mirror of
https://github.com/mpv-player/mpv
synced 2025-01-11 17:39:38 +00:00
87dad2a470
MPlayer volume control was originally implemented with the assumption that it controls a system-wide volume setting which keeps its value even if a process closes and reopens the audio device. However, this is not actually true for --softvol mode or some audio output APIs that only consider volume as a per-client setting for software mixing. This could have annoying results, as the volume would be reset to a default value if the AO was closed and reopened, for example whem moving to a new file or crossing ordered chapter boundaries. Add code to set the previous volume again after audio reinitialization if the current audio chain is known to behave this way (softvol active or the AO driver is known to not keep persistent volume externally). This also avoids an inconsistency with the mute flag. The frontend assumed the mute status is persistent across file changes, but it could be similarly lost. The audio drivers that are assumed to not keep persistent volume are: coreaudio, dsound, esd, nas, openal, sdl. None of these changes have been tested. I'm guessing that ESD and NAS do per-connection non-persistent volume settings. Partially based on code by wm4.
255 lines
6.7 KiB
C
255 lines
6.7 KiB
C
/*
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* OpenAL audio output driver for MPlayer
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*
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* Copyleft 2006 by Reimar Döffinger (Reimar.Doeffinger@stud.uni-karlsruhe.de)
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*
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* This file is part of MPlayer.
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*
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* MPlayer is free software; you can redistribute it and/or modify
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* it under the terms of the GNU General Public License as published by
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* the Free Software Foundation; either version 2 of the License, or
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* (at your option) any later version.
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*
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* MPlayer 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 General Public License for more details.
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*
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* You should have received a copy of the GNU General Public License along
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* along with MPlayer; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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#include "config.h"
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#include <stdlib.h>
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#include <stdio.h>
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#include <inttypes.h>
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#ifdef OPENAL_AL_H
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#include <OpenAL/alc.h>
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#include <OpenAL/al.h>
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#else
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#include <AL/alc.h>
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#include <AL/al.h>
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#endif
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#include "mp_msg.h"
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#include "audio_out.h"
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#include "audio_out_internal.h"
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#include "libaf/af_format.h"
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#include "osdep/timer.h"
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#include "subopt-helper.h"
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static const ao_info_t info =
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{
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"OpenAL audio output",
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"openal",
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"Reimar Döffinger <Reimar.Doeffinger@stud.uni-karlsruhe.de>",
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""
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};
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LIBAO_EXTERN(openal)
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#define MAX_CHANS 8
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#define NUM_BUF 128
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#define CHUNK_SIZE 512
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static ALuint buffers[MAX_CHANS][NUM_BUF];
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static ALuint sources[MAX_CHANS];
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static int cur_buf[MAX_CHANS];
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static int unqueue_buf[MAX_CHANS];
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static int16_t *tmpbuf;
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static int control(int cmd, void *arg) {
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switch (cmd) {
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case AOCONTROL_GET_VOLUME:
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case AOCONTROL_SET_VOLUME: {
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ALfloat volume;
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ao_control_vol_t *vol = (ao_control_vol_t *)arg;
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if (cmd == AOCONTROL_SET_VOLUME) {
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volume = (vol->left + vol->right) / 200.0;
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alListenerf(AL_GAIN, volume);
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}
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alGetListenerf(AL_GAIN, &volume);
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vol->left = vol->right = volume * 100;
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return CONTROL_TRUE;
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}
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}
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return CONTROL_UNKNOWN;
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}
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/**
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* \brief print suboption usage help
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*/
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static void print_help(void) {
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mp_msg(MSGT_AO, MSGL_FATAL,
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"\n-ao openal commandline help:\n"
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"Example: mplayer -ao openal\n"
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"\nOptions:\n"
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);
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}
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static int init(int rate, int channels, int format, int flags) {
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float position[3] = {0, 0, 0};
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float direction[6] = {0, 0, 1, 0, -1, 0};
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float sppos[MAX_CHANS][3] = {
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{-1, 0, 0.5}, {1, 0, 0.5},
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{-1, 0, -1}, {1, 0, -1},
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{0, 0, 1}, {0, 0, 0.1},
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{-1, 0, 0}, {1, 0, 0},
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};
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ALCdevice *dev = NULL;
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ALCcontext *ctx = NULL;
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ALCint freq = 0;
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ALCint attribs[] = {ALC_FREQUENCY, rate, 0, 0};
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int i;
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const opt_t subopts[] = {
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{NULL}
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};
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global_ao->no_persistent_volume = true;
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if (subopt_parse(ao_subdevice, subopts) != 0) {
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print_help();
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return 0;
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}
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if (channels > MAX_CHANS) {
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mp_msg(MSGT_AO, MSGL_FATAL, "[OpenAL] Invalid number of channels: %i\n", channels);
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goto err_out;
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}
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dev = alcOpenDevice(NULL);
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if (!dev) {
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mp_msg(MSGT_AO, MSGL_FATAL, "[OpenAL] could not open device\n");
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goto err_out;
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}
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ctx = alcCreateContext(dev, attribs);
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alcMakeContextCurrent(ctx);
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alListenerfv(AL_POSITION, position);
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alListenerfv(AL_ORIENTATION, direction);
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alGenSources(channels, sources);
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for (i = 0; i < channels; i++) {
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cur_buf[i] = 0;
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unqueue_buf[i] = 0;
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alGenBuffers(NUM_BUF, buffers[i]);
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alSourcefv(sources[i], AL_POSITION, sppos[i]);
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alSource3f(sources[i], AL_VELOCITY, 0, 0, 0);
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}
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if (channels == 1)
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alSource3f(sources[0], AL_POSITION, 0, 0, 1);
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ao_data.channels = channels;
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alcGetIntegerv(dev, ALC_FREQUENCY, 1, &freq);
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if (alcGetError(dev) == ALC_NO_ERROR && freq)
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rate = freq;
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ao_data.samplerate = rate;
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ao_data.format = AF_FORMAT_S16_NE;
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ao_data.bps = channels * rate * 2;
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ao_data.buffersize = CHUNK_SIZE * NUM_BUF;
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ao_data.outburst = channels * CHUNK_SIZE;
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tmpbuf = malloc(CHUNK_SIZE);
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return 1;
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err_out:
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return 0;
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}
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// close audio device
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static void uninit(int immed) {
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ALCcontext *ctx = alcGetCurrentContext();
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ALCdevice *dev = alcGetContextsDevice(ctx);
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free(tmpbuf);
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if (!immed) {
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ALint state;
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alGetSourcei(sources[0], AL_SOURCE_STATE, &state);
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while (state == AL_PLAYING) {
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usec_sleep(10000);
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alGetSourcei(sources[0], AL_SOURCE_STATE, &state);
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}
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}
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reset();
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alcMakeContextCurrent(NULL);
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alcDestroyContext(ctx);
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alcCloseDevice(dev);
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}
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static void unqueue_buffers(void) {
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ALint p;
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int s;
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for (s = 0; s < ao_data.channels; s++) {
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int till_wrap = NUM_BUF - unqueue_buf[s];
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alGetSourcei(sources[s], AL_BUFFERS_PROCESSED, &p);
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if (p >= till_wrap) {
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alSourceUnqueueBuffers(sources[s], till_wrap, &buffers[s][unqueue_buf[s]]);
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unqueue_buf[s] = 0;
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p -= till_wrap;
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}
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if (p) {
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alSourceUnqueueBuffers(sources[s], p, &buffers[s][unqueue_buf[s]]);
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unqueue_buf[s] += p;
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}
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}
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}
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/**
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* \brief stop playing and empty buffers (for seeking/pause)
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*/
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static void reset(void) {
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alSourceStopv(ao_data.channels, sources);
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unqueue_buffers();
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}
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/**
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* \brief stop playing, keep buffers (for pause)
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*/
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static void audio_pause(void) {
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alSourcePausev(ao_data.channels, sources);
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}
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/**
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* \brief resume playing, after audio_pause()
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*/
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static void audio_resume(void) {
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alSourcePlayv(ao_data.channels, sources);
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}
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static int get_space(void) {
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ALint queued;
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unqueue_buffers();
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alGetSourcei(sources[0], AL_BUFFERS_QUEUED, &queued);
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queued = NUM_BUF - queued - 3;
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if (queued < 0) return 0;
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return queued * CHUNK_SIZE * ao_data.channels;
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}
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/**
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* \brief write data into buffer and reset underrun flag
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*/
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static int play(void *data, int len, int flags) {
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ALint state;
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int i, j, k;
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int ch;
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int16_t *d = data;
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len /= ao_data.channels * CHUNK_SIZE;
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for (i = 0; i < len; i++) {
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for (ch = 0; ch < ao_data.channels; ch++) {
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for (j = 0, k = ch; j < CHUNK_SIZE / 2; j++, k += ao_data.channels)
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tmpbuf[j] = d[k];
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alBufferData(buffers[ch][cur_buf[ch]], AL_FORMAT_MONO16, tmpbuf,
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CHUNK_SIZE, ao_data.samplerate);
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alSourceQueueBuffers(sources[ch], 1, &buffers[ch][cur_buf[ch]]);
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cur_buf[ch] = (cur_buf[ch] + 1) % NUM_BUF;
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}
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d += ao_data.channels * CHUNK_SIZE / 2;
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}
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alGetSourcei(sources[0], AL_SOURCE_STATE, &state);
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if (state != AL_PLAYING) // checked here in case of an underrun
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alSourcePlayv(ao_data.channels, sources);
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return len * ao_data.channels * CHUNK_SIZE;
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}
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static float get_delay(void) {
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ALint queued;
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unqueue_buffers();
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alGetSourcei(sources[0], AL_BUFFERS_QUEUED, &queued);
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return queued * CHUNK_SIZE / 2 / (float)ao_data.samplerate;
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}
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