Resample files when transcoding to mp3 using libsamplerate.

1 files changed, 54 insertions, 0 deletions

diff --git a/resample.c b/resample.c
new file mode 100644
index 0000000..c77367e
--- /dev/null
+++ b/resample.c
@@ -0,0 +1,54 @@
+#include "resample.h"
+
+#include <stdint.h>
+
+static GQuark resample_quark() {
+	return g_quark_from_string("resample");
+}
+
+SRC_STATE *resample_init(int channels, GError **error) {
+	int err;
+	SRC_STATE *state = src_new(SRC_LINEAR, channels, &err);
+	if(!state) {
+		*error = g_error_new(resample_quark(), 0, g_strdup(src_strerror(err)));
+	}
+
+	return state;
+}
+
+void resample(SRC_STATE *state, unsigned int from_rate, unsigned int to_rate, int channels,
+		gpointer in_data, gsize in_size, gpointer *out_data, gsize *out_size, gboolean done, GError **error) {
+	int len = in_size / 2;
+	int frames = in_size / 2 / channels;
+	float *temp_in = g_new(float, len);
+	float *temp_out = g_new(float, len);
+
+	src_short_to_float_array(in_data, temp_in, len);
+
+	SRC_DATA data = {
+		.data_in = temp_in,
+		.data_out = temp_out,
+		.input_frames = frames,
+		.output_frames = frames,
+		.src_ratio = (double)to_rate / (double)from_rate,
+		.end_of_input = done ? 1 : 0
+	};
+
+	int err = src_process(state, &data);
+	if(err) {
+		*error = g_error_new(resample_quark(), 0, g_strdup(src_strerror(err)));
+		goto resample_free;
+	}
+
+	*out_data = g_new(int16_t, data.output_frames_gen * channels);
+	src_float_to_short_array(temp_out, *out_data, data.output_frames_gen * channels);
+	*out_size = sizeof(int16_t) * data.output_frames_gen * channels;
+
+resample_free:
+	g_free(temp_in);
+	g_free(temp_out);
+}
+
+void resample_free(SRC_STATE *state) {
+	src_delete(state);
+}