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authorB. Watson <yalhcru@gmail.com>2014-05-15 03:40:17 +0700
committerWilly Sudiarto Raharjo <willysr@slackbuilds.org>2014-05-15 03:40:17 +0700
commite218c10caacaeb661c4b2027920cd3466b70a525 (patch)
tree345b7ee8954a3c8c0aecb88c96e37700b94f8d46
parentb4ef1ccdc7e47dac4ffc68ea2a19f46beffd4192 (diff)
downloadslackbuilds-e218c10caacaeb661c4b2027920cd3466b70a525.tar.gz
multimedia/ffmpeg2theora: Applied upstream fix for ffmpeg.
Signed-off-by: Willy Sudiarto Raharjo <willysr@slackbuilds.org>
-rw-r--r--multimedia/ffmpeg2theora/ffmpeg2theora.SlackBuild10
-rw-r--r--multimedia/ffmpeg2theora/patches/10-avcodec_max_audio_frame_size.diff42
-rw-r--r--multimedia/ffmpeg2theora/patches/20-dont_print_uninitialized_memory.diff22
-rw-r--r--multimedia/ffmpeg2theora/patches/30-avcodec_decode_audio4.diff405
4 files changed, 477 insertions, 2 deletions
diff --git a/multimedia/ffmpeg2theora/ffmpeg2theora.SlackBuild b/multimedia/ffmpeg2theora/ffmpeg2theora.SlackBuild
index 7080fef483..aae5c080f8 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 0000000000..c8d10b81a9
--- /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 0000000000..ef57b82e41
--- /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 0000000000..2a50e2d317
--- /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);