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