE
static SDL_AudioDeviceID sdlCaptureDevice;
static cvar_t *s_sdlCapture;
static float sdlMasterGain = 1.0f;
#endif
/*
===============
SNDDMA_AudioCallback
===============
*/
static void SNDDMA_AudioCallback(void *userdata, Uint8 *stream, int len)
{
int pos = (dmapos * (dma.samplebits/8));
if (pos >= dmasize)
dmapos = pos = 0;
if (!snd_inited) /* shouldn't happen, but just in case... */
{
memset(stream, '\0', len);
return;
}
else
{
int tobufend = dmasize - pos; /* bytes to buffer's end. */
int len1 = len;
int len2 = 0;
if (len1 > tobufend)
{
len1 = tobufend;
len2 = len - len1;
}
memcpy(stream, dma.buffer + pos, len1);
if (len2 <= 0)
dmapos += (len1 / (dma.samplebits/8));
else /* wraparound? */
{
memcpy(stream+len1, dma.buffer, len2);
dmapos = (len2 / (dma.samplebits/8));
}
}
if (dmapos >= dmasize)
dmapos = 0;
#ifdef USE_SDL_AUDIO_CAPTURE
if (sdlMasterGain != 1.0f)
{
int i;
if (dma.isfloat && (dma.samplebits == 32))
{
float *ptr = (float *) stream;
len /= sizeof (*ptr);
for (i = 0; i < len; i++, ptr++)
{
*ptr *= sdlMasterGain;
}
}
else if (dma.samplebits == 16)
{
Sint16 *ptr = (Sint16 *) stream;
len /= sizeof (*ptr);
for (i = 0; i < len; i++, ptr++)
{
*ptr = (Sint16) (((float) *ptr) * sdlMasterGain);
}
}
else if (dma.samplebits == 8)
{
Uint8 *ptr = (Uint8 *) stream;
len /= sizeof (*ptr);
for (i = 0; i < len; i++, ptr++)
{
*ptr = (Uint8) (((float) *ptr) * sdlMasterGain);
}
}
}
#endif
}
static struct
{
Uint16 enumFormat;
char *stringFormat;
} formatToStringTable[ ] =
{
{ AUDIO_U8, "AUDIO_U8" },
{ AUDIO_S8, "AUDIO_S8" },
{ AUDIO_U16LSB, "AUDIO_U16LSB" },
{ AUDIO_S16LSB, "AUDIO_S16LSB" },
{ AUDIO_U16MSB, "AUDIO_U16MSB" },
{ AUDIO_S16MSB, "AUDIO_S16MSB" },
{ AUDIO_F32LSB, "AUDIO_F32LSB" },
{ AUDIO_F32MSB, "AUDIO_F32MSB" }
};
static int formatToStringTableSize = ARRAY_LEN( formatToStringTable );
/*
===============
SNDDMA_PrintAudiospec
===============
*/
static void SNDDMA_PrintAudiospec(const char *str, const SDL_AudioSpec *spec)
{
int i;
char *fmt = NULL;
Com_Printf("%s:\n", str);
for( i = 0; i < formatToStringTableSize; i++ ) {
if( spec->format == formatToStringTable[ i ].enumFormat ) {
fmt = formatToStringTable[ i ].stringFormat;
}
}
if( fmt ) {
Com_Printf( " Format: %s\n", fmt );
} else {
Com_Printf( " Format: " S_COLOR_RED "UNKNOWN\n");
}
Com_Printf( " Freq: %d\n", (int) spec->freq );
Com_Printf( " Samples: %d\n", (int) spec->samples );
Com_Printf( " Channels: %d\n", (int) spec->channels );
}
/*
===============
SNDDMA_Init
===============
*/
qboolean SNDDMA_Init(void)
{
SDL_AudioSpec desired;
SDL_AudioSpec obtained;
int tmp;
if (snd_inited)
return qtrue;
if (!s_sdlBits) {
s_sdlBits = Cvar_Get("s_sdlBits", "16", CVAR_ARCHIVE);
s_sdlSpeed = Cvar_Get("s_sdlSpeed", "0", CVAR_ARCHIVE);
s_sdlChannels = Cvar_Get("s_sdlChannels", "2", CVAR_ARCHIVE);
s_sdlDevSamps = Cvar_Get("s_sdlDevSamps", "0", CVAR_ARCHIVE);
s_sdlMixSamps = Cvar_Get("s_sdlMixSamps", "0", CVAR_ARCHIVE);
}
Com_Printf( "SDL_Init( SDL_INIT_AUDIO )... " );
if (SDL_Init(SDL_INIT_AUDIO) != 0)
{
Com_Printf( "FAILED (%s)\n", SDL_GetError( ) );
return qfalse;
}
Com_Printf( "OK\n" );
Com_Printf( "SDL audio driver is \"%s\".\n", SDL_GetCurrentAudioDriver( ) );
memset(&desired, '\0', sizeof (desired));
memset(&obtained, '\0', sizeof (obtained));
tmp = ((int) s_sdlBits->value);
if ((tmp != 16) && (tmp != 8))
tmp = 16;
desired.freq = (int) s_sdlSpeed->value;
if(!desired.freq) desired.freq = 22050;
desired.format = ((tmp == 16) ? AUDIO_S16SYS : AUDIO_U8);
// I dunno if this is the best idea, but I'll give it a try...
// should probably check a cvar for this...
if (s_sdlDevSamps->value)
desired.samples = s_sdlDevSamps->value;
else
{
// just pick a sane default.
if (desired.freq <= 11025)
desired.samples = 256;
else if (desired.freq <= 22050)
desired.samples = 512;
else if (desired.freq <= 44100)
desired.samples = 1024;
else
desired.samples = 2048; // (*shrug*)
}
desired.channels = (int) s_sdlChannels->value;
desired.callback = SNDDMA_AudioCallback;
sdlPlaybackDevice = SDL_OpenAudioDevice(NULL, SDL_FALSE, &desired, &obtained, SDL_AUDIO_ALLOW_ANY_CHANGE);
if (sdlPlaybackDevice == 0)
{
Com_Printf("SDL_OpenAudioDevice() failed: %s\n", SDL_GetError());
SDL_QuitSubSystem(SDL_INIT_AUDIO);
return qfalse;
}
SNDDMA_PrintAudiospec("SDL_AudioSpec", &obtained);
// dma.samples needs to be big, or id's mixer will just refuse to
// work at all; we need to keep it significantly bigger than the
// amount of SDL callback samples, and just copy a little each time
// the callback runs.
// 32768 is what the OSS driver filled in here on my system. I don't
// know if it's a good value overall, but at least we know it's
// reasonable...this is why I let the user override.
tmp = s_sdlMixSamps->value;
if (!tmp)
tmp = (obtained.samples * obtained.channels) * 10;
// samples must be divisible by number of channels
tmp -= tmp % obtained.channels;
dmapos = 0;
dma.samplebits = SDL_AUDIO_BITSIZE(obtained.format);
dma.isfloat = SDL_AUDIO_ISFLOAT(obtained.format);
dma.channels = obtained.channels;
dma.samples = tmp;
dma.fullsamples = dma.samples / dma.channels;
dma.submission_chunk = 1;
dma.speed = obtained.freq;
dmasize = (dma.samples * (dma.samplebits/8));
dma.buffer = calloc(1, dmasize);
#ifdef USE_SDL_AUDIO_CAPTURE
// !!! FIXME: some of these SDL_OpenAudioDevice() values should be cvars.
s_sdlCapture = Cvar_Get( "s_sdlCapture", "1", CVAR_ARCHIVE | CVAR_LATCH );
if (!s_sdlCapture->integer)
{
Com_Printf("SDL audio capture support disabled by user ('+set s_sdlCapture 1' to enable)\n");
}
#if USE_MUMBLE
else if (cl_useMumble->integer)
{
Com_Printf("SDL audio capture support disabled for Mumble support\n");
}
#endif
else
{
/* !!! FIXME: list available devices and let cvar specify one, like OpenAL does */
SDL_AudioSpec spec;
SDL_zero(spec);
spec.freq = 48000;
spec.format = AUDIO_S16SYS;
spec.channels = 1;
spec.samples = VOIP_MAX_PACKET_SAMPLES * 4;
sdlCaptureDevice = SDL_OpenAudioDevice(NULL, SDL_TRUE, &spec, NULL, 0);
Com_Printf( "SDL capture device %s.\n",
(sdlCaptureDevice == 0) ? "failed to open" : "opened");
}
sdlMasterGain = 1.0f;
#endif
Com_Printf("Starting SDL audio callback...\n");
SDL_PauseAudioDevice(sdlPlaybackDevice, 0); // start callback.
// don't unpause the capture device; we'll do that in StartCapture.
Com_Printf("SDL audio initialized.\n");
snd_inited = qtrue;
return qtrue;
}
/*
===============
SNDDMA_GetDMAPos
===============
*/
int SNDDMA_GetDMAPos(void)
{
return dmapos;
}
/*
===============
SNDDMA_Shutdown
===============
*/
void SNDDMA_Shutdown(void)
{
if (sdlPlaybackDevice != 0)
{
Com_Printf("Closing SDL audio playback device...\n");
SDL_CloseAudioDevice(sdlPlaybackDevice);
Com_Printf("SDL audio playback device closed.\n");
sdlPlaybackDevice = 0;
}
#ifdef USE_SDL_AUDIO_CAPTURE
if (sdlCaptureDevice)
{
Com_Printf("Closing SDL audio capture device...\n");
SDL_CloseAudioDevice(sdlCaptureDevice);
Com_Printf("SDL audio capture device closed.\n");
sdlCaptureDevice = 0;
}
#endif
SDL_QuitSubSystem(SDL_INIT_AUDIO);
free(dma.buffer);
dma.buffer = NULL;
dmapos = dmasize = 0;
snd_inited = qfalse;
Com_Printf("SDL audio shut down.\n");
}
/*
===============
SNDDMA_Submit
Send sound to device if buffer isn't really the dma buffer
===============
*/
void SNDDMA_Submit(void)
{
SDL_UnlockAudioDevice(sdlPlaybackDevice);
}
/*
===============
SNDDMA_BeginPainting
===============
*/
void SNDDMA_BeginPainting (void)
{
SDL_LockAudioDevice(sdlPlaybackDevice);
}
#ifdef USE_VOIP
void SNDDMA_StartCapture(void)
{
#ifdef USE_SDL_AUDIO_CAPTURE
if (sdlCaptureDevice)
{
SDL_ClearQueuedAudio(sdlCaptureDevice);
SDL_PauseAudioDevice(sdlCaptureDevice, 0);
}
#endif
}
int SNDDMA_AvailableCaptureSamples(void)
{
#ifdef USE_SDL_AUDIO_CAPTURE
// divided by 2 to convert from bytes to (mono16) samples.
return sdlCaptureDevice ? (SDL_GetQueuedAudioSize(sdlCaptureDevice) / 2) : 0;
#else
return 0;
#endif
}
void SNDDMA_Capture(int samples, byte *data)
{
#ifdef USE_SDL_AUDIO_CAPTURE
// multiplied by 2 to convert from (mono16) samples to bytes.
if (sdlCaptureDevice)
{
SDL_DequeueAudio(sdlCaptureDevice, data, samples * 2);
}
else
#endif
{
SDL_memset(data, '\0', samples * 2);
}
}
void SNDDMA_StopCapture(void)
{
#ifdef USE_SDL_AUDIO_CAPTURE
if (sdlCaptureDevice)
{
SDL_PauseAudioDevice(sdlCaptureDevice, 1);
}
#endif
}
void SNDDMA_MasterGain( float val )
{
#ifdef USE_SDL_AUDIO_CAPTURE
sdlMasterGain = val;
#endif
}
#endif