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mpv/audio/out/internal.h
wm4 90dd229871 audio/out: add helper code to do 24 bit conversion in AO
I plan to remove the S24 sample formats in mpv. It seems like we should
still support this _somehow_ in AOs though. So the idea is to convert
the data to more obscure representations (that would not be useful for
filtering etc. anyway) within the AO.

This commit adds helper to enable this. ao_convert_fmt is meant to
provide mechanisms for this, rather than a generic audio format
description (as the latter leads only to overly generic misery). The
conversion also supports only cases which we think will be needed at
all.

The main advantage of this approach is that we get S24 out of sight,
and that we could support other crazy formats (like S20). The main
disadvantage is that usually S32 will be selected (if both S32 and S24
are available), and there's no user control to force S24. That doesn't
really matter though, and at worst makes testing harder or will lead
to unpleasant arguments with audiophiles (they'd be wrong anyway).

ao_convert_fmt.pad_lsb is ignored, although if we ever find a case in
which playing S32 with data in the LSBs breaks when playing it as padded
24 bit format. (For example, WAVEFORMATEXTENSIBLE recommends setting the
unused bits to 0 if wValidBitsPerSample implies LSB padding.)
2017-07-07 17:54:05 +02:00

234 lines
9.3 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 *api; // entrypoints to the wrapper (push.c/pull.c)
const struct ao_driver *driver;
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 is typically the period size, and the
// smallest unit the driver will accept in one piece (although if
// AOPLAY_FINAL_CHUNK is set, the driver must accept everything).
// 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;
// Device actually chosen by the AO
char *detected_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_int events_;
int buffer;
double def_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
* 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
* 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()
double (*get_delay)(struct ao *ao);
// push based: block until all queued audio is played (optional)
void (*drain)(struct ao *ao);
// Optional. Return true if audio has stopped in any way.
bool (*get_eof)(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);
// 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_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_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);
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);
int ao_read_data_converted(struct ao *ao, struct ao_convert_fmt *fmt,
void **data, int samples, int64_t out_time_us);
#endif