ppc: Centralize compiler-specific altivec.h #include handling in one place

[ffmpeg] / libavcodec / libvorbis.c

diff --git a/libavcodec/libvorbis.c b/libavcodec/libvorbis.c
index 07973e63799fdea09962922647bdbffd5bc8633e..86c1ed6a75a5360703480ace0627f202af15469e 100644 (file)
--- a/libavcodec/libvorbis.c
+++ b/libavcodec/libvorbis.c
@@ -58,7 +58,7 @@ typedef struct LibvorbisContext {
     vorbis_comment vc;                  /**< VorbisComment info             */
     ogg_packet op;                      /**< ogg packet                     */
     double iblock;                      /**< impulse block bias option      */
-    VorbisParseContext vp;              /**< parse context to get durations */
+    AVVorbisParseContext *vp;           /**< parse context to get durations */
     AudioFrameQueue afq;                /**< frame queue for timestamps     */
 } LibvorbisContext;
 
@@ -91,14 +91,14 @@ static av_cold int libvorbis_setup(vorbis_info *vi, AVCodecContext *avctx)
     double cfreq;
     int ret;
 
-    if (avctx->flags & CODEC_FLAG_QSCALE || !avctx->bit_rate) {
+    if (avctx->flags & AV_CODEC_FLAG_QSCALE || !avctx->bit_rate) {
         /* variable bitrate
          * NOTE: we use the oggenc range of -1 to 10 for global_quality for
          *       user convenience, but libvorbis uses -0.1 to 1.0.
          */
         float q = avctx->global_quality / (float)FF_QP2LAMBDA;
         /* default to 3 if the user did not set quality or bitrate */
-        if (!(avctx->flags & CODEC_FLAG_QSCALE))
+        if (!(avctx->flags & AV_CODEC_FLAG_QSCALE))
             q = 3.0;
         if ((ret = vorbis_encode_setup_vbr(vi, avctx->channels,
                                            avctx->sample_rate,
@@ -163,6 +163,8 @@ static av_cold int libvorbis_encode_close(AVCodecContext *avctx)
     ff_af_queue_close(&s->afq);
     av_freep(&avctx->extradata);
 
+    av_vorbis_parse_free(&s->vp);
+
     return 0;
 }
 
@@ -204,7 +206,7 @@ static av_cold int libvorbis_encode_init(AVCodecContext *avctx)
                                 xiph_len(header_comm.bytes) +
                                 header_code.bytes;
     p = avctx->extradata = av_malloc(avctx->extradata_size +
-                                     FF_INPUT_BUFFER_PADDING_SIZE);
+                                     AV_INPUT_BUFFER_PADDING_SIZE);
     if (!p) {
         ret = AVERROR(ENOMEM);
         goto error;
@@ -221,7 +223,8 @@ static av_cold int libvorbis_encode_init(AVCodecContext *avctx)
     offset += header_code.bytes;
     assert(offset == avctx->extradata_size);
 
-    if ((ret = avpriv_vorbis_parse_extradata(avctx, &s->vp)) < 0) {
+    s->vp = av_vorbis_parse_init(avctx->extradata, avctx->extradata_size);
+    if (!s->vp) {
         av_log(avctx, AV_LOG_ERROR, "invalid extradata\n");
         return ret;
     }
@@ -318,7 +321,7 @@ static int libvorbis_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
 
     avpkt->pts = ff_samples_to_time_base(avctx, op.granulepos);
 
-    duration = avpriv_vorbis_parse_frame(&s->vp, avpkt->data, avpkt->size);
+    duration = av_vorbis_parse_frame(s->vp, avpkt->data, avpkt->size);
     if (duration > 0) {
         /* we do not know encoder delay until we get the first packet from
          * libvorbis, so we have to update the AudioFrameQueue counts */
@@ -343,7 +346,7 @@ AVCodec ff_libvorbis_encoder = {
     .init           = libvorbis_encode_init,
     .encode2        = libvorbis_encode_frame,
     .close          = libvorbis_encode_close,
-    .capabilities   = CODEC_CAP_DELAY,
+    .capabilities   = AV_CODEC_CAP_DELAY,
     .sample_fmts    = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
                                                       AV_SAMPLE_FMT_NONE },
     .priv_class     = &class,