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https://github.com/mpv-player/mpv
synced 2025-03-11 08:37:59 +00:00
ao_sdl: use new pull API helpers
One strange issue is that we apparently can't stop the audio API on audio reset (ao_driver.reset). We could use SDL_PauseAudio, but that doesn't specify whether remaining audio is dropped. We also could use SDL_LockAudio, but holding that over a long time will probably be bad, and it probably doesn't drop audio. This means we simply play silence after a reset, instead of stopping the callback completely. (The existing code ran into an underrun in this situation.) The delay estimation works about the same. We simply assume that the callback is locked to audio timing (like ao_jack), and that 1 callback corresponds to 1 period. It seems this (removed) code fragment assumes there 1 one period size delay: // delay subcomponent: remaining audio from the next played buffer, as // provided by the callback buffer_interval += callback_interval; so we explicitly do that too.
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wm4
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3ca034228d
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@ -24,58 +24,34 @@
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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 <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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void *data[1] = {stream};
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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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if (len % ao->sstride)
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MP_ERR(ao, "SDL audio callback not sample aligned");
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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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// 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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SDL_UnlockMutex(priv->buffer_mutex);
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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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@ -84,34 +60,13 @@ static void uninit(struct ao *ao)
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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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@ -172,7 +127,7 @@ static int init(struct ao *ao)
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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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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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@ -227,125 +182,26 @@ static int init(struct ao *ao)
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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);
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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);
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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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if (!priv->paused)
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SDL_PauseAudio(SDL_TRUE);
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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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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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@ -355,20 +211,14 @@ const struct ao_driver audio_out_sdl = {
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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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