view libao2/ao_openal.c @ 24589:9118be6575da

demux_audio.c: Fix timestamp handling The code calculated the pts values of audio packets by adding the length of the current packet to the pts of the previous one. The length of the previous packet should be added instead. This broke WAV timestamps near the end of the stream where a short packet occurs. Change the code to store the pts of the next packet instead of the last one. This fixes the WAV timestamps and allows some simplifications. MP3 timestamps are not affected as packets are always treated as constant decoded length, and FLAC timestamps still have worse problems (FLAC is treated as as if it was constant bitrate even though it isn't). Also store the timestamps as double instead of float.

/* 
 * 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 ([e-mail])
 */

#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 <[e-mail]>",
  ""
};

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;
}