mpv/audio/out/internal.h

235 lines
9.8 KiB
C

/*
* 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/>.
*/
#ifndef MP_AO_INTERNAL_H_
#define MP_AO_INTERNAL_H_
#include <stdbool.h>
#include <pthread.h>
#include "osdep/atomic.h"
#include "audio/out/ao.h"
/* 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
int device_buffer; // device buffer in samples (guessed by
// common init code if not set by driver)
const struct ao_driver *driver;
bool driver_initialized;
void *priv;
struct mpv_global *global;
struct encode_lavc_context *encode_lavc_ctx;
void (*wakeup_cb)(void *ctx);
void *wakeup_ctx;
struct mp_log *log; // Using e.g. "[ao/coreaudio]" as prefix
int init_flags; // AO_INIT_* flags
bool stream_silence; // if audio inactive, just play silence
// Set by the driver on init.
// This value is in complete samples (i.e. 1 for stereo means 1 sample
// for both channels each).
// Used for push based API only.
int period_size;
// The device as selected by the user, usually using ao_device_desc.name
// from an entry from the list returned by driver->list_devices. If the
// default device should be used, this is set to NULL.
char *device;
// Application name to report to the audio API.
char *client_name;
// Used during init: if init fails, redirect to this ao
char *redirect;
// Internal events (use ao_request_reload(), ao_hotplug_event())
atomic_uint events_;
// Float gain multiplicator
mp_atomic_float gain;
int buffer;
double def_buffer;
struct buffer_state *buffer_state;
void *api_priv;
};
void init_buffer_pre(struct ao *ao);
bool init_buffer_post(struct ao *ao);
struct mp_pcm_state {
// Note: free_samples+queued_samples <= ao->device_buffer; the sum may be
// less if the audio API can report partial periods played, while
// free_samples should be period-size aligned.
int free_samples; // number of free space in ring buffer
int queued_samples; // number of samples to play in ring buffer
double delay; // total latency in seconds (includes queued_samples)
bool playing; // set if underlying API is actually playing audio;
// the AO must unset it on underrun (accidental
// underrun and EOF are indistinguishable; the upper
// layers decide what it was)
// real pausing may assume playing=true
};
/* 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->write is NULL, it assumes
* that the driver uses a callback based audio API, otherwise push based.
*
* Requirements:
* a+b) Mandatory for both types:
* init
* uninit
* start
* Optional for both types:
* control
* a) ->write is called to queue audio. push.c creates a thread to regularly
* refill audio device buffers with ->write, but all driver functions are
* always called under an exclusive lock.
* Mandatory:
* reset
* write
* get_state
* Optional:
* set_pause
* b) ->write must be NULL. ->start 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).
* Also, the following optional callbacks can be provided:
* reset (stops the audio callback, start() restarts it)
*/
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;
// This requires waiting for a AO_EVENT_INITIAL_UNBLOCK event before the
// first write() call is done. Encode mode uses this, and push mode
// respects it automatically (don't use with pull mode).
bool initially_blocked;
// 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);
// Stop all audio playback, clear buffers, back to state after init().
// Optional for pull AOs.
void (*reset)(struct ao *ao);
// push based: set pause state. Only called after start() and before reset().
// returns success (this is intended for paused=true; if it
// returns false, playback continues, and the core emulates via
// reset(); unpausing always works)
bool (*set_pause)(struct ao *ao, bool paused);
// pull based: start the audio callback
// push based: start playing queued data
// AO should call ao_wakeup_playthread() if a period boundary
// is crossed, or playback stops due to external reasons
// (including underruns or device removal)
// must set mp_pcm_state.playing; unset on error/underrun/end
void (*start)(struct ao *ao);
// push based: queue new data. This won't try to write more data than the
// reported free space (samples <= mp_pcm_state.free_samples).
// This must NOT start playback. start() does that, and write() may be
// called multiple times before start() is called. It may also happen that
// reset() is called to discard the buffer. start() without write() will
// immediately reported an underrun.
// Return false on failure.
bool (*write)(struct ao *ao, void **data, int samples);
// push based: return mandatory stream information
void (*get_state)(struct ao *ao, struct mp_pcm_state *state);
// Return the list of devices currently available in the system. Use
// ao_device_list_add() to add entries. The selected device will be set as
// ao->device (using ao_device_desc.name).
// Warning: the ao struct passed is not initialized with ao_driver->init().
// Instead, hotplug_init/hotplug_uninit is called. If these
// callbacks are not set, no driver initialization call is done
// on the ao struct.
void (*list_devs)(struct ao *ao, struct ao_device_list *list);
// If set, these are called before/after ao_driver->list_devs is called.
// It is also assumed that the driver can do hotplugging - which means
// it is expected to call ao_hotplug_event(ao) whenever the system's
// audio device list changes. The player will then call list_devs() again.
int (*hotplug_init)(struct ao *ao);
void (*hotplug_uninit)(struct ao *ao);
// For option parsing (see vo.h)
int priv_size;
const void *priv_defaults;
const struct m_option *options;
const char *options_prefix;
const struct m_sub_options *global_opts;
};
// These functions can be called by AOs.
int ao_read_data(struct ao *ao, void **data, int samples, int64_t out_time_us);
bool ao_chmap_sel_adjust(struct ao *ao, const struct mp_chmap_sel *s,
struct mp_chmap *map);
bool ao_chmap_sel_adjust2(struct ao *ao, const struct mp_chmap_sel *s,
struct mp_chmap *map, bool safe_multichannel);
bool ao_chmap_sel_get_def(struct ao *ao, const struct mp_chmap_sel *s,
struct mp_chmap *map, int num);
// Add a deep copy of e to the list.
// Call from ao_driver->list_devs callback only.
void ao_device_list_add(struct ao_device_list *list, struct ao *ao,
struct ao_device_desc *e);
void ao_post_process_data(struct ao *ao, void **data, int num_samples);
struct ao_convert_fmt {
int src_fmt; // source AF_FORMAT_*
int channels; // number of channels
int dst_bits; // total target data sample size
int pad_msb; // padding in the MSB (i.e. required shifting)
int pad_lsb; // padding in LSB (required 0 bits) (ignored)
};
bool ao_can_convert_inplace(struct ao_convert_fmt *fmt);
bool ao_need_conversion(struct ao_convert_fmt *fmt);
void ao_convert_inplace(struct ao_convert_fmt *fmt, void **data, int num_samples);
void ao_wakeup_playthread(struct ao *ao);
int ao_read_data_converted(struct ao *ao, struct ao_convert_fmt *fmt,
void **data, int samples, int64_t out_time_us);
#endif