view libao2/ao_openal.c @ 28992:947ef23ba798

Test if create_vdp_decoder() might succeed by calling it from config() with a small value for max_reference_frames. This does not make automatic recovery by using software decoder possible, but lets MPlayer fail more graciously on - actually existing - buggy hardware that does not support certain H264 widths when using hardware accelerated decoding (784, 864, 944, 1024, 1808, 1888 pixels on NVIDIA G98) and if the user tries to hardware-decode more samples at the same time than supported. Might break playback of H264 Intra-Only samples on hardware with very little video memory.
author cehoyos
date Sat, 21 Mar 2009 20:11:05 +0000
parents 9a5b8c2ed6de
children 0f1b5b68af32
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 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;
  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;
}