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mpv/audio/out/ao_sdl.c
wm4 d908fbd584 audio/out/pull, ao_sdl: implement new underrun reporting
See previous commits. ao_sdl is worthless, but it might be a good test
for pull-based AOs.

This stops using the old underrun reporting if the new one is enabled.
Also, since the AO's behavior can in theory not be according to
expectations, this needs to be enabled for every single pull AO
separately.

For some reason, in certain cases I get multiple underrun warnings while
cache-pausing is active. It fills the cache, restarts the AO,
immediately underruns again, and then fills the cache again. I'm not
sure why this happens; maybe ao_sdl tries to catch up when it shouldn't.
Who knows.
2019-10-11 20:02:23 +02:00

219 lines
5.5 KiB
C

/*
* audio output driver for SDL 1.2+
* Copyright (C) 2012 Rudolf Polzer <divVerent@xonotic.org>
*
* This file is part of mpv.
*
* mpv 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 2.1 of the License, or (at your option) any later version.
*
* mpv 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 mpv. If not, see <http://www.gnu.org/licenses/>.
*/
#include "config.h"
#include "audio/format.h"
#include "mpv_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 const int fmtmap[][2] = {
{AF_FORMAT_U8, AUDIO_U8},
{AF_FORMAT_S16, AUDIO_S16SYS},
#ifdef AUDIO_S32SYS
{AF_FORMAT_S32, AUDIO_S32SYS},
#endif
#ifdef AUDIO_F32SYS
{AF_FORMAT_FLOAT, AUDIO_F32SYS},
#endif
{0}
};
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 = {0};
desired.format = AUDIO_S16SYS;
for (int n = 0; fmtmap[n][0]; n++) {
if (ao->format == fmtmap[n][0]) {
desired.format = fmtmap[n][1];
break;
}
}
desired.freq = ao->samplerate;
desired.channels = ao->channels.num;
if (priv->buflen) {
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);
SDL_AudioSpec 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;
ao->format = 0;
for (int n = 0; fmtmap[n][0]; n++) {
if (obtained.format == fmtmap[n][1]) {
ao->format = fmtmap[n][0];
break;
}
}
if (!ao->format) {
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,
.reports_underruns = true,
.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}
},
.options_prefix = "sdl",
};