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/*
* audio output driver for SDL 1.2+
* Copyright (C) 2012 Rudolf Polzer <divVerent@xonotic.org>
*
* This file is part of mpv.
*
* MPlayer 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.
*
* MPlayer 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 MPlayer; if not, write to the Free Software Foundation, Inc.,
* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
*/
#include "config.h"
#include "audio/format.h"
#include "talloc.h"
#include "ao.h"
#include "internal.h"
#include "common/common.h"
#include "common/msg.h"
#include "options/m_option.h"
#include "osdep/timer.h"
#include <SDL.h>
struct priv
{
bool paused;
float buflen;
};
static void audio_callback(void *userdata, Uint8 *stream, int len)
{
struct ao *ao = userdata;
void *data[1] = {stream};
if (len % ao->sstride)
MP_ERR(ao, "SDL audio callback not sample aligned");
// Time this buffer will take, plus assume 1 period (1 callback invocation)
// fixed latency.
double delay = 2 * len / (double)ao->bps;
ao_read_data(ao, data, len / ao->sstride, mp_time_us() + 1000000LL * delay);
}
static void uninit(struct ao *ao)
{
struct priv *priv = ao->priv;
if (!priv)
return;
if (SDL_WasInit(SDL_INIT_AUDIO)) {
// make sure the callback exits
SDL_LockAudio();
// close audio device
SDL_QuitSubSystem(SDL_INIT_AUDIO);
}
}
static unsigned int ceil_power_of_two(unsigned int x)
{
int y = 1;
while (y < x)
y *= 2;
return y;
}
static int init(struct ao *ao)
{
if (SDL_WasInit(SDL_INIT_AUDIO)) {
MP_ERR(ao, "already initialized\n");
return -1;
}
struct priv *priv = ao->priv;
if (SDL_InitSubSystem(SDL_INIT_AUDIO)) {
if (!ao->probing)
MP_ERR(ao, "SDL_Init failed\n");
uninit(ao);
return -1;
}
struct mp_chmap_sel sel = {0};
mp_chmap_sel_add_waveext_def(&sel);
if (!ao_chmap_sel_adjust(ao, &sel, &ao->channels)) {
uninit(ao);
return -1;
}
ao->format = af_fmt_from_planar(ao->format);
SDL_AudioSpec desired, obtained;
switch (ao->format) {
case AF_FORMAT_U8: desired.format = AUDIO_U8; break;
case AF_FORMAT_S8: desired.format = AUDIO_S8; break;
case AF_FORMAT_U16_LE: desired.format = AUDIO_U16LSB; break;
case AF_FORMAT_U16_BE: desired.format = AUDIO_U16MSB; break;
default:
case AF_FORMAT_S16_LE: desired.format = AUDIO_S16LSB; break;
case AF_FORMAT_S16_BE: desired.format = AUDIO_S16MSB; break;
#ifdef AUDIO_S32LSB
case AF_FORMAT_S32_LE: desired.format = AUDIO_S32LSB; break;
#endif
#ifdef AUDIO_S32MSB
case AF_FORMAT_S32_BE: desired.format = AUDIO_S32MSB; break;
#endif
#ifdef AUDIO_F32LSB
case AF_FORMAT_FLOAT_LE: desired.format = AUDIO_F32LSB; break;
#endif
#ifdef AUDIO_F32MSB
case AF_FORMAT_FLOAT_BE: desired.format = AUDIO_F32MSB; break;
#endif
}
desired.freq = ao->samplerate;
desired.channels = ao->channels.num;
desired.samples = MPMIN(32768, ceil_power_of_two(ao->samplerate *
priv->buflen));
desired.callback = audio_callback;
desired.userdata = ao;
MP_VERBOSE(ao, "requested format: %d Hz, %d channels, %x, "
"buffer size: %d samples\n",
(int) desired.freq, (int) desired.channels,
(int) desired.format, (int) desired.samples);
obtained = desired;
if (SDL_OpenAudio(&desired, &obtained)) {
if (!ao->probing)
MP_ERR(ao, "could not open audio: %s\n", SDL_GetError());
uninit(ao);
return -1;
}
MP_VERBOSE(ao, "obtained format: %d Hz, %d channels, %x, "
"buffer size: %d samples\n",
(int) obtained.freq, (int) obtained.channels,
(int) obtained.format, (int) obtained.samples);
// The sample count is usually the number of samples the callback requests,
// which we assume is the period size. Normally, ao.c will allocate a large
// enough buffer. But in case the period size should be pathologically
// large, this will help.
ao->device_buffer = 3 * obtained.samples;
switch (obtained.format) {
case AUDIO_U8: ao->format = AF_FORMAT_U8; break;
case AUDIO_S8: ao->format = AF_FORMAT_S8; break;
case AUDIO_S16LSB: ao->format = AF_FORMAT_S16_LE; break;
case AUDIO_S16MSB: ao->format = AF_FORMAT_S16_BE; break;
case AUDIO_U16LSB: ao->format = AF_FORMAT_U16_LE; break;
case AUDIO_U16MSB: ao->format = AF_FORMAT_U16_BE; break;
#ifdef AUDIO_S32LSB
case AUDIO_S32LSB: ao->format = AF_FORMAT_S32_LE; break;
#endif
#ifdef AUDIO_S32MSB
case AUDIO_S32MSB: ao->format = AF_FORMAT_S32_BE; break;
#endif
#ifdef AUDIO_F32LSB
case AUDIO_F32LSB: ao->format = AF_FORMAT_FLOAT_LE; break;
#endif
#ifdef AUDIO_F32MSB
case AUDIO_F32MSB: ao->format = AF_FORMAT_FLOAT_BE; break;
#endif
default:
if (!ao->probing)
MP_ERR(ao, "could not find matching format\n");
uninit(ao);
return -1;
}
if (!ao_chmap_sel_get_def(ao, &sel, &ao->channels, obtained.channels)) {
uninit(ao);
return -1;
}
ao->samplerate = obtained.freq;
priv->paused = 1;
return 1;
}
static void reset(struct ao *ao)
{
struct priv *priv = ao->priv;
if (!priv->paused)
SDL_PauseAudio(SDL_TRUE);
priv->paused = 1;
}
static void resume(struct ao *ao)
{
struct priv *priv = ao->priv;
if (priv->paused)
SDL_PauseAudio(SDL_FALSE);
priv->paused = 0;
}
#define OPT_BASE_STRUCT struct priv
const struct ao_driver audio_out_sdl = {
.description = "SDL Audio",
.name = "sdl",
.init = init,
.uninit = uninit,
.reset = reset,
.resume = resume,
.priv_size = sizeof(struct priv),
.priv_defaults = &(const struct priv) {
.buflen = 0, // use SDL default
},
.options = (const struct m_option[]) {
OPT_FLOAT("buflen", buflen, 0),
{0}
},
};
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