multimedia/ffmpeg2theora: Updated for version 0.30.

Signed-off-by: Willy Sudiarto Raharjo <willysr@slackbuilds.org>

5 files changed, 9 insertions, 484 deletions

diff --git a/multimedia/ffmpeg2theora/ffmpeg2theora.SlackBuild b/multimedia/ffmpeg2theora/ffmpeg2theora.SlackBuild
index f2efde2012ea5..d562ef4ef0fae 100644
--- a/multimedia/ffmpeg2theora/ffmpeg2theora.SlackBuild
+++ b/multimedia/ffmpeg2theora/ffmpeg2theora.SlackBuild
@@ -3,7 +3,7 @@
 # Slackware Package Build Script for ffmpeg2theora
 # Home Page http://v2v.cc/~j/ffmpeg2theora/
 
-# Copyright (c) 2008-2011, Nishant Limbachia, Hoffman Estates, IL, USA
+# Copyright (c) 2008-2016, Nishant Limbachia, Hoffman Estates, IL, USA
 # (nishant _AT_ mnspace _DOT_ net)
 # All rights reserved.
 #
@@ -28,13 +28,13 @@
 # Modified by SlackBuilds.org --dsomero,rworkman
 
 PRGNAM="ffmpeg2theora"
-VERSION=${VERSION:-0.29}
-BUILD=${BUILD:-3}
+VERSION=${VERSION:-0.30}
+BUILD=${BUILD:-1}
 TAG=${TAG:-_SBo}
 
 if [ -z "$ARCH" ]; then
   case "$( uname -m )" in
-    i?86) ARCH=i486 ;;
+    i?86) ARCH=i586 ;;
     arm*) ARCH=arm ;;
        *) ARCH=$( uname -m ) ;;
   esac
@@ -45,8 +45,8 @@ TMP=${TMP:-/tmp/SBo}
 PKG=$TMP/package-$PRGNAM
 OUTPUT=${OUTPUT:-/tmp}
 
-if [ "$ARCH" = "i486" ]; then
-  SLKCFLAGS="-O2 -march=i486 -mtune=i686"
+if [ "$ARCH" = "i586" ]; then
+  SLKCFLAGS="-O2 -march=i586 -mtune=i686"
 elif [ "$ARCH" = "i686" ]; then
   SLKCFLAGS="-O2 -march=i686 -mtune=i686"
 elif [ "$ARCH" = "x86_64" ]; then
@@ -67,12 +67,6 @@ find -L . \
  \( -perm 666 -o -perm 664 -o -perm 640 -o -perm 600 -o -perm 444 \
  -o -perm 440 -o -perm 400 \) -exec chmod 644 {} \;
 
-# include some patches cherry-picked from upstream's git.
-# two of them are ffmpeg API fixes and one is a small bugfix.
-for diff in $CWD/patches/*.diff; do
-  echo $diff
-  patch -p1 < $diff
-done
 # Fix underlinking on -current
 # thanks to Debian https://bugs.debian.org/cgi-bin/bugreport.cgi?bug=768674
 patch -p1 < $CWD/link-libm.patch
diff --git a/multimedia/ffmpeg2theora/ffmpeg2theora.info b/multimedia/ffmpeg2theora/ffmpeg2theora.info
index 0cab843bd99fa..e5c05376a3425 100644
--- a/multimedia/ffmpeg2theora/ffmpeg2theora.info
+++ b/multimedia/ffmpeg2theora/ffmpeg2theora.info
@@ -1,8 +1,8 @@
 PRGNAM="ffmpeg2theora"
-VERSION="0.29"
+VERSION="0.30"
 HOMEPAGE="http://v2v.cc/~j/ffmpeg2theora/"
-DOWNLOAD="http://v2v.cc/~j/ffmpeg2theora/downloads/ffmpeg2theora-0.29.tar.bz2"
-MD5SUM="90855361d16d1d225fde094a5bae99f5"
+DOWNLOAD="http://v2v.cc/~j/ffmpeg2theora/downloads/ffmpeg2theora-0.30.tar.bz2"
+MD5SUM="b1f0c21097e236c0a4558415a914458f"
 DOWNLOAD_x86_64=""
 MD5SUM_x86_64=""
 REQUIRES="ffmpeg libkate"
diff --git a/multimedia/ffmpeg2theora/patches/10-avcodec_max_audio_frame_size.diff b/multimedia/ffmpeg2theora/patches/10-avcodec_max_audio_frame_size.diff
deleted file mode 100644
index c8d10b81a9782..0000000000000
--- a/multimedia/ffmpeg2theora/patches/10-avcodec_max_audio_frame_size.diff
+++ /dev/null
@@ -1,42 +0,0 @@
-From: Jan Gerber <j@xiph.org>
-Date: Sun, 26 May 2013 13:25:42 +0000 (+0200)
-Subject: don't use deprecated AVCODEC_MAX_AUDIO_FRAME_SIZE
-X-Git-Url: http://git.xiph.org/?p=ffmpeg2theora.git;a=commitdiff_plain;h=d3435a6a83dc656379de9e6523ecf8d565da6ca6
-
-don't use deprecated AVCODEC_MAX_AUDIO_FRAME_SIZE
----
-
-diff --git a/src/ffmpeg2theora.c b/src/ffmpeg2theora.c
-index cde63b9..8bebf28 100644
---- a/src/ffmpeg2theora.c
-+++ b/src/ffmpeg2theora.c
-@@ -47,6 +47,9 @@
- #include "ffmpeg2theora.h"
- #include "avinfo.h"
- 
-+#define MAX_AUDIO_FRAME_SIZE 192000 // 1 second of 48khz 32bit audio
-+
-+
- #define LENGTH(x) (sizeof(x) / sizeof(*x))
- 
- enum {
-@@ -1069,8 +1072,8 @@ void ff2theora_output(ff2theora this) {
-         int first = 1;
-         int audio_eos = 0, video_eos = 0, audio_done = 0, video_done = 0;
-         int ret;
--        int16_t *audio_buf=av_malloc(4*AVCODEC_MAX_AUDIO_FRAME_SIZE);
--        int16_t *resampled=av_malloc(4*AVCODEC_MAX_AUDIO_FRAME_SIZE);
-+        int16_t *audio_buf=av_malloc(4*MAX_AUDIO_FRAME_SIZE);
-+        int16_t *resampled=av_malloc(4*MAX_AUDIO_FRAME_SIZE);
-         int16_t *audio_p=NULL;
-         int no_frames;
-         int no_samples;
-@@ -1531,7 +1534,7 @@ void ff2theora_output(ff2theora this) {
-                 while((audio_eos && !audio_done) || avpkt.size > 0 ) {
-                     int samples=0;
-                     int samples_out=0;
--                    int data_size = 4*AVCODEC_MAX_AUDIO_FRAME_SIZE;
-+                    int data_size = 4*MAX_AUDIO_FRAME_SIZE;
-                     int bytes_per_sample = av_get_bytes_per_sample(aenc->sample_fmt);
- 
-                     if (avpkt.size > 0) {
diff --git a/multimedia/ffmpeg2theora/patches/20-dont_print_uninitialized_memory.diff b/multimedia/ffmpeg2theora/patches/20-dont_print_uninitialized_memory.diff
deleted file mode 100644
index ef57b82e41d00..0000000000000
--- a/multimedia/ffmpeg2theora/patches/20-dont_print_uninitialized_memory.diff
+++ /dev/null
@@ -1,22 +0,0 @@
-From: Jan Gerber <j@xiph.org>
-Date: Fri, 16 Aug 2013 08:40:53 +0000 (+0200)
-Subject: print error instead of printing uninitialized memory to terminal if no input is specified
-X-Git-Url: http://git.xiph.org/?p=ffmpeg2theora.git;a=commitdiff_plain;h=d462b50fa8d9462b847a4e574b2d50fc4d191352
-
-print error instead of printing uninitialized memory to terminal if no input is specified
----
-
-diff --git a/src/ffmpeg2theora.c b/src/ffmpeg2theora.c
-index 8bebf28..410d502 100644
---- a/src/ffmpeg2theora.c
-+++ b/src/ffmpeg2theora.c
-@@ -2773,6 +2773,9 @@ int main(int argc, char **argv) {
-             outputfile_set=1;
-         }
-         optind++;
-+    } else {
-+        fprintf(stderr, "ERROR: no input specified\n");
-+        exit(1);
-     }
-     if(optind<argc) {
-         fprintf(stderr, "WARNING: Only one input file supported, others will be ignored\n");
diff --git a/multimedia/ffmpeg2theora/patches/30-avcodec_decode_audio4.diff b/multimedia/ffmpeg2theora/patches/30-avcodec_decode_audio4.diff
deleted file mode 100644
index 2a50e2d31747e..0000000000000
--- a/multimedia/ffmpeg2theora/patches/30-avcodec_decode_audio4.diff
+++ /dev/null
@@ -1,405 +0,0 @@
-diff -Naur ffmpeg2theora-0.29/SConstruct ffmpeg2theora-0.29.patched/SConstruct
---- ffmpeg2theora-0.29/SConstruct	2012-06-25 13:15:16.000000000 -0400
-+++ ffmpeg2theora-0.29.patched/SConstruct	2014-05-14 15:02:17.000000000 -0400
-@@ -162,6 +162,14 @@
-       '-Iffmpeg'
-     ])
- 
-+  if conf.CheckPKG('libavresample'):
-+    FFMPEG_LIBS.append('libavresample')
-+  else:
-+    FFMPEG_LIBS.append('libswresample')
-+    env.Append(CCFLAGS=[
-+      '-DUSE_SWRESAMPLE'
-+    ])
-+
-   if not conf.CheckPKG(' '.join(FFMPEG_LIBS)): 
-     print """
-         Could not find %s.
-diff -Naur ffmpeg2theora-0.29/src/ffmpeg2theora.c ffmpeg2theora-0.29.patched/src/ffmpeg2theora.c
---- ffmpeg2theora-0.29/src/ffmpeg2theora.c	2014-05-14 14:57:30.000000000 -0400
-+++ ffmpeg2theora-0.29.patched/src/ffmpeg2theora.c	2014-05-14 14:59:43.000000000 -0400
-@@ -33,6 +33,11 @@
- #include "libswscale/swscale.h"
- #include "libpostproc/postprocess.h"
- 
-+#include "libavutil/opt.h"
-+#include "libavutil/channel_layout.h"
-+#include "libavutil/samplefmt.h"
-+#include "libswresample_compat.h"
-+
- #include "theora/theoraenc.h"
- #include "vorbis/codec.h"
- #include "vorbis/vorbisenc.h"
-@@ -537,6 +542,11 @@
-     int synced = this->start_time == 0.0;
-     AVRational display_aspect_ratio, sample_aspect_ratio;
- 
-+    struct SwrContext *swr_ctx;
-+    uint8_t **dst_audio_data = NULL;
-+    int dst_linesize;
-+    int src_nb_samples = 1024, dst_nb_samples, max_dst_nb_samples;
-+
-     if (this->audiostream >= 0 && this->context->nb_streams > this->audiostream) {
-         AVCodecContext *enc = this->context->streams[this->audiostream]->codec;
-         if (enc->codec_type == AVMEDIA_TYPE_AUDIO) {
-@@ -961,22 +971,43 @@
-         if (acodec != NULL && avcodec_open2 (aenc, acodec, NULL) >= 0) {
-             if (this->sample_rate != sample_rate
-                 || this->channels != aenc->channels
--                || aenc->sample_fmt != AV_SAMPLE_FMT_S16) {
--                // values take from libavcodec/resample.c
--                this->audio_resample_ctx = av_audio_resample_init(this->channels,    aenc->channels,
--                                                                  this->sample_rate, sample_rate,
--                                                                  AV_SAMPLE_FMT_S16,    aenc->sample_fmt,
--                                                                  16, 10, 0, 0.8);
--                if (!this->audio_resample_ctx) {
--                    this->channels = aenc->channels;
-+                || aenc->sample_fmt != AV_SAMPLE_FMT_FLTP) {
-+                swr_ctx = swr_alloc();
-+                /* set options */
-+                if (aenc->channel_layout) {
-+                    av_opt_set_int(swr_ctx, "in_channel_layout",    aenc->channel_layout, 0);
-+                } else {
-+                    av_opt_set_int(swr_ctx, "in_channel_layout", av_get_default_channel_layout(aenc->channels), 0);
-+                }
-+                av_opt_set_int(swr_ctx, "in_sample_rate",       aenc->sample_rate, 0);
-+                av_opt_set_sample_fmt(swr_ctx, "in_sample_fmt", aenc->sample_fmt, 0);
-+
-+                av_opt_set_int(swr_ctx, "out_channel_layout", av_get_default_channel_layout(this->channels), 0);
-+                av_opt_set_int(swr_ctx, "out_sample_rate",       this->sample_rate, 0);
-+                av_opt_set_sample_fmt(swr_ctx, "out_sample_fmt", AV_SAMPLE_FMT_FLTP, 0);
-+
-+                /* initialize the resampling context */
-+                if (swr_init(swr_ctx) < 0) {
-+                    fprintf(stderr, "Failed to initialize the resampling context\n");
-+                    exit(1);
-                 }
-+
-+                max_dst_nb_samples = dst_nb_samples =
-+                    av_rescale_rnd(src_nb_samples, this->sample_rate, sample_rate, AV_ROUND_UP);
-+
-+                if (av_samples_alloc_array_and_samples(&dst_audio_data, &dst_linesize, this->channels,
-+                                                         dst_nb_samples, AV_SAMPLE_FMT_FLTP, 0) < 0) {
-+                    fprintf(stderr, "Could not allocate destination samples\n");
-+                    exit(1);
-+                }
-+
-                 if (!info.frontend && this->sample_rate!=sample_rate)
-                     fprintf(stderr, "  Resample: %dHz => %dHz\n", sample_rate,this->sample_rate);
-                 if (!info.frontend && this->channels!=aenc->channels)
-                     fprintf(stderr, "  Channels: %d => %d\n",aenc->channels,this->channels);
-             }
-             else{
--                this->audio_resample_ctx=NULL;
-+                swr_ctx = NULL;
-             }
-         }
-         else{
-@@ -1067,13 +1098,12 @@
-         AVPacket pkt;
-         AVPacket avpkt;
-         int len1;
--        int got_picture;
-+        int got_frame;
-         int first = 1;
-         int audio_eos = 0, video_eos = 0, audio_done = 0, video_done = 0;
-         int ret;
--        int16_t *audio_buf=av_malloc(4*MAX_AUDIO_FRAME_SIZE);
--        int16_t *resampled=av_malloc(4*MAX_AUDIO_FRAME_SIZE);
--        int16_t *audio_p=NULL;
-+        AVFrame *audio_frame = NULL;
-+        uint8_t **audio_p = NULL;
-         int no_frames;
-         int no_samples;
- 
-@@ -1369,7 +1399,7 @@
-                       first frame decodec in case its not a keyframe
-                     */
-                     if (pkt.stream_index == this->video_index) {
--                      avcodec_decode_video2(venc, frame, &got_picture, &pkt);
-+                      avcodec_decode_video2(venc, frame, &got_frame, &pkt);
-                     }
-                     av_free_packet (&pkt);
-                     continue;
-@@ -1388,9 +1418,9 @@
-                 while(video_eos || avpkt.size > 0) {
-                     int dups = 0;
-                     static th_ycbcr_buffer ycbcr;
--                    len1 = avcodec_decode_video2(venc, frame, &got_picture, &avpkt);
-+                    len1 = avcodec_decode_video2(venc, frame, &got_frame, &avpkt);
-                     if (len1>=0) {
--                        if (got_picture) {
-+                        if (got_frame) {
-                             // this is disabled by default since it does not work
-                             // for all input formats the way it should.
-                             if (this->sync == 1 && pkt.dts != AV_NOPTS_VALUE) {
-@@ -1427,7 +1457,7 @@
- 
-                             if (venc_pix_fmt != this->pix_fmt) {
-                                 sws_scale(this->sws_colorspace_ctx,
--                                frame->data, frame->linesize, 0, display_height,
-+                                (const uint8_t * const*)frame->data, frame->linesize, 0, display_height,
-                                 output_tmp->data, output_tmp->linesize);
-                             }
-                             else{
-@@ -1471,7 +1501,7 @@
-                             }
-                             if (this->sws_scale_ctx) {
-                                 sws_scale(this->sws_scale_ctx,
--                                    output_cropped->data,
-+                                    (const uint8_t * const*)output_cropped->data,
-                                     output_cropped->linesize, 0,
-                                     display_height - (this->frame_topBand + this->frame_bottomBand),
-                                     output_resized->data,
-@@ -1499,7 +1529,7 @@
-                     //now output_resized
- 
-                     if (!first) {
--                        if (got_picture || video_eos) {
-+                        if (got_frame || video_eos) {
-                             prepare_ycbcr_buffer(this, ycbcr, output_buffered);
-                             if(dups>0) {
-                                 //this only works if dups < keyint,
-@@ -1519,11 +1549,11 @@
-                                 info.videotime = this->frame_count / av_q2d(this->framerate);
-                         }
-                     }
--                    if (got_picture) {
-+                    if (got_frame) {
-                         first=0;
-                         av_picture_copy((AVPicture *)output_buffered, (AVPicture *)output_padded, this->pix_fmt, this->frame_width, this->frame_height);
-                     }
--                    if (!got_picture) {
-+                    if (!got_frame) {
-                         break;
-                     }
-                 }
-@@ -1531,42 +1561,62 @@
-             if (info.passno!=1)
-               if ((audio_eos && !audio_done) || (ret >= 0 && pkt.stream_index == this->audio_index)) {
-                 while((audio_eos && !audio_done) || avpkt.size > 0 ) {
--                    int samples=0;
--                    int samples_out=0;
--                    int data_size = 4*MAX_AUDIO_FRAME_SIZE;
-                     int bytes_per_sample = av_get_bytes_per_sample(aenc->sample_fmt);
- 
-                     if (avpkt.size > 0) {
--                        len1 = avcodec_decode_audio3(astream->codec, audio_buf, &data_size, &avpkt);
-+                        if (!audio_frame && !(audio_frame = avcodec_alloc_frame())) {
-+                            fprintf(stderr, "Failed to allocate memory\n");
-+                            exit(1);
-+                        }
-+                        len1 = avcodec_decode_audio4(astream->codec, audio_frame, &got_frame, &avpkt);
-                         if (len1 < 0) {
-                             /* if error, we skip the frame */
-                             break;
-                         }
--                        avpkt.size -= len1;
--                        avpkt.data += len1;
--                        if (data_size >0) {
--                            samples = data_size / (aenc->channels * bytes_per_sample);
--                            samples_out = samples;
--                            if (this->audio_resample_ctx) {
--                                samples_out = audio_resample(this->audio_resample_ctx, resampled, audio_buf, samples);
--                                audio_p = resampled;
-+                        /* Some audio decoders decode only part of the packet, and have to be
-+                         * called again with the remainder of the packet data.
-+                         * Sample: http://fate-suite.libav.org/lossless-audio/luckynight-partial.shn
-+                         * Also, some decoders might over-read the packet. */
-+                        len1 = FFMIN(len1, avpkt.size);
-+                        if (got_frame) {
-+                            dst_nb_samples = audio_frame->nb_samples;
-+                            if (swr_ctx) {
-+                                dst_nb_samples = av_rescale_rnd(audio_frame->nb_samples,
-+                                    this->sample_rate, aenc->sample_rate, AV_ROUND_UP);
-+                                if (dst_nb_samples > max_dst_nb_samples) {
-+                                    av_free(dst_audio_data[0]);
-+                                    if (av_samples_alloc(dst_audio_data, &dst_linesize, this->channels,
-+                                                           dst_nb_samples, AV_SAMPLE_FMT_FLTP, 1) < 0) {
-+                                        fprintf(stderr, "Error while converting audio\n");
-+                                        exit(1);
-+                                    }
-+                                    max_dst_nb_samples = dst_nb_samples;
-+                                }
-+                                if (swr_convert(swr_ctx, dst_audio_data, dst_nb_samples,
-+                                    (const uint8_t**)audio_frame->extended_data, audio_frame->nb_samples) < 0) {
-+                                    fprintf(stderr, "Error while converting audio\n");
-+                                    exit(1);
-+                                }
-+                                audio_p = dst_audio_data;
-+                            } else {
-+                                audio_p = audio_frame->extended_data;
-                             }
--                            else
--                                audio_p = audio_buf;
-                         }
-+                        avpkt.size -= len1;
-+                        avpkt.data += len1;
-                     }
--
--                    if (no_samples > 0 && this->sample_count + samples_out > no_samples) {
--                        audio_eos = 1;
--                        samples_out = no_samples - this->sample_count;
--                        if (samples_out <= 0) {
--                            break;
-+                    if(got_frame || audio_eos) {
-+                        if (no_samples > 0 && this->sample_count + dst_nb_samples > no_samples) {
-+                            audio_eos = 1;
-+                            dst_nb_samples = no_samples - this->sample_count;
-+                            if (dst_nb_samples <= 0) {
-+                                break;
-+                            }
-                         }
-+                        oggmux_add_audio(&info, audio_p, dst_nb_samples, audio_eos);
-+                        avcodec_free_frame(&audio_frame);
-+                        this->sample_count += dst_nb_samples;
-                     }
--
--                    oggmux_add_audio(&info, audio_p,
--                        samples_out * (this->channels), samples_out, audio_eos);
--                    this->sample_count += samples_out;
-                     if(audio_eos) {
-                         audio_done = 1;
-                     }
-@@ -1751,8 +1801,8 @@
-             avcodec_close(venc);
-         }
-         if (this->audio_index >= 0) {
--            if (this->audio_resample_ctx)
--                audio_resample_close(this->audio_resample_ctx);
-+            if (swr_ctx)
-+                swr_free(&swr_ctx);
-             avcodec_close(aenc);
-         }
- 
-@@ -1773,8 +1823,12 @@
-             frame_dealloc(output_cropped_p);
-             frame_dealloc(output_padded_p);
-         }
--        av_free(audio_buf);
--        av_free(resampled);
-+        if (dst_audio_data)
-+            av_freep(&dst_audio_data[0]);
-+        av_freep(&dst_audio_data);
-+        if(swr_ctx) {
-+            swr_close(swr_ctx);
-+        }
-     }
-     else{
-         fprintf(stderr, "No video or audio stream found.\n");
-diff -Naur ffmpeg2theora-0.29/src/ffmpeg2theora.c.orig ffmpeg2theora-0.29.patched/src/ffmpeg2theora.c.orig
---- ffmpeg2theora-0.29/src/ffmpeg2theora.c.orig	2014-05-14 14:57:25.000000000 -0400
-+++ ffmpeg2theora-0.29.patched/src/ffmpeg2theora.c.orig	2014-05-14 14:57:30.000000000 -0400
-@@ -2772,6 +2772,9 @@
-             outputfile_set=1;
-         }
-         optind++;
-+    } else {
-+        fprintf(stderr, "ERROR: no input specified\n");
-+        exit(1);
-     }
-     if(optind<argc) {
-         fprintf(stderr, "WARNING: Only one input file supported, others will be ignored\n");
-diff -Naur ffmpeg2theora-0.29/src/ffmpeg2theora.h ffmpeg2theora-0.29.patched/src/ffmpeg2theora.h
---- ffmpeg2theora-0.29/src/ffmpeg2theora.h	2010-10-10 10:56:00.000000000 -0400
-+++ ffmpeg2theora-0.29.patched/src/ffmpeg2theora.h	2014-05-14 14:59:43.000000000 -0400
-@@ -62,7 +62,6 @@
-     double fps;
-     struct SwsContext *sws_colorspace_ctx; /* for image resampling/resizing */
-     struct SwsContext *sws_scale_ctx; /* for image resampling/resizing */
--    ReSampleContext *audio_resample_ctx;
-     ogg_int32_t aspect_numerator;
-     ogg_int32_t aspect_denominator;
-     int colorspace;
-diff -Naur ffmpeg2theora-0.29/src/libswresample_compat.h ffmpeg2theora-0.29.patched/src/libswresample_compat.h
---- ffmpeg2theora-0.29/src/libswresample_compat.h	1969-12-31 19:00:00.000000000 -0500
-+++ ffmpeg2theora-0.29.patched/src/libswresample_compat.h	2014-05-14 14:59:43.000000000 -0400
-@@ -0,0 +1,23 @@
-+// This header serves to smooth out the differences in FFmpeg and LibAV.
-+
-+#ifdef USE_SWRESAMPLE
-+
-+    #include <libswresample/swresample.h>
-+
-+    //swr does not have the equivalent so this does nothing
-+    void swr_close(SwrContext *ctx) {};
-+
-+#else
-+
-+    #include <libavresample/avresample.h>
-+
-+    #define SwrContext AVAudioResampleContext
-+    #define swr_init(ctx) avresample_open(ctx)
-+    #define swr_close(ctx) avresample_close(ctx)
-+    #define swr_free(ctx) avresample_free(ctx)
-+    #define swr_alloc() avresample_alloc_context()
-+    #define swr_get_delay(ctx, ...) avresample_get_delay(ctx)
-+    #define swr_convert(ctx, out, out_count, in, in_count) \
-+       avresample_convert(ctx, out, 0, out_count, (uint8_t **)in, 0, in_count)
-+
-+#endif
-diff -Naur ffmpeg2theora-0.29/src/theorautils.c ffmpeg2theora-0.29.patched/src/theorautils.c
---- ffmpeg2theora-0.29/src/theorautils.c	2012-06-21 17:36:01.000000000 -0400
-+++ ffmpeg2theora-0.29.patched/src/theorautils.c	2014-05-14 14:59:43.000000000 -0400
-@@ -1219,17 +1219,16 @@
- /**
-  * adds audio samples to encoding sink
-  * @param buffer pointer to buffer
-- * @param bytes bytes in buffer
-  * @param samples samples in buffer
-  * @param e_o_s 1 indicates end of stream.
-  */
--void oggmux_add_audio (oggmux_info *info, int16_t * buffer, int bytes, int samples, int e_o_s) {
-+void oggmux_add_audio (oggmux_info *info, uint8_t **buffer, int samples, int e_o_s) {
-     ogg_packet op;
- 
-     int i, j, k, count = 0;
-     float **vorbis_buffer;
- 
--    if (bytes <= 0 && samples <= 0) {
-+    if (samples <= 0) {
-         /* end of audio stream */
-         if (e_o_s)
-             vorbis_analysis_wrote (&info->vd, 0);
-@@ -1252,7 +1251,7 @@
-                         default: k = j;
-                     }
-                 }
--                vorbis_buffer[k][i] = buffer[count++] / 32768.f;
-+                vorbis_buffer[k][i] = ((const float  *)buffer[j])[i];
-             }
-         }
-         vorbis_analysis_wrote (&info->vd, samples);
-@@ -1291,8 +1290,8 @@
-                 if (op.packetno != 4) {
-                     /* We only expect negative start granule in the first content
-                        packet, not any of the others... */
--                    fprintf(stderr, "WARNING: vorbis packet %lld has calculated start"
--                            " granule of %lld, but it should be non-negative!",
-+                    fprintf(stderr, "WARNING: vorbis packet %" PRId64 " has calculated start"
-+                            " granule of %" PRId64 ", but it should be non-negative!",
-                             op.packetno, start_granule);
-                 }
-                 start_granule = 0;
-@@ -1302,7 +1301,7 @@
-                    allowed by the specification in the last packet only, and the
-                    trailing samples should be discarded and not played/indexed. */
-                 if (!op.e_o_s) {
--                    fprintf(stderr, "WARNING: vorbis packet %lld (granulepos %lld) starts before"
-+                    fprintf(stderr, "WARNING: vorbis packet %" PRId64 " (granulepos %" PRId64 ") starts before"
-                             " the end of the preceeding packet!", op.packetno, op.granulepos);
-                 }
-                 start_granule = info->vorbis_granulepos;
-diff -Naur ffmpeg2theora-0.29/src/theorautils.h ffmpeg2theora-0.29.patched/src/theorautils.h
---- ffmpeg2theora-0.29/src/theorautils.h	2011-09-15 16:20:46.000000000 -0400
-+++ ffmpeg2theora-0.29.patched/src/theorautils.h	2014-05-14 14:59:43.000000000 -0400
-@@ -168,7 +168,7 @@
- extern void oggmux_setup_kate_streams(oggmux_info *info, int n_kate_streams);
- extern void oggmux_init (oggmux_info *info);
- extern void oggmux_add_video (oggmux_info *info, th_ycbcr_buffer ycbcr, int e_o_s);
--extern void oggmux_add_audio (oggmux_info *info, int16_t * readbuffer, int bytesread, int samplesread,int e_o_s);
-+extern void oggmux_add_audio (oggmux_info *info, uint8_t **buffer, int samples,int e_o_s);
- #ifdef HAVE_KATE
- extern void oggmux_add_kate_text (oggmux_info *info, int idx, double t0, double t1, const char *text, size_t len, int x1, int x2, int y1, int y2);
- extern void oggmux_add_kate_image (oggmux_info *info, int idx, double t0, double t1, const kate_region *kr, const kate_palette *kp, const kate_bitmap *kb);