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>

1 files changed, 3 insertions, 5 deletions

diff --git a/libavcodec/libvorbisenc.c b/libavcodec/libvorbisenc.c
index b042e0c..6064dd5 100644
--- a/libavcodec/libvorbisenc.c
+++ b/libavcodec/libvorbisenc.c
@@ -290,18 +290,16 @@ static int oggvorbis_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
 
     /* send samples to libvorbis */
     if (frame) {
-        const float *audio = (const float *)frame->data[0];
         const int samples = frame->nb_samples;
         float **buffer;
         int c, channels = s->vi.channels;
 
         buffer = vorbis_analysis_buffer(&s->vd, samples);
         for (c = 0; c < channels; c++) {
-            int i;
             int co = (channels > 8) ? c :
                      ff_vorbis_encoding_channel_layout_offsets[channels - 1][c];
-            for (i = 0; i < samples; i++)
-                buffer[c][i] = audio[i * channels + co];
+            memcpy(buffer[c], frame->extended_data[co],
+                   samples * sizeof(*buffer[c]));
         }
         if ((ret = vorbis_analysis_wrote(&s->vd, samples)) < 0) {
             av_log(avctx, AV_LOG_ERROR, "error in vorbis_analysis_wrote()\n");
@@ -383,7 +381,7 @@ AVCodec ff_libvorbis_encoder = {
     .encode2        = oggvorbis_encode_frame,
     .close          = oggvorbis_encode_close,
     .capabilities   = CODEC_CAP_DELAY,
-    .sample_fmts    = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLT,
+    .sample_fmts    = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
                                                       AV_SAMPLE_FMT_NONE },
     .long_name      = NULL_IF_CONFIG_SMALL("libvorbis"),
     .priv_class     = &class,