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mpv/libao2/ao_openal.c

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/*
* ao_openal.c - OpenAL audio output driver for MPlayer
*
* This driver is under the same license as MPlayer.
* (http://www.mplayerhq.hu)
*
* Copyleft 2006 by Reimar Döffinger (Reimar.Doeffinger@stud.uni-karlsruhe.de)
*/
#include "config.h"
#include <stdlib.h>
#include <stdio.h>
#include <inttypes.h>
#ifdef OPENAL_AL_H
#include <OpenAL/alc.h>
#include <OpenAL/al.h>
#else
#include <AL/alc.h>
#include <AL/al.h>
#endif
#include "mp_msg.h"
#include "help_mp.h"
#include "audio_out.h"
#include "audio_out_internal.h"
#include "libaf/af_format.h"
#include "osdep/timer.h"
#include "subopt-helper.h"
static ao_info_t info =
{
"OpenAL audio output",
"openal",
"Reimar Döffinger <Reimar.Doeffinger@stud.uni-karlsruhe.de>",
""
};
LIBAO_EXTERN(openal)
#define MAX_CHANS 6
#define NUM_BUF 128
#define CHUNK_SIZE 512
static ALuint buffers[MAX_CHANS][NUM_BUF];
static ALuint sources[MAX_CHANS];
static int cur_buf[MAX_CHANS];
static int unqueue_buf[MAX_CHANS];
static int16_t *tmpbuf;
static int control(int cmd, void *arg) {
switch (cmd) {
case AOCONTROL_GET_VOLUME:
case AOCONTROL_SET_VOLUME: {
ALfloat volume;
ao_control_vol_t *vol = (ao_control_vol_t *)arg;
if (cmd == AOCONTROL_SET_VOLUME) {
volume = (vol->left + vol->right) / 200.0;
alListenerf(AL_GAIN, volume);
}
alGetListenerf(AL_GAIN, &volume);
vol->left = vol->right = volume * 100;
return CONTROL_TRUE;
}
}
return CONTROL_UNKNOWN;
}
/**
* \brief print suboption usage help
*/
static void print_help(void) {
mp_msg(MSGT_AO, MSGL_FATAL,
"\n-ao openal commandline help:\n"
"Example: mplayer -ao openal\n"
"\nOptions:\n"
);
}
static int init(int rate, int channels, int format, int flags) {
float position[3] = {0, 0, 0};
float direction[6] = {0, 0, 1, 0, -1, 0};
float sppos[6][3] = {
{-1, 0, 0.5}, {1, 0, 0.5},
{-1, 0, -1}, {1, 0, -1},
{0, 0, 1}, {0, 0, 0.1},
};
ALCdevice *dev = NULL;
ALCcontext *ctx = NULL;
ALCint freq = 0;
ALCint attribs[] = {ALC_FREQUENCY, rate, 0, 0};
int i;
opt_t subopts[] = {
{NULL}
};
if (subopt_parse(ao_subdevice, subopts) != 0) {
print_help();
return 0;
}
if (channels > MAX_CHANS) {
mp_msg(MSGT_AO, MSGL_FATAL, "[OpenAL] Invalid number of channels: %i\n", channels);
goto err_out;
}
dev = alcOpenDevice(NULL);
if (!dev) {
mp_msg(MSGT_AO, MSGL_FATAL, "[OpenAL] could not open device\n");
goto err_out;
}
ctx = alcCreateContext(dev, attribs);
alcMakeContextCurrent(ctx);
alListenerfv(AL_POSITION, position);
alListenerfv(AL_ORIENTATION, direction);
alGenSources(channels, sources);
for (i = 0; i < channels; i++) {
cur_buf[i] = 0;
unqueue_buf[i] = 0;
alGenBuffers(NUM_BUF, buffers[i]);
alSourcefv(sources[i], AL_POSITION, sppos[i]);
alSource3f(sources[i], AL_VELOCITY, 0, 0, 0);
}
if (channels == 1)
alSource3f(sources[0], AL_POSITION, 0, 0, 1);
ao_data.channels = channels;
alcGetIntegerv(dev, ALC_FREQUENCY, 1, &freq);
if (alcGetError(dev) == ALC_NO_ERROR && freq)
rate = freq;
ao_data.samplerate = rate;
ao_data.format = AF_FORMAT_S16_NE;
ao_data.bps = channels * rate * 2;
ao_data.buffersize = CHUNK_SIZE * NUM_BUF;
ao_data.outburst = channels * CHUNK_SIZE;
tmpbuf = malloc(CHUNK_SIZE);
return 1;
err_out:
return 0;
}
// close audio device
static void uninit(int immed) {
ALCcontext *ctx = alcGetCurrentContext();
ALCdevice *dev = alcGetContextsDevice(ctx);
free(tmpbuf);
if (!immed) {
ALint state;
alGetSourcei(sources[0], AL_SOURCE_STATE, &state);
while (state == AL_PLAYING) {
usec_sleep(10000);
alGetSourcei(sources[0], AL_SOURCE_STATE, &state);
}
}
reset();
alcMakeContextCurrent(NULL);
alcDestroyContext(ctx);
alcCloseDevice(dev);
}
static void unqueue_buffers(void) {
ALint p;
int s, i;
for (s = 0; s < ao_data.channels; s++) {
alGetSourcei(sources[s], AL_BUFFERS_PROCESSED, &p);
for (i = 0; i < p; i++) {
alSourceUnqueueBuffers(sources[s], 1, &buffers[s][unqueue_buf[s]]);
unqueue_buf[s] = (unqueue_buf[s] + 1) % NUM_BUF;
}
}
}
/**
* \brief stop playing and empty buffers (for seeking/pause)
*/
static void reset(void) {
alSourceRewindv(ao_data.channels, sources);
unqueue_buffers();
}
/**
* \brief stop playing, keep buffers (for pause)
*/
static void audio_pause(void) {
alSourcePausev(ao_data.channels, sources);
}
/**
* \brief resume playing, after audio_pause()
*/
static void audio_resume(void) {
alSourcePlayv(ao_data.channels, sources);
}
static int get_space(void) {
ALint queued;
unqueue_buffers();
alGetSourcei(sources[0], AL_BUFFERS_QUEUED, &queued);
return (NUM_BUF - queued) * CHUNK_SIZE * ao_data.channels;
}
/**
* \brief write data into buffer and reset underrun flag
*/
static int play(void *data, int len, int flags) {
ALint state;
int i, j, k;
int ch;
int16_t *d = data;
len /= ao_data.outburst;
for (i = 0; i < len; i++) {
for (ch = 0; ch < ao_data.channels; ch++) {
for (j = 0, k = ch; j < CHUNK_SIZE / 2; j++, k += ao_data.channels)
tmpbuf[j] = d[k];
alBufferData(buffers[ch][cur_buf[ch]], AL_FORMAT_MONO16, tmpbuf,
CHUNK_SIZE, ao_data.samplerate);
alSourceQueueBuffers(sources[ch], 1, &buffers[ch][cur_buf[ch]]);
cur_buf[ch] = (cur_buf[ch] + 1) % NUM_BUF;
}
d += ao_data.channels * CHUNK_SIZE / 2;
}
alGetSourcei(sources[0], AL_SOURCE_STATE, &state);
if (state != AL_PLAYING) // checked here in case of an underrun
alSourcePlayv(ao_data.channels, sources);
return len * ao_data.outburst;
}
static float get_delay(void) {
ALint queued;
unqueue_buffers();
alGetSourcei(sources[0], AL_BUFFERS_QUEUED, &queued);
return queued * CHUNK_SIZE / 2 / (float)ao_data.samplerate;
}