mirror of
https://github.com/mpv-player/mpv
synced 2024-12-28 01:52:19 +00:00
7221d96ba3
Assume obtained.samples contains the number of samples the SDL audio callback will request at once. Then make sure ao.c will set the buffer size at least to 3 times that value (or more). Might help with bad SDL audio backends like ESD, which supposedly uses a 500ms buffer.
231 lines
6.5 KiB
C
231 lines
6.5 KiB
C
/*
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* audio output driver for SDL 1.2+
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* Copyright (C) 2012 Rudolf Polzer <divVerent@xonotic.org>
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*
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* This file is part of mpv.
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*
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* MPlayer is free software; you can redistribute it and/or modify
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* it under the terms of the GNU General Public License as published by
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* the Free Software Foundation; either version 2 of the License, or
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* (at your option) any later version.
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*
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* MPlayer is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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* GNU General Public License for more details.
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*
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* You should have received a copy of the GNU General Public License along
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* with MPlayer; if not, write to the Free Software Foundation, Inc.,
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* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
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*/
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#include "config.h"
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#include "audio/format.h"
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#include "talloc.h"
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#include "ao.h"
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#include "internal.h"
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#include "common/common.h"
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#include "common/msg.h"
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#include "options/m_option.h"
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#include "osdep/timer.h"
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#include <SDL.h>
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struct priv
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{
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bool paused;
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float buflen;
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};
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static void audio_callback(void *userdata, Uint8 *stream, int len)
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{
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struct ao *ao = userdata;
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void *data[1] = {stream};
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if (len % ao->sstride)
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MP_ERR(ao, "SDL audio callback not sample aligned");
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// Time this buffer will take, plus assume 1 period (1 callback invocation)
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// fixed latency.
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double delay = 2 * len / (double)ao->bps;
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ao_read_data(ao, data, len / ao->sstride, mp_time_us() + 1000000LL * delay);
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}
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static void uninit(struct ao *ao)
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{
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struct priv *priv = ao->priv;
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if (!priv)
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return;
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if (SDL_WasInit(SDL_INIT_AUDIO)) {
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// make sure the callback exits
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SDL_LockAudio();
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// close audio device
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SDL_QuitSubSystem(SDL_INIT_AUDIO);
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}
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}
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static unsigned int ceil_power_of_two(unsigned int x)
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{
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int y = 1;
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while (y < x)
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y *= 2;
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return y;
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}
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static int init(struct ao *ao)
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{
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if (SDL_WasInit(SDL_INIT_AUDIO)) {
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MP_ERR(ao, "already initialized\n");
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return -1;
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}
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struct priv *priv = ao->priv;
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if (SDL_InitSubSystem(SDL_INIT_AUDIO)) {
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if (!ao->probing)
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MP_ERR(ao, "SDL_Init failed\n");
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uninit(ao);
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return -1;
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}
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struct mp_chmap_sel sel = {0};
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mp_chmap_sel_add_waveext_def(&sel);
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if (!ao_chmap_sel_adjust(ao, &sel, &ao->channels)) {
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uninit(ao);
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return -1;
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}
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ao->format = af_fmt_from_planar(ao->format);
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SDL_AudioSpec desired, obtained;
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switch (ao->format) {
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case AF_FORMAT_U8: desired.format = AUDIO_U8; break;
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case AF_FORMAT_S8: desired.format = AUDIO_S8; break;
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case AF_FORMAT_U16_LE: desired.format = AUDIO_U16LSB; break;
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case AF_FORMAT_U16_BE: desired.format = AUDIO_U16MSB; break;
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default:
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case AF_FORMAT_S16_LE: desired.format = AUDIO_S16LSB; break;
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case AF_FORMAT_S16_BE: desired.format = AUDIO_S16MSB; break;
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#ifdef AUDIO_S32LSB
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case AF_FORMAT_S32_LE: desired.format = AUDIO_S32LSB; break;
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#endif
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#ifdef AUDIO_S32MSB
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case AF_FORMAT_S32_BE: desired.format = AUDIO_S32MSB; break;
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#endif
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#ifdef AUDIO_F32LSB
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case AF_FORMAT_FLOAT_LE: desired.format = AUDIO_F32LSB; break;
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#endif
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#ifdef AUDIO_F32MSB
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case AF_FORMAT_FLOAT_BE: desired.format = AUDIO_F32MSB; break;
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#endif
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}
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desired.freq = ao->samplerate;
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desired.channels = ao->channels.num;
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desired.samples = MPMIN(32768, ceil_power_of_two(ao->samplerate *
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priv->buflen));
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desired.callback = audio_callback;
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desired.userdata = ao;
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MP_VERBOSE(ao, "requested format: %d Hz, %d channels, %x, "
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"buffer size: %d samples\n",
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(int) desired.freq, (int) desired.channels,
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(int) desired.format, (int) desired.samples);
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obtained = desired;
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if (SDL_OpenAudio(&desired, &obtained)) {
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if (!ao->probing)
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MP_ERR(ao, "could not open audio: %s\n", SDL_GetError());
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uninit(ao);
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return -1;
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}
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MP_VERBOSE(ao, "obtained format: %d Hz, %d channels, %x, "
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"buffer size: %d samples\n",
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(int) obtained.freq, (int) obtained.channels,
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(int) obtained.format, (int) obtained.samples);
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// The sample count is usually the number of samples the callback requests,
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// which we assume is the period size. Normally, ao.c will allocate a large
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// enough buffer. But in case the period size should be pathologically
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// large, this will help.
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ao->device_buffer = 3 * obtained.samples;
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switch (obtained.format) {
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case AUDIO_U8: ao->format = AF_FORMAT_U8; break;
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case AUDIO_S8: ao->format = AF_FORMAT_S8; break;
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case AUDIO_S16LSB: ao->format = AF_FORMAT_S16_LE; break;
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case AUDIO_S16MSB: ao->format = AF_FORMAT_S16_BE; break;
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case AUDIO_U16LSB: ao->format = AF_FORMAT_U16_LE; break;
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case AUDIO_U16MSB: ao->format = AF_FORMAT_U16_BE; break;
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#ifdef AUDIO_S32LSB
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case AUDIO_S32LSB: ao->format = AF_FORMAT_S32_LE; break;
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#endif
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#ifdef AUDIO_S32MSB
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case AUDIO_S32MSB: ao->format = AF_FORMAT_S32_BE; break;
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#endif
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#ifdef AUDIO_F32LSB
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case AUDIO_F32LSB: ao->format = AF_FORMAT_FLOAT_LE; break;
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#endif
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#ifdef AUDIO_F32MSB
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case AUDIO_F32MSB: ao->format = AF_FORMAT_FLOAT_BE; break;
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#endif
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default:
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if (!ao->probing)
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MP_ERR(ao, "could not find matching format\n");
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uninit(ao);
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return -1;
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}
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if (!ao_chmap_sel_get_def(ao, &sel, &ao->channels, obtained.channels)) {
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uninit(ao);
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return -1;
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}
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ao->samplerate = obtained.freq;
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priv->paused = 1;
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return 1;
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}
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static void pause(struct ao *ao)
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{
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struct priv *priv = ao->priv;
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if (!priv->paused)
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SDL_PauseAudio(SDL_TRUE);
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priv->paused = 1;
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}
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static void resume(struct ao *ao)
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{
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struct priv *priv = ao->priv;
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if (priv->paused)
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SDL_PauseAudio(SDL_FALSE);
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priv->paused = 0;
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}
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#define OPT_BASE_STRUCT struct priv
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const struct ao_driver audio_out_sdl = {
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.description = "SDL Audio",
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.name = "sdl",
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.init = init,
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.uninit = uninit,
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.pause = pause,
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.resume = resume,
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.priv_size = sizeof(struct priv),
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.priv_defaults = &(const struct priv) {
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.buflen = 0, // use SDL default
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},
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.options = (const struct m_option[]) {
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OPT_FLOAT("buflen", buflen, 0),
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{0}
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},
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};
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