Slightly improve audio debug messages

git-svn-id: svn://cook.msm.cam.ac.uk:745/thrust3d/thrust3d@123 84d2e878-0bd5-11dd-ad15-13eda11d74c5

1 files changed, 11 insertions, 11 deletions

diff --git a/src/audio.c b/src/audio.c
index 82d70f9..2690c61 100644
--- a/src/audio.c
+++ b/src/audio.c
@@ -63,12 +63,12 @@ static void audio_mix(void *data, Uint8 *stream8, int len) {
 			if ( a->sounds[i].dpos == a->sounds[i].dlen ) {
 				if ( a->sounds[i].repeat ) {
 					a->sounds[i].dpos = 0;
-					if ( a->debug ) printf("AU: Looping sound %i at buffer offset %i/%i\n", i, j, len);
+					if ( a->debug ) printf("AU: Channel %i: Looping at buffer offset %i/%i\n", i, j, len);
 				} else {
 					a->sounds[i].inuse = 0;
 					a->sounds[i].playing = 0;
 					free(a->sounds[i].data);
-					if ( a->debug ) printf("AU: End of sound %i\n", i);
+					if ( a->debug ) printf("AU: Channel %i: End of data\n", i);
 					break;
 				}
 			}
@@ -112,11 +112,11 @@ static void audio_play_wav(AudioContext *a, char *filename, float volume, int re
 		return;
 	}
 	a->sounds[idx].inuse = 1;
-	if ( a->debug ) printf("AU: Selected channel %i for sound '%s'\n", idx, filename);
+	if ( a->debug ) printf("AU: Channel %i: Selected this channel for sound '%s'\n", idx, filename);
 	
 	/* Load the sound file and convert it to 16-bit stereo at 44.1 kHz */
 	if ( SDL_LoadWAV(filename, &wave, &data, &dlen) == NULL ) {
-		fprintf(stderr, "Couldn't load %s: %s\n", filename, SDL_GetError());
+		fprintf(stderr, "Couldn't load %s: (channel %i) %s\n", filename, idx, SDL_GetError());
 		return;
 	}
 	SDL_BuildAudioCVT(&cvt, wave.format, wave.channels, wave.freq, AUDIO_S16, 2, 44100);
@@ -162,30 +162,30 @@ static void *audio_play_vorbis(void *add_void) {
 		return NULL;
 	}
 	a->sounds[idx].inuse = 1;
-	if ( a->debug ) printf("AU: Selected channel %i for sound '%s'\n", idx, filename);
+	if ( a->debug ) printf("AU: Channel %i: Selected this channel for sound '%s'\n", idx, filename);
 	
 	err = ov_fopen(filename, &vf);
 	if ( err != 0 ) {
-		fprintf(stderr, "Couldn't open Vorbis file '%s' (code %i,%i)\n", filename, err, errno);
+		fprintf(stderr, "Couldn't open Vorbis file '%s' (channel %i, code %i,%i)\n", filename, idx, err, errno);
 		return NULL;
 	}
 	
 	len = ov_pcm_total(&vf, -1);	/* Length in samples 'per channel' */
-	if ( a->debug ) printf("AU: Length is %i samples 'per channel'\n", len);
+	if ( a->debug ) printf("AU: Channel %i: Length is %i samples 'per channel'\n", idx, len);
 	vi = ov_info(&vf,-1);
-	if ( a->debug ) printf("AU: %i channels, %li Hz\n", vi->channels, vi->rate);
+	if ( a->debug ) printf("AU: Channel %i: %i channels, %li Hz\n", idx, vi->channels, vi->rate);
 	data = malloc(vi->channels*2*len);	/* Two bytes per sample per channel */
 	
 	offs = 0;
 	finished = 0;
-	if ( a->debug ) printf("AU: Decoding Vorbis stream...\n");
+	if ( a->debug ) printf("AU: Channel %i: Decoding Vorbis stream...\n", idx);
 	while ( finished == 0 ) {
 		long rval;
 		rval = ov_read(&vf, data+offs, 2*2*len, 0, 2, 1, &current_section);
 		if ( rval == 0 ) {
 			finished = 1;
 		} else if ( rval < 0 ) {
-			fprintf(stderr, "Vorbis stream error\n");
+			fprintf(stderr, "Vorbis stream error (channel %i)\n", idx);
 		} else {
 			offs += rval;
 		}
@@ -212,7 +212,7 @@ static void *audio_play_vorbis(void *add_void) {
 	
 	free(add->filename);
 	free(add);
-	if ( a->debug ) printf("AU: audio_play_vorbis dispatch thread completed.\n");
+	if ( a->debug ) printf("AU: Channel %i: audio_play_vorbis dispatch thread completed.\n", idx);
 	return NULL;
 	
 }