[PATCH - resample 1/1] Using float instead of s16 for libsamplerate

[Pavel Hofman]

Natively libsamplerate operates in floats. Using the new convert_float API instead of dropping resoluton in convert_s16.

Signed-off-by: Pavel Hofman <pavel.hofman at ivitera.com>

diff --git a/rate/rate_samplerate.c b/rate/rate_samplerate.c
index 53a0627..c612983 100644
--- a/rate/rate_samplerate.c
+++ b/rate/rate_samplerate.c
@@ -23,12 +23,12 @@
 #include <alsa/asoundlib.h>
 #include <alsa/pcm_rate.h>
 
+typedef float float_t;
+
 struct rate_src {
 	double ratio;
 	int converter;
 	unsigned int channels;
-	float *src_buf;
-	float *dst_buf;
 	SRC_STATE *state;
 	SRC_DATA data;
 };
@@ -53,9 +53,6 @@ static void pcm_src_free(void *obj)
 {
 	struct rate_src *rate = obj;
 
-	free(rate->src_buf);
-	free(rate->dst_buf);
-	rate->src_buf = rate->dst_buf = NULL;
 
 	if (rate->state) {
 		src_delete(rate->state);
@@ -79,17 +76,6 @@ static int pcm_src_init(void *obj, snd_pcm_rate_info_t *info)
 
 	rate->ratio = (double)info->out.rate / (double)info->in.rate;
 
-	free(rate->src_buf);
-	rate->src_buf = malloc(sizeof(float) * rate->channels * info->in.period_size);
-	free(rate->dst_buf);
-	rate->dst_buf = malloc(sizeof(float) * rate->channels * info->out.period_size);
-	if (! rate->src_buf || ! rate->dst_buf) {
-		pcm_src_free(rate);
-		return -ENOMEM;
-	}
-
-	rate->data.data_in = rate->src_buf;
-	rate->data.data_out = rate->dst_buf;
 	rate->data.src_ratio = rate->ratio;
 	rate->data.end_of_input = 0;
 
@@ -112,8 +98,8 @@ static void pcm_src_reset(void *obj)
 	src_reset(rate->state);
 }
 
-static void pcm_src_convert_s16(void *obj, int16_t *dst, unsigned int dst_frames,
-				const int16_t *src, unsigned int src_frames)
+static void pcm_src_convert_float(void *obj, float_t *dst, unsigned int dst_frames,
+				const float_t *src, unsigned int src_frames)
 {
 	struct rate_src *rate = obj;
 	unsigned int ofs;
@@ -121,15 +107,17 @@ static void pcm_src_convert_s16(void *obj, int16_t *dst, unsigned int dst_frames
 	rate->data.input_frames = src_frames;
 	rate->data.output_frames = dst_frames;
 	rate->data.end_of_input = 0;
+	rate->data.data_in = src;
+	rate->data.data_out = dst;
+
 	
-	src_short_to_float_array(src, rate->src_buf, src_frames * rate->channels);
 	src_process(rate->state, &rate->data);
 	if (rate->data.output_frames_gen < dst_frames)
 		ofs = dst_frames - rate->data.output_frames_gen;
 	else
 		ofs = 0;
-	src_float_to_short_array(rate->dst_buf, dst + ofs * rate->channels,
-				 rate->data.output_frames_gen * rate->channels);
+	/*src_float_to_short_array(rate->dst_buf, dst + ofs * rate->channels,
+				 rate->data.output_frames_gen * rate->channels); */
 }
 
 static void pcm_src_close(void *obj)
@@ -157,7 +145,7 @@ static snd_pcm_rate_ops_t pcm_src_ops = {
 	.free = pcm_src_free,
 	.reset = pcm_src_reset,
 	.adjust_pitch = pcm_src_adjust_pitch,
-	.convert_s16 = pcm_src_convert_s16,
+	.convert_float = pcm_src_convert_float,
 	.input_frames = input_frames,
 	.output_frames = output_frames,
 #if SND_PCM_RATE_PLUGIN_VERSION >= 0x010002
-- 
1.7.0.4


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