mirror of https://github.com/mpv-player/mpv
374 lines
10 KiB
C
374 lines
10 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 "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 <libavutil/fifo.h>
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#include <libavutil/common.h>
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#include <SDL.h>
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// hack because SDL can't be asked about the current delay
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#define ESTIMATE_DELAY
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struct priv
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{
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AVFifoBuffer *buffer;
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SDL_mutex *buffer_mutex;
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SDL_cond *underrun_cond;
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bool unpause;
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bool paused;
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#ifdef ESTIMATE_DELAY
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int64_t callback_time0;
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int64_t callback_time1;
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#endif
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float buflen;
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float bufcnt;
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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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struct priv *priv = ao->priv;
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SDL_LockMutex(priv->buffer_mutex);
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#ifdef ESTIMATE_DELAY
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priv->callback_time1 = priv->callback_time0;
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priv->callback_time0 = mp_time_us();
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#endif
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while (len > 0 && !priv->paused) {
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int got = av_fifo_size(priv->buffer);
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if (got > len)
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got = len;
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if (got > 0) {
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av_fifo_generic_read(priv->buffer, stream, got, NULL);
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len -= got;
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stream += got;
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}
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if (len > 0)
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SDL_CondWait(priv->underrun_cond, priv->buffer_mutex);
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}
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SDL_UnlockMutex(priv->buffer_mutex);
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}
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static void uninit(struct ao *ao, bool cut_audio)
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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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// abort the callback
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priv->paused = 1;
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if (SDL_WasInit(SDL_INIT_AUDIO)) {
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if (priv->buffer_mutex)
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SDL_LockMutex(priv->buffer_mutex);
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if (priv->underrun_cond)
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SDL_CondSignal(priv->underrun_cond);
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if (priv->buffer_mutex)
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SDL_UnlockMutex(priv->buffer_mutex);
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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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// get rid of the mutex
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if (priv->underrun_cond)
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SDL_DestroyCond(priv->underrun_cond);
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if (priv->buffer_mutex)
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SDL_DestroyMutex(priv->buffer_mutex);
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if (priv->buffer)
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av_fifo_free(priv->buffer);
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talloc_free(ao->priv);
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ao->priv = NULL;
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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, true);
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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, true);
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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 = FFMIN(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, true);
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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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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, true);
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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, true);
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return -1;
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}
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ao->samplerate = obtained.freq;
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priv->buffer = av_fifo_alloc(obtained.size * priv->bufcnt);
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priv->buffer_mutex = SDL_CreateMutex();
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if (!priv->buffer_mutex) {
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MP_ERR(ao, "SDL_CreateMutex failed\n");
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uninit(ao, true);
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return -1;
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}
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priv->underrun_cond = SDL_CreateCond();
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if (!priv->underrun_cond) {
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MP_ERR(ao, "SDL_CreateCond failed\n");
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uninit(ao, true);
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return -1;
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}
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priv->unpause = 1;
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priv->paused = 1;
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priv->callback_time0 = priv->callback_time1 = mp_time_us();
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return 1;
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}
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static void reset(struct ao *ao)
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{
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struct priv *priv = ao->priv;
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SDL_LockMutex(priv->buffer_mutex);
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av_fifo_reset(priv->buffer);
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SDL_UnlockMutex(priv->buffer_mutex);
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}
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static int get_space(struct ao *ao)
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{
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struct priv *priv = ao->priv;
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SDL_LockMutex(priv->buffer_mutex);
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int space = av_fifo_space(priv->buffer);
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SDL_UnlockMutex(priv->buffer_mutex);
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return space / ao->sstride;
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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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SDL_PauseAudio(SDL_TRUE);
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priv->unpause = 0;
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priv->paused = 1;
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SDL_CondSignal(priv->underrun_cond);
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}
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static void do_resume(struct ao *ao)
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{
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struct priv *priv = ao->priv;
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priv->paused = 0;
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SDL_PauseAudio(SDL_FALSE);
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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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SDL_LockMutex(priv->buffer_mutex);
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int free = av_fifo_space(priv->buffer);
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SDL_UnlockMutex(priv->buffer_mutex);
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if (free)
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priv->unpause = 1;
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else
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do_resume(ao);
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}
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static int play(struct ao *ao, void **data, int samples, int flags)
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{
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struct priv *priv = ao->priv;
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int len = samples * ao->sstride;
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SDL_LockMutex(priv->buffer_mutex);
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int free = av_fifo_space(priv->buffer);
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if (len > free) len = free;
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av_fifo_generic_write(priv->buffer, data[0], len, NULL);
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SDL_CondSignal(priv->underrun_cond);
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SDL_UnlockMutex(priv->buffer_mutex);
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if (priv->unpause) {
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priv->unpause = 0;
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do_resume(ao);
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}
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return len / ao->sstride;
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}
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static float get_delay(struct ao *ao)
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{
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struct priv *priv = ao->priv;
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SDL_LockMutex(priv->buffer_mutex);
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int sz = av_fifo_size(priv->buffer);
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#ifdef ESTIMATE_DELAY
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int64_t callback_time0 = priv->callback_time0;
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int64_t callback_time1 = priv->callback_time1;
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#endif
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SDL_UnlockMutex(priv->buffer_mutex);
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// delay component: our FIFO's length
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float delay = sz / (float) ao->bps;
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#ifdef ESTIMATE_DELAY
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// delay component: outstanding audio living in SDL
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int64_t current_time = mp_time_us();
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// interval between callbacks
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int64_t callback_interval = callback_time0 - callback_time1;
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int64_t elapsed_interval = current_time - callback_time0;
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if (elapsed_interval > callback_interval)
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elapsed_interval = callback_interval;
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// delay subcomponent: remaining audio from the currently played buffer
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int64_t buffer_interval = callback_interval - elapsed_interval;
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// delay subcomponent: remaining audio from the next played buffer, as
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// provided by the callback
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buffer_interval += callback_interval;
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delay += buffer_interval / 1000000.0;
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#endif
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return delay;
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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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.get_space = get_space,
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.play = play,
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.get_delay = get_delay,
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.pause = pause,
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.resume = resume,
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.reset = reset,
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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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.bufcnt = 2,
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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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OPT_FLOAT("bufcnt", bufcnt, 0),
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{0}
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},
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};
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