Progressive Vorbis decoding

git-svn-id: svn://cook.msm.cam.ac.uk:745/thrust3d/thrust3d@153 84d2e878-0bd5-11dd-ad15-13eda11d74c5

2 files changed, 80 insertions, 40 deletions

diff --git a/src/audio.c b/src/audio.c
index f0c0e4f..bc66d83 100644
--- a/src/audio.c
+++ b/src/audio.c
@@ -21,6 +21,7 @@
 #include <vorbis/vorbisfile.h>
 #include <vorbis/codec.h>
 #include <pthread.h>
+#include <unistd.h>
 
 #include "types.h"
 
@@ -171,7 +172,7 @@ static void *audio_play_vorbis(void *add_void) {
 	int len;
 	OggVorbis_File vf;
 	vorbis_info *vi;
-	int finished, err, current_section;
+	int finished, err, current_section, decode_done, decode_samples_done;
 	size_t offs;
 	SDL_AudioCVT cvt;
 	
@@ -200,55 +201,90 @@ static void *audio_play_vorbis(void *add_void) {
 	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: Channel %i: %i channels, %li Hz\n", idx, vi->channels, vi->rate);
-	data = malloc(vi->channels*2*len);	/* Two bytes per sample per channel */
+	
+	/* Decode ten seconds of audio every 8 seconds until the job's done */
+	if ( a->debug ) printf("AU: Channel %i: Started decoding Vorbis stream\n", idx);
+	/* Do this now because cvt.len_mult is needed to work out the size of 'data' */
+	SDL_BuildAudioCVT(&cvt, AUDIO_S16, vi->channels, vi->rate, AUDIO_S16, 2, 44100);
+	data = malloc(10*vi->rate*2*vi->channels*cvt.len_mult);
 	if ( data == NULL ) {
 		fprintf(stderr, "Not enough memory to decode Vorbis stream (channel %i)\n", idx);
 		a->sounds[idx].inuse = 0;
 		ov_clear(&vf);
 		goto out;
 	}
-	
-	offs = 0;
-	finished = 0;
-	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 (channel %i)\n", idx);
-		} else {
-			offs += rval;
+	decode_done = 0;
+	decode_samples_done = 0;
+	a->sounds[idx].decode_pos = 0;	/* Position (in sounds[idx].data) at which to write the next block */
+	while ( !decode_done ) {
+	
+		long decode_block_samples;
+		long decode_block_length;
+		int w;
+		
+		/* Decide how much data to shovel this time */
+		decode_block_samples = 10*vi->rate;	/* 10 seconds in samples */
+		if ( decode_samples_done + decode_block_samples > len ) {
+			/* This is the last block.  Yay! */
+			decode_block_samples = len - decode_samples_done;
+			decode_done = 1;
 		}
+		if ( a->debug ) printf("AU: Channel %i: Decoding %li samples this time\n", idx, decode_block_samples);
+		decode_block_length = vi->channels*2*decode_block_samples;
+		decode_samples_done += decode_block_samples;
+		
+		/* Read the chosen amount of data */
+		offs = 0;  finished = 0;
+		while ( finished == 0 ) {
+			long rval;
+			rval = ov_read(&vf, data+offs, decode_block_length-offs, 0, 2, 1, &current_section);
+			if ( rval < 0 ) {
+				fprintf(stderr, "Vorbis stream error (channel %i)\n", idx);
+			} else {
+				offs += rval;
+			}
+			if ( offs == decode_block_length ) finished = 1;
+		}
+		
+		/* Convert to 44.1 kHz */
+		cvt.buf = (Uint8 *)data;
+		cvt.len = decode_block_length;
+		SDL_ConvertAudio(&cvt);
+		
+		/* Paste this seamlessly into the playing thread's buffer */
+		if ( !a->sounds[idx].playing ) {
+			a->sounds[idx].data = malloc(len*vi->channels*2);
+		}
+		memcpy(a->sounds[idx].data+a->sounds[idx].decode_pos, cvt.buf, cvt.len_cvt);
+		/* I have ABSOLUTELY NO IDEA why the division by two is needed. Anyone? */
+		a->sounds[idx].decode_pos += cvt.len_cvt / 2;
+		
+		/* It's safe to start playing at this point */
+		if ( !a->sounds[idx].playing ) {
+			a->sounds[idx].dlen = vi->channels*(len-1);
+			a->sounds[idx].dpos = 0;
+			a->sounds[idx].repeat = add->repeat;
+			a->sounds[idx].volume = add->volume;
+			a->sounds[idx].playing = 1;	/* Must be done last - tell the mixer thread it can use this */
+		}
+		
+		/* Sleep for eight seconds while periodically checking for shutdown */
+		for ( w=0; w<80; w++ ) {
+			if ( a->shutdown ) {
+				ov_clear(&vf);
+				free(data);
+				goto out;
+			}
+			usleep(100000);	/* 0.1 seconds */
+		}
+		
 	}
-	
-	/* Convert to 44.1 kHz */
-	SDL_BuildAudioCVT(&cvt, AUDIO_S16, vi->channels, vi->rate, AUDIO_S16, 2, 44100);
-	cvt.buf = malloc(vi->channels*2*len*cvt.len_mult);
-	if ( cvt.buf == NULL ) {
-		fprintf(stderr, "Not enough memory to convert audio (channel %i)\n", idx);
-		free(cvt.buf);
-		a->sounds[idx].inuse = 0;
-		goto out;
-	}
-	memcpy(cvt.buf, data, vi->channels*2*len);
-	cvt.len = vi->channels*2*len;
-	SDL_ConvertAudio(&cvt);
+	if ( a->debug ) printf("AU: Channel %i: Finished decoding the Vorbis stream.\n", idx);
 	free(data);
 	
 	/* Needed until now */
 	ov_clear(&vf);
-	
-	/* Put the sound data in the slot */
-	a->sounds[idx].data = (Sint16 *)cvt.buf;
-	/* Assuming that the same dud last (stereo) sample occurs here. This may not be the case. */
-	a->sounds[idx].dlen = cvt.len_cvt / 2 - 2;
-	a->sounds[idx].dpos = 0;
-	a->sounds[idx].repeat = add->repeat;
-	a->sounds[idx].volume = add->volume;
-	a->sounds[idx].playing = 1;	/* Must be done last - tell the mixer thread it can use this */
-	
+		
 out:
 	free(add->filename);
 	if ( a->debug ) printf("AU: Channel %i: Vorbis dispatch thread completed.\n", idx);
@@ -383,6 +419,7 @@ AudioContext *audio_setup(int debug, int no_music) {
 	a->startup = 1;
 	a->startup_volume = 0.0;
 	a->paused = 1;
+	a->shutdown = 0;
 	pthread_mutex_init(&a->sounds_mutex, NULL);
 	for ( i=0; i<AUDIO_MAX_SOUNDS; i++ ) {
 		a->sounds[i].inuse = 0;
@@ -424,6 +461,7 @@ void audio_shutdown(AudioContext *a) {
 	
 	if ( a == NULL ) return;
 	
+	a->shutdown = 1;
 	SDL_CloseAudio();
 	
 	/* Wait for dispatch threads */
diff --git a/src/types.h b/src/types.h
index bfbe36a..950d26e 100644
--- a/src/types.h
+++ b/src/types.h
@@ -160,12 +160,13 @@ typedef struct {
 
 	struct sound {
 		Sint16 *data;
-		int dpos;
-		int dlen;
+		long dpos;
+		long dlen;
 		int inuse;
 		int playing;
 		int repeat;
 		float volume;
+		size_t decode_pos;
 	} sounds[AUDIO_MAX_SOUNDS];
 	
 	pthread_mutex_t sounds_mutex;			/* Mutex for the list of sounds being played */
@@ -176,6 +177,7 @@ typedef struct {
 	int startup;
 	float startup_volume;
 	int paused;
+	int shutdown;
 	
 } AudioContext;