mpv/audio/out/internal.h

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/*
* This file is part of MPlayer.
*
* 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.
*/
#ifndef MP_AO_INTERNAL_H_
#define MP_AO_INTERNAL_H_
#include <stdbool.h>
#include <pthread.h>
#include "audio/chmap.h"
#include "audio/chmap_sel.h"
// Minimum buffer size in seconds.
#define MIN_BUFFER 0.2
// If ao_get_delay() reaches this value after ao_play() was called with the
// AOPLAY_FINAL_CHUNK flag set, the playback core expects that the audio has
// all been played.
#define AO_EOF_DELAY 0.05
/* global data used by ao.c and ao drivers */
struct ao {
int samplerate;
struct mp_chmap channels;
int format; // one of AF_FORMAT_...
int bps; // bytes per second (per plane)
int sstride; // size of a sample on each plane
// (format_size*num_channels/num_planes)
int num_planes;
bool probing; // if true, don't fail loudly on init
bool untimed; // don't assume realtime playback
bool no_persistent_volume; // the AO does the equivalent of af_volume
bool per_application_mixer; // like above, but volume persists (per app)
int device_buffer; // device buffer in samples (guessed by
// common init code if not set by driver)
const struct ao_driver *api; // entrypoints to the wrapper (push.c/pull.c)
const struct ao_driver *driver;
void *priv;
struct encode_lavc_context *encode_lavc_ctx;
struct input_ctx *input_ctx;
struct mp_log *log; // Using e.g. "[ao/coreaudio]" as prefix
int buffer;
void *api_priv;
};
extern const struct ao_driver ao_api_push;
extern const struct ao_driver ao_api_pull;
/* Note:
*
* In general, there are two types of audio drivers:
* a) push based (the user queues data that should be played)
* b) pull callback based (the audio API calls a callback to get audio)
*
* The ao.c code can handle both. It basically implements two audio paths
* and provides a uniform API for them. If ao_driver->play is NULL, it assumes
* that the driver uses a callback based audio API, otherwise push based.
*
* Requirements:
* a) ->play is called to queue audio. push.c creates a thread to regularly
* refill audio device buffers with ->play, but all driver functions are
* always called under an exclusive lock.
* Mandatory:
* init
* uninit
* reset
* get_space
* play
* get_delay
* pause
* resume
audio/out/pull: remove race conditions There were subtle and minor race conditions in the pull.c code, and AOs using it (jack, portaudio, sdl, wasapi). Attempt to remove these. There was at least a race condition in the ao_reset() implementation: mp_ring_reset() was called concurrently to the audio callback. While the ringbuffer uses atomics to allow concurrent access, the reset function wasn't concurrency-safe (and can't easily be made to). Fix this by stopping the audio callback before doing a reset. After that, we can do anything without needing synchronization. The callback is resumed when resuming playback at a later point. Don't call driver->pause, and make driver->resume and driver->reset start/stop the audio callback. In the initial state, the audio callback must be disabled. JackAudio of course is different. Maybe there is no way to suspend the audio callback without "disconnecting" it (what jack_deactivate() would do), so I'm not trying my luck, and implemented a really bad hack doing active waiting until we get the audio callback into a state where it won't interfere. Once the callback goes from AO_STATE_WAIT to NONE, we can be sure that the callback doesn't access the ringbuffer or anything else anymore. Since both sched_yield() and pthread_yield() apparently are not always available, use mp_sleep_us(1) to avoid burning CPU during active waiting. The ao_jack.c change also removes a race condition: apparently we didn't initialize _all_ ao fields before starting the audio callback. In ao_wasapi.c, I'm not sure whether reset really waits for the audio callback to return. Kovensky says it's not guaranteed, so disable the reset callback - for now the behavior of ao_wasapi.c is like with ao_jack.c, and active waiting is used to deal with the audio callback.
2014-05-29 00:24:17 +00:00
* Optional:
* control
* drain
* wait
* wakeup
* b) ->play must be NULL. ->resume must be provided, and should make the
* audio API start calling the audio callback. Your audio callback should
* in turn call ao_read_data() to get audio data. Most functions are
* optional and will be emulated if missing (e.g. pausing is emulated as
* silence). ->get_delay and ->get_space are never called.
* Mandatory:
* init
* uninit
audio/out/pull: remove race conditions There were subtle and minor race conditions in the pull.c code, and AOs using it (jack, portaudio, sdl, wasapi). Attempt to remove these. There was at least a race condition in the ao_reset() implementation: mp_ring_reset() was called concurrently to the audio callback. While the ringbuffer uses atomics to allow concurrent access, the reset function wasn't concurrency-safe (and can't easily be made to). Fix this by stopping the audio callback before doing a reset. After that, we can do anything without needing synchronization. The callback is resumed when resuming playback at a later point. Don't call driver->pause, and make driver->resume and driver->reset start/stop the audio callback. In the initial state, the audio callback must be disabled. JackAudio of course is different. Maybe there is no way to suspend the audio callback without "disconnecting" it (what jack_deactivate() would do), so I'm not trying my luck, and implemented a really bad hack doing active waiting until we get the audio callback into a state where it won't interfere. Once the callback goes from AO_STATE_WAIT to NONE, we can be sure that the callback doesn't access the ringbuffer or anything else anymore. Since both sched_yield() and pthread_yield() apparently are not always available, use mp_sleep_us(1) to avoid burning CPU during active waiting. The ao_jack.c change also removes a race condition: apparently we didn't initialize _all_ ao fields before starting the audio callback. In ao_wasapi.c, I'm not sure whether reset really waits for the audio callback to return. Kovensky says it's not guaranteed, so disable the reset callback - for now the behavior of ao_wasapi.c is like with ao_jack.c, and active waiting is used to deal with the audio callback.
2014-05-29 00:24:17 +00:00
* resume (starts the audio callback)
* Also, the following optional callbacks can be provided:
* reset (stops the audio callback, resume() restarts it)
* control
*/
struct ao_driver {
// If true, use with encoding only.
bool encode;
// Name used for --ao.
const char *name;
// Description shown with --ao=help.
const char *description;
// Init the device using ao->format/ao->channels/ao->samplerate. If the
// device doesn't accept these parameters, you can attempt to negotiate
// fallback parameters, and set the ao format fields accordingly.
int (*init)(struct ao *ao);
// Optional. See ao_control() etc. in ao.c
int (*control)(struct ao *ao, enum aocontrol cmd, void *arg);
void (*uninit)(struct ao *ao);
// push based: see ao_reset()
// pull based: stop the audio callback
void (*reset)(struct ao*ao);
// push based: see ao_pause()
void (*pause)(struct ao *ao);
// push based: see ao_resume()
// pull based: start the audio callback
void (*resume)(struct ao *ao);
// push based: see ao_play()
int (*get_space)(struct ao *ao);
// push based: see ao_play()
int (*play)(struct ao *ao, void **data, int samples, int flags);
// push based: see ao_get_delay()
float (*get_delay)(struct ao *ao);
// push based: block until all queued audio is played (optional)
void (*drain)(struct ao *ao);
// Wait until the audio buffer needs to be refilled. The lock is the
// internal mutex usually protecting the internal AO state (and used to
// protect driver calls), and must be temporarily unlocked while waiting.
// ->wakeup will be called (with lock held) if the wait should be canceled.
// Returns 0 on success, -1 on error.
// Optional; if this is not provided, generic code using audio timing is
// used to estimate when the AO needs to be refilled.
// Warning: it's only called if the feed thread truly needs to know when
// the audio thread takes data again. Often, it will just copy
// the complete soft-buffer to the AO, and then wait for the
// decoder instead. Don't do necessary work in this callback.
int (*wait)(struct ao *ao, pthread_mutex_t *lock);
// In combination with wait(). Lock may or may not be held.
void (*wakeup)(struct ao *ao);
// For option parsing (see vo.h)
int priv_size;
const void *priv_defaults;
const struct m_option *options;
};
// These functions can be called by AOs.
int ao_play_silence(struct ao *ao, int samples);
int ao_read_data(struct ao *ao, void **data, int samples, int64_t out_time_us);
struct pollfd;
int ao_wait_poll(struct ao *ao, struct pollfd *fds, int num_fds,
pthread_mutex_t *lock);
void ao_wakeup_poll(struct ao *ao);
bool ao_chmap_sel_adjust(struct ao *ao, const struct mp_chmap_sel *s,
struct mp_chmap *map);
bool ao_chmap_sel_get_def(struct ao *ao, const struct mp_chmap_sel *s,
struct mp_chmap *map, int num);
#endif