making slow progress on sound quality

streamed oggs playback cleaning when not resampling, but flacs are still crap

include/snd_render.h

@@ -121,13 +121,13 @@ struct sfxbuffer_s {
 	*/
 	void        (*setpos) (sfxbuffer_t *buffer, unsigned int pos);
 	sfx_t      *sfx;			//!< owning sfx_t instance
-	/** Points to the beginning of the sample data within sample_data.
+	/**			resampler state information
 	*/
-	float      *data;
+	void       *state;
 	/** Sample data. The block at the beginning of the buffer (size depends on
-		sample size) holds the state information for resampling.
+		sample size)
 	*/
-	float       sample_data[1];
+	float       data[1];
 };
 
 /** Representation of sound loaded that is streamed in as needed.
@@ -398,6 +398,12 @@ void SND_SetPaint (sfxbuffer_t *sc);
 	\ingroup sound_render
 */
 //@{
+/** Set up the various parameters that depend on the actual sample rate.
+	\param sc		buffer to setup
+	\param streamed	non-zero if this is for a stream.
+*/
+void SND_SetupResampler (sfxbuffer_t *sc, int streamed);
+
 /** Copy/resample data into buffer, resampling as necessary.
 	\param sc		buffer to write resampled sound
 	\param data		raw sample data

libs/audio/renderer/flac.c

@@ -158,20 +158,21 @@ flac_write_func (const FLAC__StreamDecoder *decoder,
 	flacfile_t *ff = (flacfile_t *) client_data;
 	float      *out;
 	float       scale = 2.0 / (1 << ff->info.bits_per_sample);
+	int         step = ff->info.channels;
 	unsigned    i, j;
 
 	if (!ff->buffer)
-		ff->buffer = malloc (ff->info.max_blocksize * ff->info.channels
+		ff->buffer = calloc (ff->info.max_blocksize * ff->info.channels,
-							 * sizeof (float));
+							 sizeof (float));
+	ff->size = frame->header.blocksize;
+	ff->pos = 0;
 	for (j = 0; j < ff->info.channels; j++) {
 		const FLAC__int32 *in = buffer[j];
 
 		out = ff->buffer + j;
-		for (i = 0; i < frame->header.blocksize; i++, out += ff->info.channels)
+		for (i = 0; i < ff->size; i++, out += step)
 			*out = *in++ * scale;
 	}
-	ff->size = frame->header.blocksize;
-	ff->pos = 0;
 	return FLAC__STREAM_DECODER_WRITE_STATUS_CONTINUE;
 }
 
@@ -296,6 +297,7 @@ flac_load (flacfile_t *ff, sfxblock_t *block, cache_allocator_t allocator)
 	if (flac_read (ff, data, info->frames) < 0)
 		goto bail;
 	SND_SetPaint (sc);
+	SND_SetupResampler (sc, 0);
 	SND_Resample (sc, data, info->frames);
 	sc->head = sc->length;
   bail:

libs/audio/renderer/snd_mem.c

@@ -381,14 +381,12 @@ SND_GetCache (long frames, int rate, int channels,
 
 	stepscale = (float) rate / snd_shm->speed;
 	len = size = frames / stepscale;
-//	printf ("%ld %d\n", frames, size);
 	size *= sizeof (float) * channels;
 	sc = allocator (&block->cache, sizeof (sfxbuffer_t) + size, sfx->name);
 	if (!sc)
 		return 0;
 	memset (sc, 0, sizeof (sfxbuffer_t) + size);
 	sc->length = len;
-	sc->data = sc->sample_data;
 	memcpy (sc->data + len * channels, "\xde\xad\xbe\xef", 4);
 	return sc;
 }

libs/audio/renderer/snd_resample.c

@@ -70,21 +70,8 @@ SND_Resample (sfxbuffer_t *sc, float *data, int length)
 	SRC_DATA    src_data;
 
 	stepscale = (double) snd_shm->speed / inrate;
-
 	outcount = length * stepscale;
 
-	sc->sfx->length = info->frames * stepscale;
-	//printf ("%s %d %g %d\n", sc->sfx->name, length, stepscale, sc->sfx->length);
-	if (info->loopstart != (unsigned int)-1)
-		sc->sfx->loopstart = info->loopstart * stepscale;
-	else
-		sc->sfx->loopstart = (unsigned int)-1;
-
-	sc->channels = info->channels;
-
-	if (!length)
-		return;
-
 	src_data.data_in = data;
 	src_data.data_out = sc->data + sc->head * sc->channels;
 	src_data.input_frames = length;
@@ -97,6 +84,63 @@ SND_Resample (sfxbuffer_t *sc, float *data, int length)
 	check_buffer_integrity (sc, outwidth, __FUNCTION__);
 }
 
+static void
+SND_ResampleStream (sfxbuffer_t *sc, float *data, int length)
+{
+	SRC_DATA    src_data;
+	SRC_STATE  *state = (SRC_STATE *) sc->state;
+
+	int			outcount;
+	double		stepscale;
+	wavinfo_t  *info = sc->sfx->wavinfo (sc->sfx);
+	int         inrate = info->rate;
+
+	stepscale = (double) snd_shm->speed / inrate;
+	outcount = length * stepscale;
+
+	src_data.data_in = data;
+	src_data.data_out = sc->data + sc->head * sc->channels;
+	src_data.input_frames = length;
+	src_data.output_frames = outcount;
+	src_data.src_ratio = stepscale;
+	src_data.end_of_input = 0; //XXX
+
+	src_process (state, &src_data);
+}
+
+void
+SND_SetupResampler (sfxbuffer_t *sc, int streamed)
+{
+	double		stepscale;
+	wavinfo_t  *info = sc->sfx->wavinfo (sc->sfx);
+	int         inrate = info->rate;
+
+	stepscale = (double) snd_shm->speed / inrate;
+
+	sc->sfx->length = info->frames * stepscale;
+	if (info->loopstart != (unsigned int)-1)
+		sc->sfx->loopstart = info->loopstart * stepscale;
+	else
+		sc->sfx->loopstart = (unsigned int)-1;
+
+	sc->channels = info->channels;
+
+	if (streamed) {
+		int         err;
+		sfxstream_t *stream = sc->sfx->data.stream;
+
+		if (snd_shm->speed == inrate) {
+			sc->state = 0;
+			stream->resample = 0;
+		} else {
+			sc->state = src_new (SRC_LINEAR, info->channels, &err);
+			stream->resample = SND_ResampleStream;
+		}
+	} else {
+		sc->state = 0;
+	}
+}
+
 void
 SND_Convert (byte *idata, float *fdata, int frames, int channels, int width)
 {

libs/audio/renderer/snd_sfx.c

@@ -132,12 +132,11 @@ SND_SFX_StreamOpen (sfx_t *sfx, void *file,
 
 	frames = snd_shm->speed * 0.3;
 	frames = (frames + 255) & ~255;
-	size = frames + 1;		// one extra sample for the resampler
+	size = frames * info->channels * sizeof (float);
-	size *= info->channels * sizeof (float);
 
 	stream = calloc (1, sizeof (sfxstream_t) + size);
 	new_sfx->data.stream = stream;
-	memcpy ((byte *) stream->buffer.sample_data + size, "\xde\xad\xbe\xef", 4);
+	memcpy ((byte *) stream->buffer.data + size, "\xde\xad\xbe\xef", 4);
 	stream->file = file;
 	stream->sfx = new_sfx;
 	stream->resample = SND_Resample;
@@ -150,10 +149,9 @@ SND_SFX_StreamOpen (sfx_t *sfx, void *file,
 	stream->buffer.advance = SND_StreamAdvance;
 	stream->buffer.setpos = SND_StreamSetPos;
 	stream->buffer.sfx = new_sfx;
-	stream->buffer.data = &stream->buffer.sample_data[info->channels];
 	SND_SetPaint (&stream->buffer);
 
-	stream->resample (&stream->buffer, 0, 0);		// get sfx setup properly
+	SND_SetupResampler (&stream->buffer, 1);		// get sfx setup properly
 	stream->buffer.setpos (&stream->buffer, 0);		// pre-fill the buffer
 
 	return new_sfx;

libs/audio/renderer/vorbis.c

@@ -174,6 +174,7 @@ vorbis_load (OggVorbis_File *vf, sfxblock_t *block, cache_allocator_t allocator)
 	if (vorbis_read (vf, data, info->frames, info) < 0)
 		goto bail;
 	SND_SetPaint (sc);
+	SND_SetupResampler (sc, 0);
 	SND_Resample (sc, data, info->frames);
 	sc->head = sc->length;
   bail:

libs/audio/renderer/wav.c

@@ -78,6 +78,7 @@ wav_callback_load (void *object, cache_allocator_t allocator)
 						   info->channels, block, allocator);
 	buffer->sfx = sfx;
 	SND_SetPaint (buffer);
+	SND_SetupResampler (buffer, 0);
 	SND_Resample (buffer, fdata, info->frames);
 	buffer->head = buffer->length;
 	free (data);