mirror of https://github.com/mpv-player/mpv
232 lines
6.1 KiB
C
232 lines
6.1 KiB
C
/*
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* null audio output driver
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*
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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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/*
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* Note: this does much more than just ignoring audio output. It simulates
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* (to some degree) an ideal AO.
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*/
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#include <stdio.h>
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#include <stdlib.h>
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#include <math.h>
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#include "mpv_talloc.h"
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#include "config.h"
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#include "osdep/timer.h"
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#include "options/m_option.h"
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#include "common/common.h"
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#include "common/msg.h"
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#include "audio/format.h"
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#include "ao.h"
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#include "internal.h"
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struct priv {
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bool paused;
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double last_time;
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float buffered; // samples
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int buffersize; // samples
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bool playing;
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int untimed;
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float bufferlen; // seconds
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float speed; // multiplier
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float latency_sec; // seconds
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float latency; // samples
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int broken_eof;
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int broken_delay;
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// Minimal unit of audio samples that can be written at once. If play() is
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// called with sizes not aligned to this, a rounded size will be returned.
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// (This is not needed by the AO API, but many AOs behave this way.)
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int outburst; // samples
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struct m_channels channel_layouts;
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int format;
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};
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static void drain(struct ao *ao)
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{
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struct priv *priv = ao->priv;
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if (ao->untimed) {
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priv->buffered = 0;
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return;
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}
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if (priv->paused)
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return;
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double now = mp_time_sec();
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if (priv->buffered > 0) {
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priv->buffered -= (now - priv->last_time) * ao->samplerate * priv->speed;
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if (priv->buffered < 0)
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priv->buffered = 0;
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}
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priv->last_time = now;
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}
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static int init(struct ao *ao)
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{
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struct priv *priv = ao->priv;
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if (priv->format)
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ao->format = priv->format;
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ao->untimed = priv->untimed;
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struct mp_chmap_sel sel = {.tmp = ao};
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if (priv->channel_layouts.num_chmaps) {
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for (int n = 0; n < priv->channel_layouts.num_chmaps; n++)
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mp_chmap_sel_add_map(&sel, &priv->channel_layouts.chmaps[n]);
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} else {
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mp_chmap_sel_add_any(&sel);
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}
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if (!ao_chmap_sel_adjust(ao, &sel, &ao->channels))
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mp_chmap_from_channels(&ao->channels, 2);
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priv->latency = priv->latency_sec * ao->samplerate;
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// A "buffer" for this many seconds of audio
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int bursts = (int)(ao->samplerate * priv->bufferlen + 1) / priv->outburst;
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ao->device_buffer = priv->outburst * bursts + priv->latency;
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priv->last_time = mp_time_sec();
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return 0;
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}
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// close audio device
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static void uninit(struct ao *ao)
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{
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}
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// stop playing and empty buffers (for seeking/pause)
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static void reset(struct ao *ao)
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{
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struct priv *priv = ao->priv;
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priv->buffered = 0;
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priv->playing = false;
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}
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static void start(struct ao *ao)
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{
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struct priv *priv = ao->priv;
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if (priv->paused)
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MP_ERR(ao, "illegal state: start() while paused\n");
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drain(ao);
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priv->paused = false;
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priv->last_time = mp_time_sec();
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priv->playing = true;
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}
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static bool set_pause(struct ao *ao, bool paused)
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{
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struct priv *priv = ao->priv;
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if (!priv->playing)
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MP_ERR(ao, "illegal state: set_pause() while not playing\n");
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if (priv->paused != paused) {
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drain(ao);
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priv->paused = paused;
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if (!priv->paused)
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priv->last_time = mp_time_sec();
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}
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return true;
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}
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static bool audio_write(struct ao *ao, void **data, int samples)
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{
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struct priv *priv = ao->priv;
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if (priv->buffered <= 0)
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priv->buffered = priv->latency; // emulate fixed latency
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priv->buffered += samples;
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return true;
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}
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static void get_state(struct ao *ao, struct mp_pcm_state *state)
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{
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struct priv *priv = ao->priv;
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drain(ao);
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state->free_samples = ao->device_buffer - priv->latency - priv->buffered;
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state->free_samples = state->free_samples / priv->outburst * priv->outburst;
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state->queued_samples = priv->buffered;
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// Note how get_state returns the delay in audio device time (instead of
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// adjusting for speed), since most AOs seem to also do that.
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state->delay = priv->buffered;
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// Drivers with broken EOF handling usually always report the same device-
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// level delay that is additional to the buffer time.
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if (priv->broken_eof && priv->buffered < priv->latency)
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state->delay = priv->latency;
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state->delay /= ao->samplerate;
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if (priv->broken_delay) { // Report only multiples of outburst
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double q = priv->outburst / (double)ao->samplerate;
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if (state->delay > 0)
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state->delay = (int)(state->delay / q) * q;
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}
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state->playing = priv->playing && priv->buffered > 0;
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}
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#define OPT_BASE_STRUCT struct priv
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const struct ao_driver audio_out_null = {
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.description = "Null audio output",
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.name = "null",
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.init = init,
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.uninit = uninit,
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.reset = reset,
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.get_state = get_state,
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.set_pause = set_pause,
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.write = audio_write,
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.start = start,
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.priv_size = sizeof(struct priv),
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.priv_defaults = &(const struct priv) {
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.bufferlen = 0.2,
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.outburst = 256,
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.speed = 1,
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},
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.options = (const struct m_option[]) {
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{"untimed", OPT_FLAG(untimed)},
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{"buffer", OPT_FLOAT(bufferlen), M_RANGE(0, 100)},
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{"outburst", OPT_INT(outburst), M_RANGE(1, 100000)},
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{"speed", OPT_FLOAT(speed), M_RANGE(0, 10000)},
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{"latency", OPT_FLOAT(latency_sec), M_RANGE(0, 100)},
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{"broken-eof", OPT_FLAG(broken_eof)},
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{"broken-delay", OPT_FLAG(broken_delay)},
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{"channel-layouts", OPT_CHANNELS(channel_layouts)},
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{"format", OPT_AUDIOFORMAT(format)},
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{0}
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},
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.options_prefix = "ao-null",
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};
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