mirror of https://github.com/mpv-player/mpv
797 lines
24 KiB
C
797 lines
24 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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#include <stdio.h>
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#include <stdlib.h>
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#include <string.h>
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#include <math.h>
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#include <assert.h>
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#include "mpv_talloc.h"
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#include "config.h"
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#include "ao.h"
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#include "internal.h"
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#include "audio/format.h"
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#include "options/options.h"
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#include "options/m_config.h"
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#include "osdep/endian.h"
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#include "common/msg.h"
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#include "common/common.h"
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#include "common/global.h"
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extern const struct ao_driver audio_out_oss;
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extern const struct ao_driver audio_out_audiotrack;
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extern const struct ao_driver audio_out_audiounit;
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extern const struct ao_driver audio_out_coreaudio;
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extern const struct ao_driver audio_out_coreaudio_exclusive;
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extern const struct ao_driver audio_out_rsound;
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extern const struct ao_driver audio_out_sndio;
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extern const struct ao_driver audio_out_pulse;
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extern const struct ao_driver audio_out_jack;
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extern const struct ao_driver audio_out_openal;
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extern const struct ao_driver audio_out_opensles;
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extern const struct ao_driver audio_out_null;
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extern const struct ao_driver audio_out_alsa;
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extern const struct ao_driver audio_out_wasapi;
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extern const struct ao_driver audio_out_pcm;
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extern const struct ao_driver audio_out_lavc;
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extern const struct ao_driver audio_out_sdl;
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static const struct ao_driver * const audio_out_drivers[] = {
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// native:
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#if HAVE_ANDROID
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&audio_out_audiotrack,
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#endif
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#if HAVE_AUDIOUNIT
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&audio_out_audiounit,
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#endif
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#if HAVE_COREAUDIO
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&audio_out_coreaudio,
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#endif
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#if HAVE_PULSE
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&audio_out_pulse,
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#endif
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#if HAVE_ALSA
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&audio_out_alsa,
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#endif
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#if HAVE_WASAPI
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&audio_out_wasapi,
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#endif
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#if HAVE_OSS_AUDIO
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&audio_out_oss,
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#endif
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// wrappers:
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#if HAVE_JACK
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&audio_out_jack,
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#endif
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#if HAVE_OPENAL
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&audio_out_openal,
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#endif
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#if HAVE_OPENSLES
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&audio_out_opensles,
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#endif
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#if HAVE_SDL2_AUDIO
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&audio_out_sdl,
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#endif
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#if HAVE_SNDIO
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&audio_out_sndio,
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#endif
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&audio_out_null,
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#if HAVE_COREAUDIO
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&audio_out_coreaudio_exclusive,
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#endif
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&audio_out_pcm,
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&audio_out_lavc,
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#if HAVE_RSOUND
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&audio_out_rsound,
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#endif
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NULL
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};
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static bool get_desc(struct m_obj_desc *dst, int index)
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{
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if (index >= MP_ARRAY_SIZE(audio_out_drivers) - 1)
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return false;
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const struct ao_driver *ao = audio_out_drivers[index];
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*dst = (struct m_obj_desc) {
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.name = ao->name,
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.description = ao->description,
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.priv_size = ao->priv_size,
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.priv_defaults = ao->priv_defaults,
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.options = ao->options,
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.options_prefix = ao->options_prefix,
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.global_opts = ao->global_opts,
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.hidden = ao->encode,
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.p = ao,
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};
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return true;
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}
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// For the ao option
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static const struct m_obj_list ao_obj_list = {
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.get_desc = get_desc,
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.description = "audio outputs",
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.allow_unknown_entries = true,
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.allow_trailer = true,
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.disallow_positional_parameters = true,
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.use_global_options = true,
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};
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#define OPT_BASE_STRUCT struct ao_opts
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const struct m_sub_options ao_conf = {
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.opts = (const struct m_option[]) {
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OPT_SETTINGSLIST("ao", audio_driver_list, UPDATE_AUDIO, &ao_obj_list, ),
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OPT_STRING("audio-device", audio_device, UPDATE_AUDIO),
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OPT_STRING("audio-client-name", audio_client_name, UPDATE_AUDIO),
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OPT_DOUBLE("audio-buffer", audio_buffer,
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UPDATE_AUDIO | M_OPT_MIN | M_OPT_MAX, .min = 0, .max = 10),
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{0}
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},
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.size = sizeof(OPT_BASE_STRUCT),
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.defaults = &(const OPT_BASE_STRUCT){
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.audio_buffer = 0.2,
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.audio_device = "auto",
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.audio_client_name = "mpv",
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},
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};
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static struct ao *ao_alloc(bool probing, struct mpv_global *global,
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void (*wakeup_cb)(void *ctx), void *wakeup_ctx,
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char *name)
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{
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assert(wakeup_cb);
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struct mp_log *log = mp_log_new(NULL, global->log, "ao");
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struct m_obj_desc desc;
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if (!m_obj_list_find(&desc, &ao_obj_list, bstr0(name))) {
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mp_msg(log, MSGL_ERR, "Audio output %s not found!\n", name);
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talloc_free(log);
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return NULL;
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};
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struct ao_opts *opts = mp_get_config_group(NULL, global, &ao_conf);
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struct ao *ao = talloc_ptrtype(NULL, ao);
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talloc_steal(ao, log);
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*ao = (struct ao) {
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.driver = desc.p,
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.probing = probing,
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.global = global,
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.wakeup_cb = wakeup_cb,
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.wakeup_ctx = wakeup_ctx,
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.log = mp_log_new(ao, log, name),
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.def_buffer = opts->audio_buffer,
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.client_name = talloc_strdup(ao, opts->audio_client_name),
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};
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talloc_free(opts);
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ao->priv = m_config_group_from_desc(ao, ao->log, global, &desc, name);
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if (!ao->priv)
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goto error;
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ao_set_gain(ao, 1.0f);
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return ao;
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error:
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talloc_free(ao);
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return NULL;
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}
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static struct ao *ao_init(bool probing, struct mpv_global *global,
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void (*wakeup_cb)(void *ctx), void *wakeup_ctx,
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struct encode_lavc_context *encode_lavc_ctx, int flags,
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int samplerate, int format, struct mp_chmap channels,
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char *dev, char *name)
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{
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struct ao *ao = ao_alloc(probing, global, wakeup_cb, wakeup_ctx, name);
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if (!ao)
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return NULL;
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ao->samplerate = samplerate;
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ao->channels = channels;
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ao->format = format;
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ao->encode_lavc_ctx = encode_lavc_ctx;
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ao->init_flags = flags;
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if (ao->driver->encode != !!ao->encode_lavc_ctx)
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goto fail;
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MP_VERBOSE(ao, "requested format: %d Hz, %s channels, %s\n",
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ao->samplerate, mp_chmap_to_str(&ao->channels),
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af_fmt_to_str(ao->format));
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ao->device = talloc_strdup(ao, dev);
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ao->api = ao->driver->play ? &ao_api_push : &ao_api_pull;
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ao->api_priv = talloc_zero_size(ao, ao->api->priv_size);
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assert(!ao->api->priv_defaults && !ao->api->options);
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ao->stream_silence = flags & AO_INIT_STREAM_SILENCE;
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ao->period_size = 1;
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int r = ao->driver->init(ao);
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if (r < 0) {
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// Silly exception for coreaudio spdif redirection
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if (ao->redirect) {
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char redirect[80], rdevice[80];
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snprintf(redirect, sizeof(redirect), "%s", ao->redirect);
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snprintf(rdevice, sizeof(rdevice), "%s", ao->device ? ao->device : "");
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talloc_free(ao);
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return ao_init(probing, global, wakeup_cb, wakeup_ctx,
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encode_lavc_ctx, flags, samplerate, format, channels,
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rdevice, redirect);
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}
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goto fail;
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}
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if (ao->period_size < 1) {
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MP_ERR(ao, "Invalid period size set.\n");
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goto fail;
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}
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ao->sstride = af_fmt_to_bytes(ao->format);
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ao->num_planes = 1;
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if (af_fmt_is_planar(ao->format)) {
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ao->num_planes = ao->channels.num;
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} else {
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ao->sstride *= ao->channels.num;
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}
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ao->bps = ao->samplerate * ao->sstride;
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if (!ao->device_buffer && ao->driver->get_space)
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ao->device_buffer = ao->driver->get_space(ao);
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if (ao->device_buffer)
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MP_VERBOSE(ao, "device buffer: %d samples.\n", ao->device_buffer);
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ao->buffer = MPMAX(ao->device_buffer, ao->def_buffer * ao->samplerate);
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ao->buffer = MPMAX(ao->buffer, 1);
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int align = af_format_sample_alignment(ao->format);
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ao->buffer = (ao->buffer + align - 1) / align * align;
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MP_VERBOSE(ao, "using soft-buffer of %d samples.\n", ao->buffer);
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if (ao->api->init(ao) < 0)
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goto fail;
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return ao;
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fail:
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talloc_free(ao);
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return NULL;
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}
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static void split_ao_device(void *tmp, char *opt, char **out_ao, char **out_dev)
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{
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*out_ao = NULL;
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*out_dev = NULL;
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if (!opt)
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return;
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if (!opt[0] || strcmp(opt, "auto") == 0)
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return;
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// Split on "/". If "/" is the final character, or absent, out_dev is NULL.
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bstr b_dev, b_ao;
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bstr_split_tok(bstr0(opt), "/", &b_ao, &b_dev);
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if (b_dev.len > 0)
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*out_dev = bstrto0(tmp, b_dev);
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*out_ao = bstrto0(tmp, b_ao);
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}
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struct ao *ao_init_best(struct mpv_global *global,
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int init_flags,
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void (*wakeup_cb)(void *ctx), void *wakeup_ctx,
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struct encode_lavc_context *encode_lavc_ctx,
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int samplerate, int format, struct mp_chmap channels)
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{
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void *tmp = talloc_new(NULL);
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struct ao_opts *opts = mp_get_config_group(tmp, global, &ao_conf);
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struct mp_log *log = mp_log_new(tmp, global->log, "ao");
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struct ao *ao = NULL;
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struct m_obj_settings *ao_list = NULL;
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int ao_num = 0;
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for (int n = 0; opts->audio_driver_list && opts->audio_driver_list[n].name; n++)
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MP_TARRAY_APPEND(tmp, ao_list, ao_num, opts->audio_driver_list[n]);
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bool forced_dev = false;
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char *pref_ao, *pref_dev;
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split_ao_device(tmp, opts->audio_device, &pref_ao, &pref_dev);
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if (!ao_num && pref_ao) {
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// Reuse the autoselection code
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MP_TARRAY_APPEND(tmp, ao_list, ao_num,
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(struct m_obj_settings){.name = pref_ao});
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forced_dev = true;
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}
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bool autoprobe = ao_num == 0;
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// Something like "--ao=a,b," means do autoprobing after a and b fail.
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if (ao_num && strlen(ao_list[ao_num - 1].name) == 0) {
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ao_num -= 1;
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autoprobe = true;
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}
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if (autoprobe) {
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for (int n = 0; audio_out_drivers[n]; n++) {
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const struct ao_driver *driver = audio_out_drivers[n];
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if (driver == &audio_out_null)
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break;
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MP_TARRAY_APPEND(tmp, ao_list, ao_num,
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(struct m_obj_settings){.name = (char *)driver->name});
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}
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}
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if (init_flags & AO_INIT_NULL_FALLBACK) {
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MP_TARRAY_APPEND(tmp, ao_list, ao_num,
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(struct m_obj_settings){.name = "null"});
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}
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for (int n = 0; n < ao_num; n++) {
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struct m_obj_settings *entry = &ao_list[n];
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bool probing = n + 1 != ao_num;
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mp_verbose(log, "Trying audio driver '%s'\n", entry->name);
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char *dev = NULL;
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if (pref_ao && pref_dev && strcmp(entry->name, pref_ao) == 0) {
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dev = pref_dev;
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mp_verbose(log, "Using preferred device '%s'\n", dev);
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}
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ao = ao_init(probing, global, wakeup_cb, wakeup_ctx, encode_lavc_ctx,
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init_flags, samplerate, format, channels, dev, entry->name);
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if (ao)
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break;
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if (!probing)
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mp_err(log, "Failed to initialize audio driver '%s'\n", entry->name);
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if (dev && forced_dev) {
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mp_err(log, "This audio driver/device was forced with the "
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"--audio-device option.\nTry unsetting it.\n");
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}
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}
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talloc_free(tmp);
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return ao;
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}
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// Uninitialize and destroy the AO. Remaining audio must be dropped.
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void ao_uninit(struct ao *ao)
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{
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if (ao)
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ao->api->uninit(ao);
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talloc_free(ao);
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}
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// Queue the given audio data. Start playback if it hasn't started yet. Return
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// the number of samples that was accepted (the core will try to queue the rest
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// again later). Should never block.
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// data: start pointer for each plane. If the audio data is packed, only
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// data[0] is valid, otherwise there is a plane for each channel.
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// samples: size of the audio data (see ao->sstride)
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// flags: currently AOPLAY_FINAL_CHUNK can be set
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int ao_play(struct ao *ao, void **data, int samples, int flags)
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{
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return ao->api->play(ao, data, samples, flags);
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}
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int ao_control(struct ao *ao, enum aocontrol cmd, void *arg)
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{
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return ao->api->control ? ao->api->control(ao, cmd, arg) : CONTROL_UNKNOWN;
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}
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// Return size of the buffered data in seconds. Can include the device latency.
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// Basically, this returns how much data there is still to play, and how long
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// it takes until the last sample in the buffer reaches the speakers. This is
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// used for audio/video synchronization, so it's very important to implement
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// this correctly.
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double ao_get_delay(struct ao *ao)
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{
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return ao->api->get_delay(ao);
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}
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// Return free size of the internal audio buffer. This controls how much audio
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// the core should decode and try to queue with ao_play().
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int ao_get_space(struct ao *ao)
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{
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return ao->api->get_space(ao);
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}
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// Stop playback and empty buffers. Essentially go back to the state after
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// ao->init().
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void ao_reset(struct ao *ao)
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{
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if (ao->api->reset)
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ao->api->reset(ao);
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atomic_fetch_and(&ao->events_, ~(unsigned int)AO_EVENT_UNDERRUN);
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}
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// Pause playback. Keep the current buffer. ao_get_delay() must return the
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// same value as before pausing.
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void ao_pause(struct ao *ao)
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{
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if (ao->api->pause)
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ao->api->pause(ao);
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}
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// Resume playback. Play the remaining buffer. If the driver doesn't support
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// pausing, it has to work around this and e.g. use ao_play_silence() to fill
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// the lost audio.
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void ao_resume(struct ao *ao)
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{
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if (ao->api->resume)
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ao->api->resume(ao);
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}
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// Block until the current audio buffer has played completely.
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void ao_drain(struct ao *ao)
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{
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if (ao->api->drain)
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ao->api->drain(ao);
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}
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bool ao_eof_reached(struct ao *ao)
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{
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return ao->api->get_eof ? ao->api->get_eof(ao) : true;
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}
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// Query the AO_EVENT_*s as requested by the events parameter, and return them.
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int ao_query_and_reset_events(struct ao *ao, int events)
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{
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return atomic_fetch_and(&ao->events_, ~(unsigned)events) & events;
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}
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void ao_add_events(struct ao *ao, int events)
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{
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atomic_fetch_or(&ao->events_, events);
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ao->wakeup_cb(ao->wakeup_ctx);
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}
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// Request that the player core destroys and recreates the AO. Fully thread-safe.
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void ao_request_reload(struct ao *ao)
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{
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ao_add_events(ao, AO_EVENT_RELOAD);
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}
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// Notify the player that the device list changed. Fully thread-safe.
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void ao_hotplug_event(struct ao *ao)
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{
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ao_add_events(ao, AO_EVENT_HOTPLUG);
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}
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void ao_underrun_event(struct ao *ao)
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{
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// Racy check, but it's just for the message.
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if (!(atomic_load(&ao->events_) & AO_EVENT_UNDERRUN))
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MP_WARN(ao, "Device underrun detected.\n");
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ao_add_events(ao, AO_EVENT_UNDERRUN);
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}
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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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{
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MP_VERBOSE(ao, "Channel layouts:\n");
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mp_chmal_sel_log(s, ao->log, MSGL_V);
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bool r = mp_chmap_sel_adjust(s, map);
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if (r)
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MP_VERBOSE(ao, "result: %s\n", mp_chmap_to_str(map));
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return r;
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}
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|
|
|
// safe_multichannel=true behaves like ao_chmap_sel_adjust.
|
|
// safe_multichannel=false is a helper for callers which do not support safe
|
|
// handling of arbitrary channel layouts. If the multichannel layouts are not
|
|
// considered "always safe" (e.g. HDMI), then allow only stereo or mono, if
|
|
// they are part of the list in *s.
|
|
bool ao_chmap_sel_adjust2(struct ao *ao, const struct mp_chmap_sel *s,
|
|
struct mp_chmap *map, bool safe_multichannel)
|
|
{
|
|
if (!safe_multichannel && (ao->init_flags & AO_INIT_SAFE_MULTICHANNEL_ONLY)) {
|
|
struct mp_chmap res = *map;
|
|
if (mp_chmap_sel_adjust(s, &res)) {
|
|
if (!mp_chmap_equals(&res, &(struct mp_chmap)MP_CHMAP_INIT_MONO) &&
|
|
!mp_chmap_equals(&res, &(struct mp_chmap)MP_CHMAP_INIT_STEREO))
|
|
{
|
|
MP_VERBOSE(ao, "Disabling multichannel output.\n");
|
|
*map = (struct mp_chmap)MP_CHMAP_INIT_STEREO;
|
|
}
|
|
}
|
|
}
|
|
|
|
return ao_chmap_sel_adjust(ao, s, map);
|
|
}
|
|
|
|
bool ao_chmap_sel_get_def(struct ao *ao, const struct mp_chmap_sel *s,
|
|
struct mp_chmap *map, int num)
|
|
{
|
|
return mp_chmap_sel_get_def(s, map, num);
|
|
}
|
|
|
|
// --- The following functions just return immutable information.
|
|
|
|
void ao_get_format(struct ao *ao,
|
|
int *samplerate, int *format, struct mp_chmap *channels)
|
|
{
|
|
*samplerate = ao->samplerate;
|
|
*format = ao->format;
|
|
*channels = ao->channels;
|
|
}
|
|
|
|
const char *ao_get_name(struct ao *ao)
|
|
{
|
|
return ao->driver->name;
|
|
}
|
|
|
|
const char *ao_get_description(struct ao *ao)
|
|
{
|
|
return ao->driver->description;
|
|
}
|
|
|
|
bool ao_get_reports_underruns(struct ao *ao)
|
|
{
|
|
return ao->driver->reports_underruns;
|
|
}
|
|
|
|
bool ao_untimed(struct ao *ao)
|
|
{
|
|
return ao->untimed;
|
|
}
|
|
|
|
// ---
|
|
|
|
struct ao_hotplug {
|
|
struct mpv_global *global;
|
|
void (*wakeup_cb)(void *ctx);
|
|
void *wakeup_ctx;
|
|
// A single AO instance is used to listen to hotplug events. It wouldn't
|
|
// make much sense to allow multiple AO drivers; all sane platforms have
|
|
// a single such audio API.
|
|
// This is _not_ the same AO instance as used for playing audio.
|
|
struct ao *ao;
|
|
// cached
|
|
struct ao_device_list *list;
|
|
bool needs_update;
|
|
};
|
|
|
|
struct ao_hotplug *ao_hotplug_create(struct mpv_global *global,
|
|
void (*wakeup_cb)(void *ctx),
|
|
void *wakeup_ctx)
|
|
{
|
|
struct ao_hotplug *hp = talloc_ptrtype(NULL, hp);
|
|
*hp = (struct ao_hotplug){
|
|
.global = global,
|
|
.wakeup_cb = wakeup_cb,
|
|
.wakeup_ctx = wakeup_ctx,
|
|
.needs_update = true,
|
|
};
|
|
return hp;
|
|
}
|
|
|
|
static void get_devices(struct ao *ao, struct ao_device_list *list)
|
|
{
|
|
if (ao->driver->list_devs) {
|
|
ao->driver->list_devs(ao, list);
|
|
} else {
|
|
ao_device_list_add(list, ao, &(struct ao_device_desc){"", ""});
|
|
}
|
|
}
|
|
|
|
bool ao_hotplug_check_update(struct ao_hotplug *hp)
|
|
{
|
|
if (hp->ao && ao_query_and_reset_events(hp->ao, AO_EVENT_HOTPLUG)) {
|
|
hp->needs_update = true;
|
|
return true;
|
|
}
|
|
return false;
|
|
}
|
|
|
|
// The return value is valid until the next call to this API.
|
|
struct ao_device_list *ao_hotplug_get_device_list(struct ao_hotplug *hp)
|
|
{
|
|
if (hp->list && !hp->needs_update)
|
|
return hp->list;
|
|
|
|
talloc_free(hp->list);
|
|
struct ao_device_list *list = talloc_zero(hp, struct ao_device_list);
|
|
hp->list = list;
|
|
|
|
MP_TARRAY_APPEND(list, list->devices, list->num_devices,
|
|
(struct ao_device_desc){"auto", "Autoselect device"});
|
|
|
|
for (int n = 0; audio_out_drivers[n]; n++) {
|
|
const struct ao_driver *d = audio_out_drivers[n];
|
|
if (d == &audio_out_null)
|
|
break; // don't add unsafe/special entries
|
|
|
|
struct ao *ao = ao_alloc(true, hp->global, hp->wakeup_cb, hp->wakeup_ctx,
|
|
(char *)d->name);
|
|
if (!ao)
|
|
continue;
|
|
|
|
if (ao->driver->hotplug_init) {
|
|
if (!hp->ao && ao->driver->hotplug_init(ao) >= 0)
|
|
hp->ao = ao; // keep this one
|
|
if (hp->ao && hp->ao->driver == d)
|
|
get_devices(hp->ao, list);
|
|
} else {
|
|
get_devices(ao, list);
|
|
}
|
|
if (ao != hp->ao)
|
|
talloc_free(ao);
|
|
}
|
|
hp->needs_update = false;
|
|
return list;
|
|
}
|
|
|
|
void ao_device_list_add(struct ao_device_list *list, struct ao *ao,
|
|
struct ao_device_desc *e)
|
|
{
|
|
struct ao_device_desc c = *e;
|
|
const char *dname = ao->driver->name;
|
|
char buf[80];
|
|
if (!c.desc || !c.desc[0]) {
|
|
if (c.name && c.name[0]) {
|
|
c.desc = c.name;
|
|
} else if (list->num_devices) {
|
|
// Assume this is the default device.
|
|
snprintf(buf, sizeof(buf), "Default (%s)", dname);
|
|
c.desc = buf;
|
|
} else {
|
|
// First default device (and maybe the only one).
|
|
c.desc = "Default";
|
|
}
|
|
}
|
|
c.name = (c.name && c.name[0]) ? talloc_asprintf(list, "%s/%s", dname, c.name)
|
|
: talloc_strdup(list, dname);
|
|
c.desc = talloc_strdup(list, c.desc);
|
|
MP_TARRAY_APPEND(list, list->devices, list->num_devices, c);
|
|
}
|
|
|
|
void ao_hotplug_destroy(struct ao_hotplug *hp)
|
|
{
|
|
if (!hp)
|
|
return;
|
|
if (hp->ao && hp->ao->driver->hotplug_uninit)
|
|
hp->ao->driver->hotplug_uninit(hp->ao);
|
|
talloc_free(hp->ao);
|
|
talloc_free(hp);
|
|
}
|
|
|
|
static void dummy_wakeup(void *ctx)
|
|
{
|
|
}
|
|
|
|
void ao_print_devices(struct mpv_global *global, struct mp_log *log)
|
|
{
|
|
struct ao_hotplug *hp = ao_hotplug_create(global, dummy_wakeup, NULL);
|
|
struct ao_device_list *list = ao_hotplug_get_device_list(hp);
|
|
mp_info(log, "List of detected audio devices:\n");
|
|
for (int n = 0; n < list->num_devices; n++) {
|
|
struct ao_device_desc *desc = &list->devices[n];
|
|
mp_info(log, " '%s' (%s)\n", desc->name, desc->desc);
|
|
}
|
|
ao_hotplug_destroy(hp);
|
|
}
|
|
|
|
void ao_set_gain(struct ao *ao, float gain)
|
|
{
|
|
atomic_store(&ao->gain, gain);
|
|
}
|
|
|
|
#define MUL_GAIN_i(d, num_samples, gain, low, center, high) \
|
|
for (int n = 0; n < (num_samples); n++) \
|
|
(d)[n] = MPCLAMP( \
|
|
((((int64_t)((d)[n]) - (center)) * (gain) + 128) >> 8) + (center), \
|
|
(low), (high))
|
|
|
|
#define MUL_GAIN_f(d, num_samples, gain) \
|
|
for (int n = 0; n < (num_samples); n++) \
|
|
(d)[n] = MPCLAMP(((d)[n]) * (gain), -1.0, 1.0)
|
|
|
|
static void process_plane(struct ao *ao, void *data, int num_samples)
|
|
{
|
|
float gain = atomic_load_explicit(&ao->gain, memory_order_relaxed);
|
|
int gi = lrint(256.0 * gain);
|
|
if (gi == 256)
|
|
return;
|
|
switch (af_fmt_from_planar(ao->format)) {
|
|
case AF_FORMAT_U8:
|
|
MUL_GAIN_i((uint8_t *)data, num_samples, gi, 0, 128, 255);
|
|
break;
|
|
case AF_FORMAT_S16:
|
|
MUL_GAIN_i((int16_t *)data, num_samples, gi, INT16_MIN, 0, INT16_MAX);
|
|
break;
|
|
case AF_FORMAT_S32:
|
|
MUL_GAIN_i((int32_t *)data, num_samples, gi, INT32_MIN, 0, INT32_MAX);
|
|
break;
|
|
case AF_FORMAT_FLOAT:
|
|
MUL_GAIN_f((float *)data, num_samples, gain);
|
|
break;
|
|
case AF_FORMAT_DOUBLE:
|
|
MUL_GAIN_f((double *)data, num_samples, gain);
|
|
break;
|
|
default:;
|
|
// all other sample formats are simply not supported
|
|
}
|
|
}
|
|
|
|
void ao_post_process_data(struct ao *ao, void **data, int num_samples)
|
|
{
|
|
bool planar = af_fmt_is_planar(ao->format);
|
|
int planes = planar ? ao->channels.num : 1;
|
|
int plane_samples = num_samples * (planar ? 1: ao->channels.num);
|
|
for (int n = 0; n < planes; n++)
|
|
process_plane(ao, data[n], plane_samples);
|
|
}
|
|
|
|
static int get_conv_type(struct ao_convert_fmt *fmt)
|
|
{
|
|
if (af_fmt_to_bytes(fmt->src_fmt) * 8 == fmt->dst_bits && !fmt->pad_msb)
|
|
return 0; // passthrough
|
|
if (fmt->src_fmt == AF_FORMAT_S32 && fmt->dst_bits == 24 && !fmt->pad_msb)
|
|
return 1; // simple 32->24 bit conversion
|
|
if (fmt->src_fmt == AF_FORMAT_S32 && fmt->dst_bits == 32 && fmt->pad_msb == 8)
|
|
return 2; // simple 32->24 bit conversion, with MSB padding
|
|
return -1; // unsupported
|
|
}
|
|
|
|
// Check whether ao_convert_inplace() can be called. As an exception, the
|
|
// planar-ness of the sample format and the number of channels is ignored.
|
|
// All other parameters must be as passed to ao_convert_inplace().
|
|
bool ao_can_convert_inplace(struct ao_convert_fmt *fmt)
|
|
{
|
|
return get_conv_type(fmt) >= 0;
|
|
}
|
|
|
|
bool ao_need_conversion(struct ao_convert_fmt *fmt)
|
|
{
|
|
return get_conv_type(fmt) != 0;
|
|
}
|
|
|
|
// The LSB is always ignored.
|
|
#if BYTE_ORDER == BIG_ENDIAN
|
|
#define SHIFT24(x) ((3-(x))*8)
|
|
#else
|
|
#define SHIFT24(x) (((x)+1)*8)
|
|
#endif
|
|
|
|
static void convert_plane(int type, void *data, int num_samples)
|
|
{
|
|
switch (type) {
|
|
case 0:
|
|
break;
|
|
case 1: /* fall through */
|
|
case 2: {
|
|
int bytes = type == 1 ? 3 : 4;
|
|
for (int s = 0; s < num_samples; s++) {
|
|
uint32_t val = *((uint32_t *)data + s);
|
|
uint8_t *ptr = (uint8_t *)data + s * bytes;
|
|
ptr[0] = val >> SHIFT24(0);
|
|
ptr[1] = val >> SHIFT24(1);
|
|
ptr[2] = val >> SHIFT24(2);
|
|
if (type == 2)
|
|
ptr[3] = 0;
|
|
}
|
|
break;
|
|
}
|
|
default:
|
|
abort();
|
|
}
|
|
}
|
|
|
|
// data[n] contains the pointer to the first sample of the n-th plane, in the
|
|
// format implied by fmt->src_fmt. src_fmt also controls whether the data is
|
|
// all in one plane, or if there is a plane per channel.
|
|
void ao_convert_inplace(struct ao_convert_fmt *fmt, void **data, int num_samples)
|
|
{
|
|
int type = get_conv_type(fmt);
|
|
bool planar = af_fmt_is_planar(fmt->src_fmt);
|
|
int planes = planar ? fmt->channels : 1;
|
|
int plane_samples = num_samples * (planar ? 1: fmt->channels);
|
|
for (int n = 0; n < planes; n++)
|
|
convert_plane(type, data[n], plane_samples);
|
|
}
|