encoder/vorbis: accept floating point input samples

Improves quality by not squeezing 32 bit samples down to 16 bit, and
then back to 32 bit floating point.  Reduces CPU usage by skipping a
conversion step.

2 files changed, 10 insertions, 8 deletions

diff --git a/NEWS b/NEWS
index 8db1acb2a..968b6be61 100644
--- a/NEWS
+++ b/NEWS
@@ -5,6 +5,7 @@ ver 0.18 (2012/??/??)
   - vorbis: skip 16 bit quantisation, provide float samples
 * encoder:
   - opus: new encoder plugin for the Opus codec
+  - vorbis: accept floating point input samples
 * output:
   - new option "tags" may be used to disable sending tags to output
 * improved decoder/output error reporting
diff --git a/src/encoder/VorbisEncoderPlugin.cxx b/src/encoder/VorbisEncoderPlugin.cxx
index 226a59abf..ff1c6b8df 100644
--- a/src/encoder/VorbisEncoderPlugin.cxx
+++ b/src/encoder/VorbisEncoderPlugin.cxx
@@ -212,7 +212,7 @@ vorbis_encoder_open(struct encoder *_encoder,
 {
 	struct vorbis_encoder *encoder = (struct vorbis_encoder *)_encoder;
 
-	audio_format->format = SAMPLE_FORMAT_S16;
+	audio_format->format = SAMPLE_FORMAT_FLOAT;
 
 	encoder->audio_format = *audio_format;
 
@@ -327,12 +327,12 @@ vorbis_encoder_tag(struct encoder *_encoder, const struct tag *tag,
 }
 
 static void
-pcm16_to_vorbis_buffer(float **dest, const int16_t *src,
-		       unsigned num_frames, unsigned num_channels)
+interleaved_to_vorbis_buffer(float **dest, const float *src,
+			     unsigned num_frames, unsigned num_channels)
 {
 	for (unsigned i = 0; i < num_frames; i++)
 		for (unsigned j = 0; j < num_channels; j++)
-			dest[j][i] = *src++ / 32768.0;
+			dest[j][i] = *src++;
 }
 
 static bool
@@ -347,10 +347,11 @@ vorbis_encoder_write(struct encoder *_encoder,
 
 	/* this is for only 16-bit audio */
 
-	pcm16_to_vorbis_buffer(vorbis_analysis_buffer(&encoder->vd,
-						      num_frames),
-			       (const int16_t *)data,
-			       num_frames, encoder->audio_format.channels);
+	interleaved_to_vorbis_buffer(vorbis_analysis_buffer(&encoder->vd,
+							    num_frames),
+				     (const float *)data,
+				     num_frames,
+				     encoder->audio_format.channels);
 
 	vorbis_analysis_wrote(&encoder->vd, num_frames);
 	vorbis_encoder_blockout(encoder);