view libao2/ao_openal.c @ 34478:8e09f1cb3ecd

Fix vo_gl unsharp filter for chroma. The syntax is a bit strange, since for inputs the components indicate swizzles, while for outputs it is only a write mask, thus the result must be at the correct position regardless of the component specified for the output. So use a 3-component vector for the constant factor. Also make the input swizzles explicit in an attempt to make the code less confusing (that part does change what the code actually does). Previous code would result in a filter strength of 0 always being used for chroma.
author reimar
date Sat, 14 Jan 2012 15:49:54 +0000
parents 32725ca88fed
children
line wrap: on
line source

/*
 * OpenAL audio output driver for MPlayer
 *
 * Copyleft 2006 by Reimar Döffinger (Reimar.Doeffinger@stud.uni-karlsruhe.de)
 *
 * This file is part of MPlayer.
 *
 * MPlayer is free software; you can redistribute it and/or modify
 * it under the terms of the GNU General Public License as published by
 * the Free Software Foundation; either version 2 of the License, or
 * (at your option) any later version.
 *
 * MPlayer is distributed in the hope that it will be useful,
 * but WITHOUT ANY WARRANTY; without even the implied warranty of
 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
 * GNU General Public License for more details.
 *
 * You should have received a copy of the GNU General Public License along
 * along with MPlayer; if not, write to the Free Software
 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
 */

#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 const ao_info_t info =
{
  "OpenAL audio output",
  "openal",
  "Reimar Döffinger <Reimar.Doeffinger@stud.uni-karlsruhe.de>",
  ""
};

LIBAO_EXTERN(openal)

#define MAX_CHANS 8
#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[MAX_CHANS][3] = {
    {-1, 0, 0.5}, {1, 0, 0.5},
    {-1, 0,  -1}, {1, 0,  -1},
    {0,  0,   1}, {0, 0, 0.1},
    {-1, 0,   0}, {1, 0,   0},
  };
  ALCdevice *dev = NULL;
  ALCcontext *ctx = NULL;
  ALCint freq = 0;
  ALCint attribs[] = {ALC_FREQUENCY, rate, 0, 0};
  int i;
  const 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;
  for (s = 0;  s < ao_data.channels; s++) {
    int till_wrap = NUM_BUF - unqueue_buf[s];
    alGetSourcei(sources[s], AL_BUFFERS_PROCESSED, &p);
    if (p >= till_wrap) {
      alSourceUnqueueBuffers(sources[s], till_wrap, &buffers[s][unqueue_buf[s]]);
      unqueue_buf[s] = 0;
      p -= till_wrap;
    }
    if (p) {
      alSourceUnqueueBuffers(sources[s], p, &buffers[s][unqueue_buf[s]]);
      unqueue_buf[s] += p;
    }
  }
}

/**
 * \brief stop playing and empty buffers (for seeking/pause)
 */
static void reset(void) {
  alSourceStopv(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);
  queued = NUM_BUF - queued - 3;
  if (queued < 0) return 0;
  return 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.channels * CHUNK_SIZE;
  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.channels * CHUNK_SIZE;
}

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;
}