Bochs/bochs/iodev/sound/soundsdl.cc
Volker Ruppert f844e846b7 Added PCM output data resampling in a separate thread. The resampler requires
either libsamplerate or the SoX resampler library installed. If not installed,
the data is only copied to the output buffer and sample rate of the output
driver is changed similar to legacy code. Related changes:
- Added check for libsamplerate or SoX resampler header files in configure
  script.
- Added functions for converting source format to float (requied by resampler)
  and float to output format.
- Added support for float type data in the audio buffer code. Buffer chain #0
  receives float data from sendwavepacket() and buffer #1 receives data for
  mixing and output in format "16-bit signed little endian stereo".
- ALSA: Disable builtin resampling feature if resampler is present to avoid
  doing it twice.
2017-02-05 08:33:03 +00:00

182 lines
4.7 KiB
C++

/////////////////////////////////////////////////////////////////////////
// $Id$
/////////////////////////////////////////////////////////////////////////
//
// Copyright (C) 2012-2017 The Bochs Project
//
// This library is free software; you can redistribute it and/or
// modify it under the terms of the GNU Lesser General Public
// License as published by the Free Software Foundation; either
// version 2 of the License, or (at your option) any later version.
//
// This library 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
// Lesser General Public License for more details.
//
// You should have received a copy of the GNU Lesser General Public
// License along with this library; if not, write to the Free Software
// Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA
// Lowlevel sound output support for SDL written by Volker Ruppert
#include "iodev.h"
#include "soundlow.h"
#include "soundsdl.h"
#if BX_HAVE_SOUND_SDL && BX_SUPPORT_SOUNDLOW
#define LOG_THIS
#include <SDL.h>
// SDL audio callback
void sdl_callback(void *thisptr, Bit8u *stream, int len)
{
memset(stream, 0, len);
((bx_soundlow_waveout_sdl_c*)thisptr)->mixer_common(stream, len);
}
// bx_soundlow_waveout_sdl_c class implemenzation
bx_soundlow_waveout_sdl_c::bx_soundlow_waveout_sdl_c()
:bx_soundlow_waveout_c()
{
WaveOpen = 0;
if (SDL_InitSubSystem(SDL_INIT_AUDIO)) {
BX_PANIC(("Initialization of sound lowlevel module 'sdl' failed"));
} else {
BX_INFO(("Sound lowlevel module 'sdl' initialized"));
}
}
bx_soundlow_waveout_sdl_c::~bx_soundlow_waveout_sdl_c()
{
if (pcm_callback_id >= 0) {
unregister_wave_callback(pcm_callback_id);
pcm_callback_id = -1;
}
WaveOpen = 0;
mixer_control = 0;
SDL_CloseAudio();
SDL_QuitSubSystem(SDL_INIT_AUDIO);
}
int bx_soundlow_waveout_sdl_c::openwaveoutput(const char *wavedev)
{
set_pcm_params(&real_pcm_param);
start_resampler_thread();
return BX_SOUNDLOW_OK;
}
int bx_soundlow_waveout_sdl_c::set_pcm_params(bx_pcm_param_t *param)
{
int signeddata = param->format & 1;
BX_DEBUG(("set_pcm_params(): %u, %u, %u, %02x", param->samplerate, param->bits,
param->channels, param->format));
fmt.freq = param->samplerate;
if (param->bits == 16) {
if (signeddata == 1)
fmt.format = AUDIO_S16;
else
fmt.format = AUDIO_U16;
} else if (param->bits == 8) {
if (signeddata == 1)
fmt.format = AUDIO_S8;
else
fmt.format = AUDIO_U8;
} else
return BX_SOUNDLOW_ERR;
fmt.channels = param->channels;
fmt.samples = fmt.freq / 10;
fmt.callback = sdl_callback;
fmt.userdata = this;
if (WaveOpen) {
SDL_CloseAudio();
} else {
pcm_callback_id = register_wave_callback(this, pcm_callback);
}
if (SDL_OpenAudio(&fmt, NULL) < 0) {
BX_PANIC(("SDL_OpenAudio() failed"));
WaveOpen = 0;
return BX_SOUNDLOW_ERR;
} else {
if (fmt.freq != param->samplerate) {
param->samplerate = fmt.freq;
BX_INFO(("changed sample rate to %d", fmt.freq));
}
WaveOpen = 1;
mixer_control = 1;
}
SDL_PauseAudio(0);
return BX_SOUNDLOW_OK;
}
void bx_soundlow_waveout_sdl_c::resampler(audio_buffer_t *inbuffer, audio_buffer_t *outbuffer)
{
Bit32u fcount;
float *fbuffer = NULL;
UNUSED(outbuffer);
fcount = resampler_common(inbuffer, &fbuffer);
SDL_LockAudio();
if (WaveOpen) {
audio_buffer_t *newbuffer = audio_buffers[1]->new_buffer(fcount << 1);
convert_float_to_s16le(fbuffer, fcount, newbuffer->data);
}
SDL_UnlockAudio();
if (fbuffer != NULL) {
delete [] fbuffer;
}
}
bx_bool bx_soundlow_waveout_sdl_c::mixer_common(Bit8u *buffer, int len)
{
Bit32u len2 = 0;
Bit8u *tmpbuffer = (Bit8u*)malloc(len);
for (int i = 0; i < cb_count; i++) {
if (get_wave[i].cb != NULL) {
memset(tmpbuffer, 0, len);
len2 = get_wave[i].cb(get_wave[i].device, fmt.freq, tmpbuffer, len);
if (len2 > 0) {
SDL_MixAudio(buffer, tmpbuffer, len2, SDL_MIX_MAXVOLUME);
}
}
}
free(tmpbuffer);
return 1;
}
void bx_soundlow_waveout_sdl_c::unregister_wave_callback(int callback_id)
{
SDL_LockAudio();
if ((callback_id >= 0) && (callback_id < BX_MAX_WAVE_CALLBACKS)) {
get_wave[callback_id].device = NULL;
get_wave[callback_id].cb = NULL;
}
SDL_UnlockAudio();
}
// bx_sound_sdl_c class implemenzation
bx_sound_sdl_c::bx_sound_sdl_c()
:bx_sound_lowlevel_c()
{
// nothing here yet
}
bx_soundlow_waveout_c* bx_sound_sdl_c::get_waveout()
{
if (waveout == NULL) {
waveout = new bx_soundlow_waveout_sdl_c();
}
return waveout;
}
#endif // BX_HAVE_SOUND_SDL