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author | Michael Niedermayer <michaelni@gmx.at> | 2012-10-07 11:23:29 +0200 |
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committer | Michael Niedermayer <michaelni@gmx.at> | 2012-10-07 11:28:38 +0200 |
commit | 79d30321a29dc648d5a475ce5086b2760d5d8c12 (patch) | |
tree | c712b09b56a0937ca08a1c89365addd8e4e33d4b /libavcodec/vorbisenc.c | |
parent | 537ef8bebf8a35aab448db6ec876e275a10f0f15 (diff) | |
parent | 31b2262dca9cc77709d20c45610ec8030e7f9257 (diff) | |
download | ffmpeg-79d30321a29dc648d5a475ce5086b2760d5d8c12.tar.gz |
Merge remote-tracking branch 'qatar/master'
* qatar/master:
wmaenc: use float planar sample format
(e)ac3enc: use planar sample format
aacenc: use planar sample format
adpcmenc: use planar sample format for adpcm_ima_wav and adpcm_ima_qt
adpcmenc: move 'ch' variable to higher scope
adpcmenc: fix 3 instances of variable shadowing
adpcm_ima_wav: simplify encoding
libvorbis: use planar sample format
libmp3lame: use planar sample formats
vorbisenc: use float planar sample format
ffm: do not write or read the audio sample format
parseutils: fix parsing of invalid alpha values
doc/RELEASE_NOTES: update for the 9 release.
smoothstreamingenc: Add a more verbose error message
smoothstreamingenc: Ignore the return value from mkdir
smoothstreamingenc: Try writing a manifest when opening the muxer
smoothstreamingenc: Move the output_chunk_list and write_manifest functions up
smoothstreamingenc: Properly return errors from ism_flush to the caller
smoothstreamingenc: Check the output UrlContext before accessing it
Conflicts:
doc/RELEASE_NOTES
libavcodec/aacenc.c
libavcodec/ac3enc_template.c
libavcodec/wmaenc.c
tests/ref/lavf/ffm
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Diffstat (limited to 'libavcodec/vorbisenc.c')
-rw-r--r-- | libavcodec/vorbisenc.c | 20 |
1 files changed, 9 insertions, 11 deletions
diff --git a/libavcodec/vorbisenc.c b/libavcodec/vorbisenc.c index b88791f274..74d6331ee3 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 { @@ -1020,7 +1018,7 @@ 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; @@ -1213,7 +1211,7 @@ 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"), }; |