[MPlayer-cvslog] CVS: main/libao2 ao_openal.c, NONE, 1.1 Makefile, 1.35, 1.36 audio_out.c, 1.51, 1.52

Reimar Döffinger CVS syncmail at mplayerhq.hu
Thu Feb 16 21:45:27 CET 2006


CVS change done by Reimar Döffinger CVS

Update of /cvsroot/mplayer/main/libao2
In directory mail:/var2/tmp/cvs-serv4691/libao2

Modified Files:
	Makefile audio_out.c 
Added Files:
	ao_openal.c 
Log Message:
OpenAL audio support, actual output is mono-only (no positioning yet).


--- NEW FILE ---
/* 
 * 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 at stud.uni-karlsruhe.de)
 */

#include <inttypes.h>
#include <AL/alc.h>
#include <AL/al.h>

#include "config.h"
#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 at 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) {
  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) {
  ALCdevice *dev = NULL;
  ALCcontext *ctx = NULL;
  ALint bufrate;
  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, NULL);
  alcMakeContextCurrent(ctx);
  for (i = 0; i < channels; i++) {
    cur_buf[i] = 0;
    unqueue_buf[i] = 0;
    alGenBuffers(NUM_BUF, buffers[i]);
  }
  alGenSources(channels, sources);
  alSource3f(sources[0], AL_POSITION, 0, 0, 10);
  ao_data.channels = channels;
  alGetBufferi(buffers[0][0], AL_FREQUENCY, &bufrate);
  ao_data.samplerate = rate = bufrate;
  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 = (int16_t *)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;
}


Index: Makefile
===================================================================
RCS file: /cvsroot/mplayer/main/libao2/Makefile,v
retrieving revision 1.35
retrieving revision 1.36
diff -u -r1.35 -r1.36
--- Makefile	27 Jan 2006 00:06:38 -0000	1.35
+++ Makefile	16 Feb 2006 20:45:25 -0000	1.36
@@ -17,6 +17,7 @@
           $(ESD_INC) \
           $(EXTRA_INC) \
           $(JACK_INC) \
+          $(OPENAL_INC) \
           $(POLYP_INC) \
           $(SDL_INC) \
           $(X11_INC) \

Index: audio_out.c
===================================================================
RCS file: /cvsroot/mplayer/main/libao2/audio_out.c,v
retrieving revision 1.51
retrieving revision 1.52
diff -u -r1.51 -r1.52
--- audio_out.c	9 Feb 2006 14:07:54 -0000	1.51
+++ audio_out.c	16 Feb 2006 20:45:25 -0000	1.52
@@ -31,6 +31,7 @@
 #ifdef USE_JACK
 extern ao_functions_t audio_out_jack;
 #endif
+extern ao_functions_t audio_out_openal;
 extern ao_functions_t audio_out_null;
 #ifdef HAVE_ALSA5
  extern ao_functions_t audio_out_alsa5;
@@ -120,6 +121,7 @@
 #ifdef HAVE_SDL
         &audio_out_sdl,
 #endif
+        &audio_out_openal,
         &audio_out_null,
 // should not be auto-selected:
 	&audio_out_pcm,




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