[FFmpeg-cvslog] vorbisenc: use float planar sample format
Justin Ruggles
git at videolan.org
Sun Oct 7 11:43:26 CEST 2012
ffmpeg | branch: master | Justin Ruggles <justin.ruggles at gmail.com> | Mon Aug 20 13:25:27 2012 -0400| [233783e2c12a224035f49562e4cba3ebd8aa7651] | committer: Justin Ruggles
vorbisenc: use float planar sample format
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=233783e2c12a224035f49562e4cba3ebd8aa7651
---
libavcodec/vorbisenc.c | 20 +++++++++-----------
1 file changed, 9 insertions(+), 11 deletions(-)
diff --git a/libavcodec/vorbisenc.c b/libavcodec/vorbisenc.c
index b0e0e68..42713f9 100644
--- a/libavcodec/vorbisenc.c
+++ b/libavcodec/vorbisenc.c
@@ -963,10 +963,10 @@ static int residue_encode(vorbis_enc_context *venc, vorbis_enc_residue *rc,
return 0;
}
-static int apply_window_and_mdct(vorbis_enc_context *venc, const signed short *audio,
- int samples)
+static int apply_window_and_mdct(vorbis_enc_context *venc,
+ float **audio, int samples)
{
- int i, j, channel;
+ int i, channel;
const float * win = venc->win[0];
int window_len = 1 << (venc->log2_blocksize[0] - 1);
float n = (float)(1 << venc->log2_blocksize[0]) / 4.;
@@ -988,9 +988,8 @@ static int apply_window_and_mdct(vorbis_enc_context *venc, const signed short *a
if (samples) {
for (channel = 0; channel < venc->channels; channel++) {
float * offset = venc->samples + channel*window_len*2 + window_len;
- j = channel;
- for (i = 0; i < samples; i++, j += venc->channels)
- offset[i] = audio[j] / 32768. / n * win[window_len - i - 1];
+ for (i = 0; i < samples; i++)
+ offset[i] = audio[channel][i] / n * win[window_len - i - 1];
}
} else {
for (channel = 0; channel < venc->channels; channel++)
@@ -1005,9 +1004,8 @@ static int apply_window_and_mdct(vorbis_enc_context *venc, const signed short *a
if (samples) {
for (channel = 0; channel < venc->channels; channel++) {
float *offset = venc->saved + channel * window_len;
- j = channel;
- for (i = 0; i < samples; i++, j += venc->channels)
- offset[i] = audio[j] / 32768. / n * win[i];
+ for (i = 0; i < samples; i++)
+ offset[i] = audio[channel][i] / n * win[i];
}
venc->have_saved = 1;
} else {
@@ -1021,7 +1019,7 @@ static int vorbis_encode_frame(AVCodecContext *avccontext, AVPacket *avpkt,
const AVFrame *frame, int *got_packet_ptr)
{
vorbis_enc_context *venc = avccontext->priv_data;
- const int16_t *audio = frame ? (const int16_t *)frame->data[0] : NULL;
+ float **audio = frame ? (float **)frame->extended_data : NULL;
int samples = frame ? frame->nb_samples : 0;
vorbis_enc_mode *mode;
vorbis_enc_mapping *mapping;
@@ -1216,7 +1214,7 @@ AVCodec ff_vorbis_encoder = {
.encode2 = vorbis_encode_frame,
.close = vorbis_encode_close,
.capabilities = CODEC_CAP_DELAY | CODEC_CAP_EXPERIMENTAL,
- .sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16,
+ .sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_FLTP,
AV_SAMPLE_FMT_NONE },
.long_name = NULL_IF_CONFIG_SMALL("Vorbis"),
};
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