multimedia/ffmpeg2theora: Applied upstream fix for ffmpeg.

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

4 files changed, 477 insertions, 2 deletions

diff --git a/multimedia/ffmpeg2theora/ffmpeg2theora.SlackBuild b/multimedia/ffmpeg2theora/ffmpeg2theora.SlackBuild
index 7080fef483c0..aae5c080f85e 100644
--- a/multimedia/ffmpeg2theora/ffmpeg2theora.SlackBuild
+++ b/multimedia/ffmpeg2theora/ffmpeg2theora.SlackBuild
@@ -29,7 +29,7 @@
 
 PRGNAM="ffmpeg2theora"
 VERSION=${VERSION:-0.29}
-BUILD=${BUILD:-2}
+BUILD=${BUILD:-3}
 TAG=${TAG:-_SBo}
 
 if [ -z "$ARCH" ]; then
@@ -67,7 +67,13 @@ 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 {} \;
 
-echo "#define AVCODEC_MAX_AUDIO_FRAME_SIZE 192000" >> src/ffmpeg2theora.h
+# 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 < $diff
+done
+
 scons install APPEND_CCFLAGS="$SLKCFLAGS" prefix=/usr destdir=$PKG
 
 find $PKG -print0 | xargs -0 file | grep -e "executable" -e "shared object" | grep ELF \
diff --git a/multimedia/ffmpeg2theora/patches/10-avcodec_max_audio_frame_size.diff b/multimedia/ffmpeg2theora/patches/10-avcodec_max_audio_frame_size.diff
new file mode 100644
index 000000000000..c8d10b81a978
--- /dev/null
+++ b/multimedia/ffmpeg2theora/patches/10-avcodec_max_audio_frame_size.diff
@@ -0,0 +1,42 @@
+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
new file mode 100644
index 000000000000..ef57b82e41d0
--- /dev/null
+++ b/multimedia/ffmpeg2theora/patches/20-dont_print_uninitialized_memory.diff
@@ -0,0 +1,22 @@
+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
new file mode 100644
index 000000000000..2a50e2d31747
--- /dev/null
+++ b/multimedia/ffmpeg2theora/patches/30-avcodec_decode_audio4.diff
@@ -0,0 +1,405 @@
+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);