Going back to calling libvorbisfile directly because libsndfile doesn't

support reading OGGV embedded in other files.  OGGV playback sort of
works (slow and distorted).  Turning it off isn't exactly right.

diff --git a/src/curses/ux_audio.c b/src/curses/ux_audio.c
index f8b505a2f38d5dd27f5169a9b135154698cf3b10..f9315c55fda10cac4d8fe5514974e7b1c28a69e6 100644 (file)
--- a/src/curses/ux_audio.c
+++ b/src/curses/ux_audio.c
@@ -67,6 +67,7 @@ static void *playaiff(EFFECT *);
 static void *playmusic(EFFECT *);
 static void *playmod(EFFECT *);
 static void *playogg(EFFECT *);
+
 static void floattopcm16(short *, float *, int);
 static void pcm16tofloat(float *, short *, int);
 static void stereoize(float *, float *, size_t);
@@ -590,18 +591,15 @@ static void *playmusic(EFFECT *raw_effect)
 
     sem_post(&playmusic_okay);
 
-    if (myeffect.type == MOD) {
-       playmod(&myeffect);
-       printf("  mod done\n\r");
-    } else if (myeffect.type == OGGV) {
-       playogg(&myeffect);
-       printf("  ogg done\n\r");
-    } else;
+    if (myeffect.type == MOD)          playmod(&myeffect);
+    else if (myeffect.type == OGGV)    playogg(&myeffect);
+    else;
 
     pthread_exit(NULL);
 
 } /* playmusic */
 
+
 /*
  * playmod
  *
@@ -721,128 +719,90 @@ static char *getfiledata(FILE *fp, long *size)
  * sure that an OGG chunk is to be played.  Volume and repeats are also
  * handled here.
  *
- * This function invoked libvorbisfile directly rather than going
- * through libsndfile.  The reason for this is that libsndfile refuses
- * to load an OGG file that's embedded in a Blorb file.
+ * Libsndfile is capable of reading OGG files, but not if the file is
+ * embedded in another file.  That's why we're using libvorbisfile
+ * directly instead of going through libsndfile.  Erikd, main developer
+ * of libsndfile is working on a fix.
  *
  */
-void *playogg(EFFECT *raw_effect)
+static void *playogg(EFFECT *raw_effect)
 {
-    long filestart;
+    ogg_int64_t toread;
+    ogg_int64_t frames_read;
+    ogg_int64_t count;
 
-    int volcount;
-    int volfactor;
+    vorbis_info *info;
+
+    OggVorbis_File vf;
 
+    int current_section;
+    short *shortbuffer;
     float *floatbuffer;
     float *floatbuffer2;
 
-    SNDFILE     *sndfile;
-    SF_INFO     sf_info;
-
-    SRC_STATE  *src_state;
-    SRC_DATA   src_data;
-    int                error;
-    sf_count_t output_count = 0;
+    long filestart;
+    int volcount;
+    int volfactor;
 
     EFFECT myeffect = *raw_effect;
 
-    sf_info.format = 0;
-    musicnum = myeffect.number;
+    fseek(myeffect.fp, myeffect.result.data.startpos, SEEK_SET);
 
-    filestart = ftell(myeffect.fp);
-    lseek(fileno(myeffect.fp), myeffect.result.data.startpos, SEEK_SET);
-    sndfile = sf_open_fd(fileno(myeffect.fp), SFM_READ, &sf_info, 0);
+    if (ov_open_callbacks(myeffect.fp, &vf, NULL, 0, OV_CALLBACKS_NOCLOSE) < 0) {
+       printf("Unable to open OGGV stream.\n\r");
+       return;
+    }
+
+    info = ov_info(&vf, -1);
+    if (info == NULL) {
+       printf("Unable to get info on OGGV stream.\n\r");
+       return;
+    }
 
     if (myeffect.vol < 1) myeffect.vol = 1;
     if (myeffect.vol > 8) myeffect.vol = 8;
     volfactor = mypower(2, -myeffect.vol + 8);
 
-    floatbuffer = malloc(BUFFSIZE * sf_info.channels * sizeof(float));
-    floatbuffer2 = malloc(BUFFSIZE * 2 * sizeof(float));
-    memset(bleepbuffer, 0, BUFFSIZE * sizeof(float) * 2);
+    shortbuffer = malloc(BUFFSIZE * info->channels * sizeof(short));
+    floatbuffer = malloc(BUFFSIZE * info->channels * sizeof(float));
 
-    /* Set up for conversion */
-    if ((src_state = src_new(DEFAULT_CONVERTER, sf_info.channels, &error)) == NULL) {
-       printf("Error: src_new() failed: %s.\n", src_strerror(error));
-       exit(1);
-    }
-    src_data.end_of_input = 0;
-    src_data.input_frames = 0;
-    src_data.data_in = floatbuffer;
-    src_data.src_ratio = (1.0 * SAMPLERATE) / sf_info.samplerate;
-    src_data.data_out = floatbuffer2;
-    src_data.output_frames = BUFFSIZE / sf_info.channels;
+    frames_read = 0;
+    toread = ov_pcm_total(&vf, -1) * 2 * info->channels;
+    count = 0;
 
     music_playing = TRUE;
 
-    while (1) {
-       /* Check if we're being told to stop. */
-       if (!bleep_playing) break;
+    while (count < toread) {
        sem_wait(&audio_empty);
        pthread_mutex_lock(&mutex);
+       memset(musicbuffer, 0, BUFFSIZE * sizeof(float) * 2);
+       if (!music_playing) break;
 
-       /* If floatbuffer is empty, refill it. */
-       if (src_data.input_frames == 0) {
-           src_data.input_frames = sf_readf_float(sndfile, floatbuffer, BUFFSIZE / sf_info.channels);
-           src_data.data_in = floatbuffer;
-           /* Mark end of input. */
-           if (src_data.input_frames < BUFFSIZE / sf_info.channels)
-               src_data.end_of_input = SF_TRUE;
-       }
-
-       /* Do the sample rate conversion. */
-       if ((error = src_process(src_state, &src_data))) {
-           printf("Error: %s\n", src_strerror(error));
-           exit(1);
-       }
-
-       musicsamples = src_data.output_frames_gen * 2;
-
-       /* Stereoize monaural sound-effects. */
-       if (sf_info.channels == 1) {
-           /* Remember that each monaural frame contains just one sample. */
-           stereoize(musicbuffer, floatbuffer2, src_data.output_frames_gen);
-       } else {
-           /* It's already stereo.  Just copy the buffer. */
-           memcpy(musicbuffer, floatbuffer2, sizeof(float) * src_data.output_frames_gen * 2);
-       }
-
-       /* Adjust volume. */
-       for (volcount = 0; volcount <= musicsamples; volcount++)
-           musicbuffer[volcount] /= volfactor;
-
-       /* If that's all, terminate and signal that we're done. */
-       if (src_data.end_of_input && src_data.output_frames_gen == 0) {
-           pthread_mutex_unlock(&mutex);
-           sem_post(&audio_full);
-           break;
-       }
+        frames_read = ov_read(&vf, (char *)shortbuffer, BUFFSIZE, 0,2,1,&current_section);
+        pcm16tofloat(floatbuffer, shortbuffer, frames_read);
+        for (volcount = 0; volcount <= frames_read / 2; volcount++) {
+            ((float *) floatbuffer)[volcount] /= volfactor;
+        }
 
-       /* Get ready for the next chunk. */
-        output_count += src_data.output_frames_gen;
-        src_data.data_in += src_data.input_frames_used * sf_info.channels;
-        src_data.input_frames -= src_data.input_frames_used;
+       musicsamples = frames_read;
+       memcpy(musicbuffer, floatbuffer, musicsamples);
+        count += frames_read;
 
-       /* By this time, the buffer is full.  Signal the mixer to play it. */
        pthread_mutex_unlock(&mutex);
        sem_post(&audio_full);
     }
-
-    /* The two ways to exit the above loop are to process all the
-     * samples in the AIFF file or else get told to stop early.
-     * Whichever, we need to clean up and terminate this thread.
-     */
-
     music_playing = FALSE;
+    memset(musicbuffer, 0, BUFFSIZE * sizeof(float) * 2);
+
     pthread_mutex_unlock(&mutex);
-//    sem_post(&audio_empty);
+    sem_post(&audio_empty);
 
-    fseek(myeffect.fp, filestart, SEEK_SET);
-    sf_close(sndfile);
+    ov_clear(&vf);
+
+    free(shortbuffer);
     free(floatbuffer);
-    free(floatbuffer2);
 
-    pthread_exit(NULL);
-} /* playmod */
+    return;
+} /* playogg */
 
 #endif /* NO_SOUND */