mirror of https://github.com/mpv-player/mpv
235 lines
9.8 KiB
C
235 lines
9.8 KiB
C
/*
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* This file is part of mpv.
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*
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* mpv is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* mpv 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 Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with mpv. If not, see <http://www.gnu.org/licenses/>.
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*/
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#ifndef MP_AO_INTERNAL_H_
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#define MP_AO_INTERNAL_H_
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#include <stdbool.h>
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#include <pthread.h>
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#include "osdep/atomic.h"
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#include "audio/out/ao.h"
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/* global data used by ao.c and ao drivers */
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struct ao {
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int samplerate;
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struct mp_chmap channels;
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int format; // one of AF_FORMAT_...
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int bps; // bytes per second (per plane)
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int sstride; // size of a sample on each plane
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// (format_size*num_channels/num_planes)
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int num_planes;
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bool probing; // if true, don't fail loudly on init
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bool untimed; // don't assume realtime playback
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int device_buffer; // device buffer in samples (guessed by
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// common init code if not set by driver)
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const struct ao_driver *driver;
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bool driver_initialized;
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void *priv;
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struct mpv_global *global;
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struct encode_lavc_context *encode_lavc_ctx;
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void (*wakeup_cb)(void *ctx);
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void *wakeup_ctx;
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struct mp_log *log; // Using e.g. "[ao/coreaudio]" as prefix
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int init_flags; // AO_INIT_* flags
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bool stream_silence; // if audio inactive, just play silence
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// Set by the driver on init.
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// This value is in complete samples (i.e. 1 for stereo means 1 sample
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// for both channels each).
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// Used for push based API only.
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int period_size;
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// The device as selected by the user, usually using ao_device_desc.name
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// from an entry from the list returned by driver->list_devices. If the
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// default device should be used, this is set to NULL.
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char *device;
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// Application name to report to the audio API.
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char *client_name;
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// Used during init: if init fails, redirect to this ao
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char *redirect;
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// Internal events (use ao_request_reload(), ao_hotplug_event())
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atomic_uint events_;
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// Float gain multiplicator
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mp_atomic_float gain;
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int buffer;
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double def_buffer;
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struct buffer_state *buffer_state;
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void *api_priv;
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};
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void init_buffer_pre(struct ao *ao);
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bool init_buffer_post(struct ao *ao);
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struct mp_pcm_state {
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// Note: free_samples+queued_samples <= ao->device_buffer; the sum may be
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// less if the audio API can report partial periods played, while
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// free_samples should be period-size aligned.
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int free_samples; // number of free space in ring buffer
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int queued_samples; // number of samples to play in ring buffer
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double delay; // total latency in seconds (includes queued_samples)
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bool playing; // set if underlying API is actually playing audio;
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// the AO must unset it on underrun (accidental
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// underrun and EOF are indistinguishable; the upper
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// layers decide what it was)
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// real pausing may assume playing=true
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};
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/* Note:
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*
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* In general, there are two types of audio drivers:
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* a) push based (the user queues data that should be played)
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* b) pull callback based (the audio API calls a callback to get audio)
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*
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* The ao.c code can handle both. It basically implements two audio paths
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* and provides a uniform API for them. If ao_driver->write is NULL, it assumes
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* that the driver uses a callback based audio API, otherwise push based.
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*
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* Requirements:
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* a+b) Mandatory for both types:
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* init
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* uninit
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* start
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* Optional for both types:
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* control
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* a) ->write is called to queue audio. push.c creates a thread to regularly
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* refill audio device buffers with ->write, but all driver functions are
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* always called under an exclusive lock.
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* Mandatory:
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* reset
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* write
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* get_state
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* Optional:
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* set_pause
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* b) ->write must be NULL. ->start must be provided, and should make the
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* audio API start calling the audio callback. Your audio callback should
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* in turn call ao_read_data() to get audio data. Most functions are
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* optional and will be emulated if missing (e.g. pausing is emulated as
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* silence).
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* Also, the following optional callbacks can be provided:
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* reset (stops the audio callback, start() restarts it)
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*/
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struct ao_driver {
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// If true, use with encoding only.
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bool encode;
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// Name used for --ao.
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const char *name;
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// Description shown with --ao=help.
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const char *description;
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// This requires waiting for a AO_EVENT_INITIAL_UNBLOCK event before the
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// first write() call is done. Encode mode uses this, and push mode
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// respects it automatically (don't use with pull mode).
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bool initially_blocked;
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// Init the device using ao->format/ao->channels/ao->samplerate. If the
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// device doesn't accept these parameters, you can attempt to negotiate
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// fallback parameters, and set the ao format fields accordingly.
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int (*init)(struct ao *ao);
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// Optional. See ao_control() etc. in ao.c
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int (*control)(struct ao *ao, enum aocontrol cmd, void *arg);
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void (*uninit)(struct ao *ao);
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// Stop all audio playback, clear buffers, back to state after init().
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// Optional for pull AOs.
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void (*reset)(struct ao *ao);
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// push based: set pause state. Only called after start() and before reset().
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// returns success (this is intended for paused=true; if it
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// returns false, playback continues, and the core emulates via
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// reset(); unpausing always works)
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bool (*set_pause)(struct ao *ao, bool paused);
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// pull based: start the audio callback
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// push based: start playing queued data
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// AO should call ao_wakeup_playthread() if a period boundary
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// is crossed, or playback stops due to external reasons
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// (including underruns or device removal)
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// must set mp_pcm_state.playing; unset on error/underrun/end
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void (*start)(struct ao *ao);
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// push based: queue new data. This won't try to write more data than the
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// reported free space (samples <= mp_pcm_state.free_samples).
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// This must NOT start playback. start() does that, and write() may be
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// called multiple times before start() is called. It may also happen that
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// reset() is called to discard the buffer. start() without write() will
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// immediately reported an underrun.
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// Return false on failure.
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bool (*write)(struct ao *ao, void **data, int samples);
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// push based: return mandatory stream information
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void (*get_state)(struct ao *ao, struct mp_pcm_state *state);
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// Return the list of devices currently available in the system. Use
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// ao_device_list_add() to add entries. The selected device will be set as
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// ao->device (using ao_device_desc.name).
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// Warning: the ao struct passed is not initialized with ao_driver->init().
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// Instead, hotplug_init/hotplug_uninit is called. If these
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// callbacks are not set, no driver initialization call is done
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// on the ao struct.
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void (*list_devs)(struct ao *ao, struct ao_device_list *list);
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// If set, these are called before/after ao_driver->list_devs is called.
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// It is also assumed that the driver can do hotplugging - which means
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// it is expected to call ao_hotplug_event(ao) whenever the system's
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// audio device list changes. The player will then call list_devs() again.
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int (*hotplug_init)(struct ao *ao);
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void (*hotplug_uninit)(struct ao *ao);
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// For option parsing (see vo.h)
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int priv_size;
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const void *priv_defaults;
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const struct m_option *options;
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const char *options_prefix;
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const struct m_sub_options *global_opts;
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};
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// These functions can be called by AOs.
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int ao_read_data(struct ao *ao, void **data, int samples, int64_t out_time_us);
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bool ao_chmap_sel_adjust(struct ao *ao, const struct mp_chmap_sel *s,
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struct mp_chmap *map);
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bool ao_chmap_sel_adjust2(struct ao *ao, const struct mp_chmap_sel *s,
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struct mp_chmap *map, bool safe_multichannel);
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bool ao_chmap_sel_get_def(struct ao *ao, const struct mp_chmap_sel *s,
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struct mp_chmap *map, int num);
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// Add a deep copy of e to the list.
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// Call from ao_driver->list_devs callback only.
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void ao_device_list_add(struct ao_device_list *list, struct ao *ao,
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struct ao_device_desc *e);
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void ao_post_process_data(struct ao *ao, void **data, int num_samples);
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struct ao_convert_fmt {
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int src_fmt; // source AF_FORMAT_*
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int channels; // number of channels
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int dst_bits; // total target data sample size
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int pad_msb; // padding in the MSB (i.e. required shifting)
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int pad_lsb; // padding in LSB (required 0 bits) (ignored)
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};
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bool ao_can_convert_inplace(struct ao_convert_fmt *fmt);
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bool ao_need_conversion(struct ao_convert_fmt *fmt);
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void ao_convert_inplace(struct ao_convert_fmt *fmt, void **data, int num_samples);
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void ao_wakeup_playthread(struct ao *ao);
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int ao_read_data_converted(struct ao *ao, struct ao_convert_fmt *fmt,
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void **data, int samples, int64_t out_time_us);
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#endif
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