[issue1455] MP3 from AIF puts noise to the end of the file

2009-10-12 Thread Justin Ruggles 

Justin Ruggles justin.rugg...@gmail.com added the comment:

Baptiste Coudurier wrote:

 Baptiste Coudurier baptiste.coudur...@gmail.com added the comment:
 
 On 10/11/09 1:02 PM, Justin Ruggles wrote:
 Justin Rugglesjustin.rugg...@gmail.com  added the comment:

 trying to replace the file with new file...

 --
 nosy:  -jbr

 _
 FFmpeg issue trackeriss...@roundup.ffmpeg.org
 https://roundup.ffmpeg.org/roundup/ffmpeg/issue1455
 _


 aiff_data_size_2.patch


 diff --git a/libavformat/aiff.c b/libavformat/aiff.c
 index 570e05d..9501a7b 100644
 --- a/libavformat/aiff.c
 +++ b/libavformat/aiff.c
 @@ -46,6 +46,10 @@ static const AVCodecTag codec_aiff_tags[] = {
   #define AIFF0
   #define AIFF_C_VERSION1 0xA2805140

 +typedef struct {
 +int64_t data_end;
 +} AIFFInputContext;
 +
 
 AIFFContext

There is already an AIFFOutputContext in the same file.  I chose to name
the new struct AIFFInputContext to distinguish between the two.  I guess
I could also merge them, even though the fields used are mutually
exclusive between the muxer and demuxer.

Well, really it would be good to split the file, but that's a different
issue...

_
FFmpeg issue tracker iss...@roundup.ffmpeg.org
https://roundup.ffmpeg.org/roundup/ffmpeg/issue1455
_

[issue1455] MP3 from AIF puts noise to the end of the file

2009-10-12 Thread Baptiste Coudurier 

Baptiste Coudurier baptiste.coudur...@gmail.com added the comment:

On 10/12/2009 3:06 PM, Justin Ruggles wrote:

 Justin Rugglesjustin.rugg...@gmail.com  added the comment:

 Baptiste Coudurier wrote:

 Baptiste Coudurierbaptiste.coudur...@gmail.com  added the comment:

 On 10/11/09 1:02 PM, Justin Ruggles wrote:
 Justin Rugglesjustin.rugg...@gmail.com   added the comment:

 trying to replace the file with new file...

 --
 nosy:  -jbr

 _
 FFmpeg issue trackeriss...@roundup.ffmpeg.org
 https://roundup.ffmpeg.org/roundup/ffmpeg/issue1455
 _


 aiff_data_size_2.patch


 diff --git a/libavformat/aiff.c b/libavformat/aiff.c
 index 570e05d..9501a7b 100644
 --- a/libavformat/aiff.c
 +++ b/libavformat/aiff.c
 @@ -46,6 +46,10 @@ static const AVCodecTag codec_aiff_tags[] = {
#define AIFF0
#define AIFF_C_VERSION1 0xA2805140

 +typedef struct {
 +int64_t data_end;
 +} AIFFInputContext;
 +

 AIFFContext

 There is already an AIFFOutputContext in the same file.  I chose to name
 the new struct AIFFInputContext to distinguish between the two.  I guess
 I could also merge them, even though the fields used are mutually
 exclusive between the muxer and demuxer.

Right, name is fine then.

 Well, really it would be good to split the file, but that's a different
 issue...

Definitely welcome :)

_
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https://roundup.ffmpeg.org/roundup/ffmpeg/issue1455
_

[issue1455] MP3 from AIF puts noise to the end of the file

2009-10-12 Thread Justin Ruggles 

Justin Ruggles justin.rugg...@gmail.com added the comment:

fixed in r20219.

--
status: open - closed
substatus: analyzed - fixed

_
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[issue1455] MP3 from AIF puts noise to the end of the file

2009-10-11 Thread Justin Ruggles 

Justin Ruggles justin.rugg...@gmail.com added the comment:

trying to replace the file with new file...

--
nosy:  -jbr

_
FFmpeg issue tracker iss...@roundup.ffmpeg.org
https://roundup.ffmpeg.org/roundup/ffmpeg/issue1455
_diff --git a/libavformat/aiff.c b/libavformat/aiff.c
index 570e05d..9501a7b 100644
--- a/libavformat/aiff.c
+++ b/libavformat/aiff.c
@@ -46,6 +46,10 @@ static const AVCodecTag codec_aiff_tags[] = {
 #define AIFF0
 #define AIFF_C_VERSION1 0xA2805140
 
+typedef struct {
+int64_t data_end;
+} AIFFInputContext;
+
 static enum CodecID aiff_codec_get_id(int bps)
 {
 if (bps = 8)
@@ -314,6 +318,7 @@ static int aiff_read_header(AVFormatContext *s,
 unsigned version = AIFF_C_VERSION1;
 ByteIOContext *pb = s-pb;
 AVStream * st;
+AIFFInputContext *ctx = s-priv_data;
 
 /* check FORM header */
 filesize = get_tag(pb, tag);
@@ -366,6 +371,7 @@ static int aiff_read_header(AVFormatContext *s,
 get_meta(s, comment  , size);
 break;
 case MKTAG('S', 'S', 'N', 'D'): /* Sampled sound chunk */
+ctx-data_end = url_ftell(pb) + size;
 offset = get_be32(pb);  /* Offset of sound data */
 get_be32(pb);   /* BlockSize... don't care */
 offset += url_ftell(pb);/* Compute absolute data offset */
@@ -420,10 +426,18 @@ static int aiff_read_packet(AVFormatContext *s,
 AVPacket *pkt)
 {
 AVStream *st = s-streams[0];
+AIFFInputContext *ctx = s-priv_data;
+int64_t max_size;
 int res;
 
+/* calculate size of remaining data */
+max_size = ctx-data_end - url_ftell(s-pb);
+if (max_size = 0)
+return AVERROR_EOF;
+
 /* Now for that packet */
-res = av_get_packet(s-pb, pkt, (MAX_SIZE / st-codec-block_align) * st-codec-block_align);
+max_size = FFMIN(max_size, (MAX_SIZE / st-codec-block_align) * st-codec-block_align);
+res = av_get_packet(s-pb, pkt, max_size);
 if (res  0)
 return res;
 
@@ -436,7 +450,7 @@ static int aiff_read_packet(AVFormatContext *s,
 AVInputFormat aiff_demuxer = {
 aiff,
 NULL_IF_CONFIG_SMALL(Audio IFF),
-0,
+sizeof(AIFFInputContext),
 aiff_probe,
 aiff_read_header,
 aiff_read_packet,

[issue1455] MP3 from AIF puts noise to the end of the file

2009-10-11 Thread Baptiste Coudurier 

Baptiste Coudurier baptiste.coudur...@gmail.com added the comment:

On 10/11/09 1:02 PM, Justin Ruggles wrote:

 Justin Rugglesjustin.rugg...@gmail.com  added the comment:

 trying to replace the file with new file...

 --
 nosy:  -jbr

 _
 FFmpeg issue trackeriss...@roundup.ffmpeg.org
 https://roundup.ffmpeg.org/roundup/ffmpeg/issue1455
 _


 aiff_data_size_2.patch


 diff --git a/libavformat/aiff.c b/libavformat/aiff.c
 index 570e05d..9501a7b 100644
 --- a/libavformat/aiff.c
 +++ b/libavformat/aiff.c
 @@ -46,6 +46,10 @@ static const AVCodecTag codec_aiff_tags[] = {
   #define AIFF0
   #define AIFF_C_VERSION1 0xA2805140

 +typedef struct {
 +int64_t data_end;
 +} AIFFInputContext;
 +

AIFFContext

   static enum CodecID aiff_codec_get_id(int bps)
   {
   if (bps= 8)
 @@ -314,6 +318,7 @@ static int aiff_read_header(AVFormatContext *s,
   unsigned version = AIFF_C_VERSION1;
   ByteIOContext *pb = s-pb;
   AVStream * st;
 +AIFFInputContext *ctx = s-priv_data;

I think I prefer *aif or *aiff for the variable name.

Patch is fine besides. Thanks for the fix.

_
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_

[issue1455] MP3 from AIF puts noise to the end of the file

2009-10-10 Thread Justin Ruggles 

Justin Ruggles justin.rugg...@gmail.com added the comment:

The issue is that the SSND chunk does not have to be the last chunk in the AIFF
file, but the AIFF demuxer reads audio data until the end of the file.  The
attached patch determines the audio data size so that aiff_read_packet() does
not return data past the end of the SSND chunk.

--
nosy: +jbr
substatus: open - analyzed

_
FFmpeg issue tracker iss...@roundup.ffmpeg.org
https://roundup.ffmpeg.org/roundup/ffmpeg/issue1455
_diff --git a/libavformat/aiff.c b/libavformat/aiff.c
index 570e05d..66088da 100644
--- a/libavformat/aiff.c
+++ b/libavformat/aiff.c
@@ -46,6 +46,12 @@ static const AVCodecTag codec_aiff_tags[] = {
 #define AIFF0
 #define AIFF_C_VERSION1 0xA2805140
 
+typedef struct {
+int64_t data_offset;
+int64_t data_size;
+int max_packet_size;
+} AIFFInputContext;
+
 static enum CodecID aiff_codec_get_id(int bps)
 {
 if (bps = 8)
@@ -314,6 +320,7 @@ static int aiff_read_header(AVFormatContext *s,
 unsigned version = AIFF_C_VERSION1;
 ByteIOContext *pb = s-pb;
 AVStream * st;
+AIFFInputContext *ctx = s-priv_data;
 
 /* check FORM header */
 filesize = get_tag(pb, tag);
@@ -368,7 +375,9 @@ static int aiff_read_header(AVFormatContext *s,
 case MKTAG('S', 'S', 'N', 'D'): /* Sampled sound chunk */
 offset = get_be32(pb);  /* Offset of sound data */
 get_be32(pb);   /* BlockSize... don't care */
+ctx-data_size = size - (8 + offset);
 offset += url_ftell(pb);/* Compute absolute data offset */
+ctx-data_offset = offset;
 if (st-codec-block_align)/* Assume COMM already parsed */
 goto got_sound;
 if (url_is_streamed(pb)) {
@@ -420,10 +429,23 @@ static int aiff_read_packet(AVFormatContext *s,
 AVPacket *pkt)
 {
 AVStream *st = s-streams[0];
+AIFFInputContext *ctx = s-priv_data;
+int64_t max_size, pos;
 int res;
 
+/* calculate size of remaining data */
+pos = url_ftell(s-pb);
+if (pos  ctx-data_offset) {
+url_fseek(s-pb, ctx-data_offset, SEEK_SET);
+pos = ctx-data_offset;
+}
+max_size = ctx-data_size - (pos - ctx-data_offset);
+if (max_size = 0)
+return AVERROR_EOF;
+
 /* Now for that packet */
-res = av_get_packet(s-pb, pkt, (MAX_SIZE / st-codec-block_align) * st-codec-block_align);
+max_size = FFMIN(max_size, (MAX_SIZE / st-codec-block_align) * st-codec-block_align);
+res = av_get_packet(s-pb, pkt, max_size);
 if (res  0)
 return res;
 
@@ -436,7 +458,7 @@ static int aiff_read_packet(AVFormatContext *s,
 AVInputFormat aiff_demuxer = {
 aiff,
 NULL_IF_CONFIG_SMALL(Audio IFF),
-0,
+sizeof(AIFFInputContext),
 aiff_probe,
 aiff_read_header,
 aiff_read_packet,

[issue1455] MP3 from AIF puts noise to the end of the file

2009-10-10 Thread Reimar Döffinger 

Reimar Döffinger b...@reimardoeffinger.de added the comment:

On Sat, Oct 10, 2009 at 05:50:50PM +, Justin Ruggles wrote:
 --- a/libavformat/aiff.c
 +++ b/libavformat/aiff.c
 @@ -46,6 +46,12 @@ static const AVCodecTag codec_aiff_tags[] = {
  #define AIFF0
  #define AIFF_C_VERSION1 0xA2805140
  
 +typedef struct {
 +int64_t data_offset;

I'd expect you could reuse AVFormatContext-data_offset, unless you want
to support multiple SSND chunks but I expect the current code not to
work well for that anyway...

 @@ -368,7 +375,9 @@ static int aiff_read_header(AVFormatContext *s,
  case MKTAG('S', 'S', 'N', 'D'): /* Sampled sound chunk */
  offset = get_be32(pb);  /* Offset of sound data */
  get_be32(pb);   /* BlockSize... don't care */
 +ctx-data_size = size - (8 + offset);
  offset += url_ftell(pb);/* Compute absolute data offset */
 +ctx-data_offset = offset;
  if (st-codec-block_align)/* Assume COMM already parsed */
  goto got_sound;
  if (url_is_streamed(pb)) {
 @@ -420,10 +429,23 @@ static int aiff_read_packet(AVFormatContext *s,
  AVPacket *pkt)
  {
  AVStream *st = s-streams[0];
 +AIFFInputContext *ctx = s-priv_data;
 +int64_t max_size, pos;
  int res;
  
 +/* calculate size of remaining data */
 +pos = url_ftell(s-pb);
 +if (pos  ctx-data_offset) {
 +url_fseek(s-pb, ctx-data_offset, SEEK_SET);
 +pos = ctx-data_offset;
 +}
 +max_size = ctx-data_size - (pos - ctx-data_offset);
 +if (max_size = 0)
 +return AVERROR_EOF;

Why not just have a single data_end field?
And what is the purpose of the pos  ctx-data_offset check anyway?
Also I wonder if/how well this works with multiple SSND chunks and
seeking (probably not that important, but a nice-to-have).

--
title: AIFF demuxer puts noise to the end of the file - MP3 from AIF puts 
noise to the end of the file

_
FFmpeg issue tracker iss...@roundup.ffmpeg.org
https://roundup.ffmpeg.org/roundup/ffmpeg/issue1455
_

[issue1455] MP3 from AIF puts noise to the end of the file

2009-10-10 Thread Reimar Döffinger 

Reimar Döffinger b...@reimardoeffinger.de added the comment:

On Sat, Oct 10, 2009 at 06:40:36PM +, Justin Ruggles wrote:
  And what is the purpose of the pos  ctx-data_offset check anyway?
 
 It was to handle the case when the file is seeked to before the start of
 data.  But if AVFormatContext-data_offset is used as you suggested,
 this should be handled in pcm_read_seek().

If you do not set data_offset, it is already automatically set to the
current position after read_headers, so check there you really need to
do anything at all.

  Also I wonder if/how well this works with multiple SSND chunks and
  seeking (probably not that important, but a nice-to-have).
 
 Right now it doesn't work for demuxing, much less seeking, with multiple
 SSND chunks.  The patch does not change that.  But yes, it should be
 fixed to handle it properly.  Now if I only had a sample...

It might not be allowed. If not even demuxing works currently, just
ignore this possibility for now I'd say.

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https://roundup.ffmpeg.org/roundup/ffmpeg/issue1455
_

[issue1455] MP3 from AIF puts noise to the end of the file

2009-10-05 Thread Jeff Harmon 

New submission from Jeff Harmon jhar...@colorhythm.com:

the following command line on Mac OS X Leopard 10.5.8 running FFMpeg 0.5_5 from
MacPorts:

ffmpeg -i filename.aif -acodec libmp3lame -ab 160k -ac2 -f mp3 preview.mp3

produces excellent results pretty much all the time so far in our testing - even
on a couple other AIFs - but on this AIF, which plays perfectly in
Quicktime/iTunes, it adds a bunch of noise at the end of the file.  Perhaps this
reveals a problem in the code, maybe in my setup?
File '22.050kHz-705kbps-16bit-Stereo.aif' not attached - you can download it 
from https://roundup.ffmpeg.org/roundup/ffmpeg/file550.

--
files: 22.050kHz-705kbps-16bit-Stereo.aif
messages: 7341
priority: normal
status: new
substatus: new
title: MP3 from AIF puts noise to the end of the file
type: bug

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_

[issue1455] MP3 from AIF puts noise to the end of the file

2009-10-05 Thread Carl Eugen Hoyos 

Carl Eugen Hoyos ceho...@rainbow.studorg.tuwien.ac.at added the comment:

Noise at the end of test.wav.

ffmpeg -i 22.050kHz-705kbps-16bit-Stereo.aif test.wav
FFmpeg version SVN-r20167, Copyright (c) 2000-2009 Fabrice Bellard, et al.
  built on Oct  5 2009 10:24:47 with icc 1110
  configuration: --cc=/opt/intel/Compiler/11.1/056/bin/intel64/icc --cpu=core2
--enable-gpl --extra-cflags=-parallel --extra-ldflags=-parallel
--enable-postproc --enable-avfilter --enable-pthreads --enable-nonfree
--enable-version3 --enable-libopencore-amrnb --enable-libopencore-amrwb
--enable-libdirac --enable-libfaac --enable-libfaad --enable-libgsm
--enable-libmp3lame --extra-cflags='-I/usr/include/openjpeg -I/usr/include/gsm'
--enable-libopenjpeg --enable-libschroedinger --enable-libspeex
--enable-libtheora --enable-libvorbis --enable-libxvid --enable-libx264
  libavutil 50. 3. 0 / 50. 3. 0
  libavcodec52.36. 0 / 52.36. 0
  libavformat   52.39. 0 / 52.39. 0
  libavdevice   52. 2. 0 / 52. 2. 0
  libavfilter0. 5. 0 /  0. 5. 0
  libswscale 0. 7. 1 /  0. 7. 1
  libpostproc   51. 2. 0 / 51. 2. 0
[aiff @ 0x145bf90]max_analyze_duration reached
Input #0, aiff, from '22.050kHz-705kbps-16bit-Stereo.aif':
  Duration: 00:00:04.87, start: 0.00, bitrate: 1295 kb/s
Stream #0.0: Audio: pcm_s16be, 22050 Hz, 2 channels, s16, 705 kb/s
Output #0, wav, to 'test.wav':
Stream #0.0: Audio: pcm_s16le, 22050 Hz, 2 channels, s16, 705 kb/s
Stream mapping:
  Stream #0.0 - #0.0
Press [q] to stop encoding
Multiple frames in a packet from stream 0
[pcm_s16be @ 0x1465340]invalid PCM packet
Error while decoding stream #0.0
size= 770kB time=8.94 bitrate= 705.6kbits/s
video:0kB audio:770kB global headers:0kB muxing overhead 0.005577%

--
status: new - open
substatus: new - open

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