[FFmpeg-cvslog] vorbisenc: cosmetics: rename variable avccontext to avctx

Justin Ruggles git at videolan.org
Tue Feb 26 12:56:32 CET 2013


ffmpeg | branch: master | Justin Ruggles <justin.ruggles at gmail.com> | Sun Feb 24 13:07:27 2013 -0500| [09031b4639667273354d6bcd1705d9f24fc9bdd4] | committer: Justin Ruggles

vorbisenc: cosmetics: rename variable avccontext to avctx

This is consistent with the rest of libavcodec.

> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=09031b4639667273354d6bcd1705d9f24fc9bdd4
---

 libavcodec/vorbisenc.c |   60 ++++++++++++++++++++++++------------------------
 1 file changed, 30 insertions(+), 30 deletions(-)

diff --git a/libavcodec/vorbisenc.c b/libavcodec/vorbisenc.c
index 42713f9..16f4d65 100644
--- a/libavcodec/vorbisenc.c
+++ b/libavcodec/vorbisenc.c
@@ -236,15 +236,15 @@ static int ready_residue(vorbis_enc_residue *rc, vorbis_enc_context *venc)
 }
 
 static int create_vorbis_context(vorbis_enc_context *venc,
-                                 AVCodecContext *avccontext)
+                                 AVCodecContext *avctx)
 {
     vorbis_enc_floor   *fc;
     vorbis_enc_residue *rc;
     vorbis_enc_mapping *mc;
     int i, book, ret;
 
-    venc->channels    = avccontext->channels;
-    venc->sample_rate = avccontext->sample_rate;
+    venc->channels    = avctx->channels;
+    venc->sample_rate = avctx->sample_rate;
     venc->log2_blocksize[0] = venc->log2_blocksize[1] = 11;
 
     venc->ncodebooks = FF_ARRAY_ELEMS(cvectors);
@@ -340,7 +340,7 @@ static int create_vorbis_context(vorbis_enc_context *venc,
         };
         fc->list[i].x = a[i - 2];
     }
-    if (ff_vorbis_ready_floor1_list(avccontext, fc->list, fc->values))
+    if (ff_vorbis_ready_floor1_list(avctx, fc->list, fc->values))
         return AVERROR_BUG;
 
     venc->nresidues = 1;
@@ -1015,10 +1015,10 @@ static int apply_window_and_mdct(vorbis_enc_context *venc,
 }
 
 
-static int vorbis_encode_frame(AVCodecContext *avccontext, AVPacket *avpkt,
+static int vorbis_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
                                const AVFrame *frame, int *got_packet_ptr)
 {
-    vorbis_enc_context *venc = avccontext->priv_data;
+    vorbis_enc_context *venc = avctx->priv_data;
     float **audio = frame ? (float **)frame->extended_data : NULL;
     int samples = frame ? frame->nb_samples : 0;
     vorbis_enc_mode *mode;
@@ -1031,14 +1031,14 @@ static int vorbis_encode_frame(AVCodecContext *avccontext, AVPacket *avpkt,
     samples = 1 << (venc->log2_blocksize[0] - 1);
 
     if ((ret = ff_alloc_packet(avpkt, 8192))) {
-        av_log(avccontext, AV_LOG_ERROR, "Error getting output packet\n");
+        av_log(avctx, AV_LOG_ERROR, "Error getting output packet\n");
         return ret;
     }
 
     init_put_bits(&pb, avpkt->data, avpkt->size);
 
     if (pb.size_in_bits - put_bits_count(&pb) < 1 + ilog(venc->nmodes - 1)) {
-        av_log(avccontext, AV_LOG_ERROR, "output buffer is too small\n");
+        av_log(avctx, AV_LOG_ERROR, "output buffer is too small\n");
         return AVERROR(EINVAL);
     }
 
@@ -1058,7 +1058,7 @@ static int vorbis_encode_frame(AVCodecContext *avccontext, AVPacket *avpkt,
         uint16_t posts[MAX_FLOOR_VALUES];
         floor_fit(venc, fc, &venc->coeffs[i * samples], posts, samples);
         if (floor_encode(venc, fc, &pb, posts, &venc->floor[i * samples], samples)) {
-            av_log(avccontext, AV_LOG_ERROR, "output buffer is too small\n");
+            av_log(avctx, AV_LOG_ERROR, "output buffer is too small\n");
             return AVERROR(EINVAL);
         }
     }
@@ -1082,17 +1082,17 @@ static int vorbis_encode_frame(AVCodecContext *avccontext, AVPacket *avpkt,
 
     if (residue_encode(venc, &venc->residues[mapping->residue[mapping->mux[0]]],
                        &pb, venc->coeffs, samples, venc->channels)) {
-        av_log(avccontext, AV_LOG_ERROR, "output buffer is too small\n");
+        av_log(avctx, AV_LOG_ERROR, "output buffer is too small\n");
         return AVERROR(EINVAL);
     }
 
     flush_put_bits(&pb);
     avpkt->size = put_bits_count(&pb) >> 3;
 
-    avpkt->duration = ff_samples_to_time_base(avccontext, avccontext->frame_size);
+    avpkt->duration = ff_samples_to_time_base(avctx, avctx->frame_size);
     if (frame)
         if (frame->pts != AV_NOPTS_VALUE)
-            avpkt->pts = ff_samples_to_time_base(avccontext, frame->pts);
+            avpkt->pts = ff_samples_to_time_base(avctx, frame->pts);
     else
         avpkt->pts = venc->next_pts;
     if (avpkt->pts != AV_NOPTS_VALUE)
@@ -1103,9 +1103,9 @@ static int vorbis_encode_frame(AVCodecContext *avccontext, AVPacket *avpkt,
 }
 
 
-static av_cold int vorbis_encode_close(AVCodecContext *avccontext)
+static av_cold int vorbis_encode_close(AVCodecContext *avctx)
 {
-    vorbis_enc_context *venc = avccontext->priv_data;
+    vorbis_enc_context *venc = avctx->priv_data;
     int i;
 
     if (venc->codebooks)
@@ -1158,42 +1158,42 @@ static av_cold int vorbis_encode_close(AVCodecContext *avccontext)
     ff_mdct_end(&venc->mdct[1]);
 
 #if FF_API_OLD_ENCODE_AUDIO
-    av_freep(&avccontext->coded_frame);
+    av_freep(&avctx->coded_frame);
 #endif
-    av_freep(&avccontext->extradata);
+    av_freep(&avctx->extradata);
 
     return 0 ;
 }
 
-static av_cold int vorbis_encode_init(AVCodecContext *avccontext)
+static av_cold int vorbis_encode_init(AVCodecContext *avctx)
 {
-    vorbis_enc_context *venc = avccontext->priv_data;
+    vorbis_enc_context *venc = avctx->priv_data;
     int ret;
 
-    if (avccontext->channels != 2) {
-        av_log(avccontext, AV_LOG_ERROR, "Current Libav Vorbis encoder only supports 2 channels.\n");
+    if (avctx->channels != 2) {
+        av_log(avctx, AV_LOG_ERROR, "Current Libav Vorbis encoder only supports 2 channels.\n");
         return -1;
     }
 
-    if ((ret = create_vorbis_context(venc, avccontext)) < 0)
+    if ((ret = create_vorbis_context(venc, avctx)) < 0)
         goto error;
 
-    avccontext->bit_rate = 0;
-    if (avccontext->flags & CODEC_FLAG_QSCALE)
-        venc->quality = avccontext->global_quality / (float)FF_QP2LAMBDA;
+    avctx->bit_rate = 0;
+    if (avctx->flags & CODEC_FLAG_QSCALE)
+        venc->quality = avctx->global_quality / (float)FF_QP2LAMBDA;
     else
         venc->quality = 3.0;
     venc->quality *= venc->quality;
 
-    if ((ret = put_main_header(venc, (uint8_t**)&avccontext->extradata)) < 0)
+    if ((ret = put_main_header(venc, (uint8_t**)&avctx->extradata)) < 0)
         goto error;
-    avccontext->extradata_size = ret;
+    avctx->extradata_size = ret;
 
-    avccontext->frame_size = 1 << (venc->log2_blocksize[0] - 1);
+    avctx->frame_size = 1 << (venc->log2_blocksize[0] - 1);
 
 #if FF_API_OLD_ENCODE_AUDIO
-    avccontext->coded_frame = avcodec_alloc_frame();
-    if (!avccontext->coded_frame) {
+    avctx->coded_frame = avcodec_alloc_frame();
+    if (!avctx->coded_frame) {
         ret = AVERROR(ENOMEM);
         goto error;
     }
@@ -1201,7 +1201,7 @@ static av_cold int vorbis_encode_init(AVCodecContext *avccontext)
 
     return 0;
 error:
-    vorbis_encode_close(avccontext);
+    vorbis_encode_close(avctx);
     return ret;
 }
 



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