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multimedia/ffmpeg2theora: Updated for version 0.30.

Browse source 
Signed-off-by: Willy Sudiarto Raharjo <willysr@slackbuilds.org>
This commit is contained in:
Nishant Limbachia 2016-08-28 14:24:46 +07:00 committed by Willy Sudiarto Raharjo
parent 869f01f64f
commit b5339604d7
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GPG key ID: 887B8374D7333381
5 changed files with 9 additions and 484 deletions
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18
multimedia/ffmpeg2theora/ffmpeg2theora.SlackBuild
View file

@ -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

6
multimedia/ffmpeg2theora/ffmpeg2theora.info
View file

@ -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"

42
multimedia/ffmpeg2theora/patches/10-avcodec_max_audio_frame_size.diff
View file

@ -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) {

22
multimedia/ffmpeg2theora/patches/20-dont_print_uninitialized_memory.diff
View file

@ -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");

405
multimedia/ffmpeg2theora/patches/30-avcodec_decode_audio4.diff
View file

@ -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);