From 9e3f843291876b2cd62196c22be1130c0031bfc0 Mon Sep 17 00:00:00 2001 From: Max Kellermann Date: Tue, 2 Oct 2012 08:19:45 +0200 Subject: [PATCH] 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. --- NEWS | 1 + src/encoder/VorbisEncoderPlugin.cxx | 17 +++++++++-------- 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);