[FFmpeg-devel] I dont know asm too well =( but I have a tip: mem+sync and machine learning

[Jon Maser]

i got this idea from postresql, a very small sql database server

i see ffmpeg is broken down into optimized asm libraries for embedding
in c etc programs, and i know a few things about asm, ie, hardware
registers on cpu+, pump in data into a register, get data out, think
it goes down the system bus

but i was thinking someone could implement an upload to memory feature
when uploading/decoding (kinda like a buffer, maybe a queue? add
matrix field of vid, range of tiles, algorithm.. pop when done,
algorithm etc ,  ) video and audio using asm, and implement a sync
mechanism (basically a queue of registers from the matrice with data
you need to process for stuff like deinterlace or fast moving video )

dont know how av is synced, maybe you can implement a buffer and a
stream pointer system?

you can do (i surmise)

scale_cuda_register_with_image img, offset

or

upload to memory
scale_cuda register_with_image image, offset
upscale_cuda_register_with_image image,  offset
sync

you can also use machine learning, which deals with matrices and i
think hardware estimation of certain numbers

it can also work well in writing/reading files, but now that i think
of it c/dma allows you to upload data to the memory and access it
until its freed, but you can do stuff like stream data into pointers,
maybe that helps, would like

it wont give a major preformance gain, but im hoping for the best


hopefully that works, ffmpeg is good software! cant wait to hit usenet B)
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[FFmpeg-devel] [RFC PATCH] avformat/rtpdec: Audio level RTP extension RFC6464

[jon]

From: Jonathan Baudanza 

libwebrtc will add audio level (in decibels) and VAD status to each RTP packet.

This patch will add both values to the packet sidedata.

I've been using this patch in production for about a year on live audio RTP
streams to detect when users are speaking without needing to decode the audio
data.
---
 libavcodec/avpacket.c |  1 +
 libavcodec/defs.h | 15 
 libavcodec/packet.h   |  5 +++
 libavformat/rtpdec.c  | 87 +++
 libavformat/rtpdec.h  |  5 +++
 libavformat/rtsp.c| 16 
 libavformat/rtsp.h|  2 +
 7 files changed, 131 insertions(+)

diff --git a/libavcodec/avpacket.c b/libavcodec/avpacket.c
index e118bbaad1..73e0341bf7 100644
--- a/libavcodec/avpacket.c
+++ b/libavcodec/avpacket.c
@@ -305,6 +305,7 @@ const char *av_packet_side_data_name(enum 
AVPacketSideDataType type)
 case AV_PKT_DATA_IAMF_MIX_GAIN_PARAM:return "IAMF Mix Gain 
Parameter Data";
 case AV_PKT_DATA_IAMF_DEMIXING_INFO_PARAM:   return "IAMF Demixing Info 
Parameter Data";
 case AV_PKT_DATA_IAMF_RECON_GAIN_INFO_PARAM: return "IAMF Recon Gain Info 
Parameter Data";
+case AV_PKT_DATA_SSRC_AUDIO_LEVEL:   return "RTP SSRC Audio Level";
 }
 return NULL;
 }
diff --git a/libavcodec/defs.h b/libavcodec/defs.h
index 00d840ec19..87e8814760 100644
--- a/libavcodec/defs.h
+++ b/libavcodec/defs.h
@@ -323,6 +323,21 @@ typedef struct AVProducerReferenceTime {
 int flags;
 } AVProducerReferenceTime;
 
+/**
+ * Audio level structure from the ssrc-audio-level RTP header extension.
+ */
+typedef struct AVAudioLevel {
+/**
+ * Audio level for this packet, measured in dBov: -127 - 0
+ */
+int8_t level;
+
+/**
+ * Set to 1 if the encoder believes this packet contains voice.
+ */
+int voice;
+} AVAudioLevel;
+
 /**
  * Encode extradata length to a buffer. Used by xiph codecs.
  *
diff --git a/libavcodec/packet.h b/libavcodec/packet.h
index 8558ae849e..f7f1deb6e0 100644
--- a/libavcodec/packet.h
+++ b/libavcodec/packet.h
@@ -330,6 +330,11 @@ enum AVPacketSideDataType {
 */
 AV_PKT_DATA_AMBIENT_VIEWING_ENVIRONMENT,
 
+/**
+ * Audio Level and VAD data from the RTP header extension as defined by 
RFC 6464.
+ */
+AV_PKT_DATA_SSRC_AUDIO_LEVEL,
+
 /**
  * The number of side data types.
  * This is not part of the public API/ABI in the sense that it may
diff --git a/libavformat/rtpdec.c b/libavformat/rtpdec.c
index fa7544cc07..479ea2e245 100644
--- a/libavformat/rtpdec.c
+++ b/libavformat/rtpdec.c
@@ -694,6 +694,79 @@ static void finalize_packet(RTPDemuxContext *s, AVPacket 
*pkt, uint32_t timestam
s->base_timestamp;
 }
 
+
+static const uint8_t* find_header_ext_data(int id, const uint8_t *buf, uint8_t 
*len) {
+int buflen = (AV_RB16(buf + 2)) * 4;
+
+const uint8_t *p = buf + 4;
+int idx = 0;
+int this_id;
+int this_len;
+
+// This is a one-byte extention format, as defined by RFC rfc5285
+if (buf[0] == 0xbe && buf[1] == 0xde) {
+while (idx + 1 < buflen) {
+if (p[idx] == 0) {
+idx++; // skip padding
+} else {
+this_id = p[idx] >> 4;
+this_len = (p[idx] & 0xf) + 1;
+
+// spec says 15 is reserved
+if (this_id == 15) {
+break; // reject
+}
+
+if (this_id == id) {
+if (this_len > buflen - idx - 1) {
+break; // reject
+}
+
+if (len != NULL)
+*len = this_len;
+
+return p + idx + 1;
+}
+
+idx += 1 + this_len;
+}
+}
+} else if (buf[0] == 0x10 && (buf[1] & 0xff) == 0) {
+// This is a two-byte extention format
+while (idx + 1 < buflen) {
+if (p[idx] == 0) {
+idx++; // Skip padding
+} else {
+this_id = p[idx];
+this_len = p[idx + 1];
+
+// spec says 15 is reserved
+if (this_id == 15) {
+break; // reject
+}
+
+if (this_id == id) {
+if (this_len > buflen - idx - 2) {
+break; // reject
+}
+
+if (len != NULL)
+*len = this_len;
+return p + idx + 2;
+}
+
+idx += 2 + this_len;
+}
+}
+}
+
+if (len != NULL)
+*len = 0;
+
+return NULL;
+}
+
+
 static int rtp_parse_packet_internal(RTPDemuxContext *s, AVPacket *pkt,
  const uint8_t *buf, int len)
 {
@@ -703,6 +776,7 @@ static int rtp_parse_packet_internal(RTPDemuxContext *s, 
AVPacket *pkt,
 AVStream *st;
 uint32_t timestamp;
 int rv = 0;
+

[FFmpeg-devel] [PATCH] libavcodec/qsvenc.c: Set height and width alignment when encoding with mjpeg_qsv

[Jon Cook]

From: JonCookCubic 

Currently width_align and height_align are zero when encoding with mjpeg_qsv 
which causes "Error submitting the frame for encoding". This patch sets the 
alignments.

There is a little bit more about the problem here 
http://ffmpeg.org/pipermail/ffmpeg-user/2019-November/046143.html

The following command line shows the problem (output is without the patch 
applied). Input video is not important, any video will do.

$ ./ffmpeg.exe -loglevel debug -y -i C:/Users/jcook/Desktop/sample.avi -c:v  
mjpeg_qsv output.mp4
ffmpeg version N-95883-ga2fbdc6898 Copyright (c) 2000-2019 the FFmpeg developers
  built with Microsoft (R) C/C++ Optimizing Compiler Version 19.16.27031.1 for 
x64
  configuration: --toolchain=msvc --target-os=win64 --arch=x86_64 
--prefix=/ffmpeg --enable-shared --disable-static --enable-yasm --enable-asm 
--enable-libmfx --enable-nonfree
  libavutil  56. 36.101 / 56. 36.101
  libavcodec 58. 62.100 / 58. 62.100
  libavformat58. 35.100 / 58. 35.100
  libavdevice58.  9.101 / 58.  9.101
  libavfilter 7. 67.100 /  7. 67.100
  libswscale  5.  6.100 /  5.  6.100
  libswresample   3.  6.100 /  3.  6.100
Splitting the commandline.
Reading option '-loglevel' ... matched as option 'loglevel' (set logging level) 
with argument 'debug'.
Reading option '-y' ... matched as option 'y' (overwrite output files) with 
argument '1'.
Reading option '-i' ... matched as input url with argument 
'C:/Users/jcook/Desktop/sample.avi'.
Reading option '-c:v' ... matched as option 'c' (codec name) with argument 
'mjpeg_qsv'.
Reading option 'output.mp4' ... matched as output url.
Finished splitting the commandline.
Parsing a group of options: global .
Applying option loglevel (set logging level) with argument debug.
Applying option y (overwrite output files) with argument 1.
Successfully parsed a group of options.
Parsing a group of options: input url C:/Users/jcook/Desktop/sample.avi.
Successfully parsed a group of options.
Opening an input file: C:/Users/jcook/Desktop/sample.avi.
[NULL @ 01E0699F8540] Opening 'C:/Users/jcook/Desktop/sample.avi' for 
reading
[file @ 01E0699D8280] Setting default whitelist 'file,crypto'
[avi @ 01E0699F8540] Format avi probed with size=2048 and score=100
[avi @ 01E0699D86C0] use odml:1
st:1 removing common factor 36 from timebase
[avi @ 01E0699F8540] Before avformat_find_stream_info() pos: 4108 bytes 
read:142280 seeks:5 nb_streams:2
[mpeg4 @ 01E069A0AE00] Format yuv420p chosen by get_format().
[avi @ 01E0699F8540] All info found
[avi @ 01E0699F8540] After avformat_find_stream_info() pos: 16409 bytes 
read:142280 seeks:5 frames:21
Input #0, avi, from 'C:/Users/jcook/Desktop/sample.avi':
  Metadata:
encoder : MEncoder SVN-r33148-4.0.1
  Duration: 00:00:05.56, start: 0.00, bitrate: 540 kb/s
Stream #0:0, 1, 1/25: Video: mpeg4 (Simple Profile), 1 reference frame 
(XVID / 0x44495658), yuv420p(left), 320x240 [SAR 1:1 DAR 4:3], 0/1, 425 kb/s, 
25 fps, 25 tbr, 25 tbn, 25 tbc
Stream #0:1, 20, 32/1225: Audio: mp3 (U[0][0][0] / 0x0055), 44100 Hz, 
stereo, fltp, 100 kb/s
Successfully opened the file.
Parsing a group of options: output url output.mp4.
Applying option c:v (codec name) with argument mjpeg_qsv.
Successfully parsed a group of options.
Opening an output file: output.mp4.
[file @ 01E06B51B9C0] Setting default whitelist 'file,crypto'
Successfully opened the file.
detected 12 logical cores
Stream mapping:
  Stream #0:0 -> #0:0 (mpeg4 (native) -> mjpeg (mjpeg_qsv))
  Stream #0:1 -> #0:1 (mp3 (mp3float) -> aac (native))
Press [q] to stop, [?] for help
cur_dts is invalid st:0 (0) [init:0 i_done:0 finish:0] (this is harmless if it 
occurs once at the start per stream)
[graph_1_in_0_1 @ 01E069A12A40] Setting 'time_base' to value '1/44100'
[graph_1_in_0_1 @ 01E069A12A40] Setting 'sample_rate' to value '44100'
[graph_1_in_0_1 @ 01E069A12A40] Setting 'sample_fmt' to value 'fltp'
[graph_1_in_0_1 @ 01E069A12A40] Setting 'channel_layout' to value '0x3'
[graph_1_in_0_1 @ 01E069A12A40] tb:1/44100 samplefmt:fltp samplerate:44100 
chlayout:0x3
[format_out_0_1 @ 01E069A2F800] Setting 'sample_fmts' to value 'fltp'
[format_out_0_1 @ 01E069A2F800] Setting 'sample_rates' to value 
'96000|88200|64000|48000|44100|32000|24000|22050|16000|12000|11025|8000|7350'
[AVFilterGraph @ 01E069AA7100] query_formats: 4 queried, 9 merged, 0 
already done, 0 delayed
cur_dts is invalid st:0 (0) [init:0 i_done:0 finish:0] (this is harmless if it 
occurs once at the start per stream)
Last message repeated 20 times
[mpeg4 @ 01E069A2EE00] Format yuv420p chosen by get_format().
cur_dts is invalid st:0 (0) [init:0 i_done:0 finish:0] (this is harmless if it 
occurs once at the start per stream)
Last message repeated 28 times
[graph 0 input from stream 0:0 @ 01E069A31640] Setting 'video_size' to 
value '320x240'
[graph 0 input from stream 0:0 @ 01E069A31640] Setting 'pix_fmt' to value 

Re: [FFmpeg-devel] [PATCH] libavcodec/qsvenc.c: Set height and width alignment when using mjpeg_qsv

[Jon Cook]

Giving up on how to update the patch. Going to delete this ticket and
create a new one!

Jon

On Tue, 21 Jan 2020 at 12:30, Jon Cook  wrote:

> Updated patch with FrameInfo.Width and Height set as per review comments
>
> Jon
>
> On Tue, 21 Jan 2020 at 11:47, Jon Cook  wrote:
>
>> Hi Zhong,
>>
>> Thanks for your reply.
>>
>> > IMHO If you set the aligned value, FrameInfo.Width/Height shoule
>> > be changed to be:
>> >
>> >q->param.mfx.FrameInfo.Width  = FFALIGN(avctx->width,
>> q->width_align);
>> >q->param.mfx.FrameInfo.Height = FFALIGN(avctx->height,
>> q->height_align);
>>
>> Makes sense. I'll update the patch. This is my first time making changes
>> to the code so forgive any mistakes as I find my way around.
>>
>> > Could you help to provide detail steps (CLI and input video) to
>> > reproduce the issue?
>>
>> The command line I'm using is below along with the output without the
>> changes applied. Input video is not important, any video will do. There is
>> a little bit more about the problem here
>> http://ffmpeg.org/pipermail/ffmpeg-user/2019-November/046143.html
>>
>> Regards,
>> Jon
>>
>> $ ./ffmpeg.exe -loglevel debug -y -i C:/Users/jcook/Desktop/sample.avi
>> -c:v  mjpeg_qsv output.mp4
>> ffmpeg version N-95883-ga2fbdc6898 Copyright (c) 2000-2019 the FFmpeg
>> developers
>>   built with Microsoft (R) C/C++ Optimizing Compiler Version
>> 19.16.27031.1 for x64
>>   configuration: --toolchain=msvc --target-os=win64 --arch=x86_64
>> --prefix=/ffmpeg --enable-shared --disable-static --enable-yasm
>> --enable-asm --enable-libmfx --enable-nonfree
>>   libavutil  56. 36.101 / 56. 36.101
>>   libavcodec 58. 62.100 / 58. 62.100
>>   libavformat58. 35.100 / 58. 35.100
>>   libavdevice58.  9.101 / 58.  9.101
>>   libavfilter 7. 67.100 /  7. 67.100
>>   libswscale  5.  6.100 /  5.  6.100
>>   libswresample   3.  6.100 /  3.  6.100
>> Splitting the commandline.
>> Reading option '-loglevel' ... matched as option 'loglevel' (set logging
>> level) with argument 'debug'.
>> Reading option '-y' ... matched as option 'y' (overwrite output files)
>> with argument '1'.
>> Reading option '-i' ... matched as input url with argument
>> 'C:/Users/jcook/Desktop/sample.avi'.
>> Reading option '-c:v' ... matched as option 'c' (codec name) with
>> argument 'mjpeg_qsv'.
>> Reading option 'output.mp4' ... matched as output url.
>> Finished splitting the commandline.
>> Parsing a group of options: global .
>> Applying option loglevel (set logging level) with argument debug.
>> Applying option y (overwrite output files) with argument 1.
>> Successfully parsed a group of options.
>> Parsing a group of options: input url C:/Users/jcook/Desktop/sample.avi.
>> Successfully parsed a group of options.
>> Opening an input file: C:/Users/jcook/Desktop/sample.avi.
>> [NULL @ 01E0699F8540] Opening 'C:/Users/jcook/Desktop/sample.avi' for
>> reading
>> [file @ 01E0699D8280] Setting default whitelist 'file,crypto'
>> [avi @ 01E0699F8540] Format avi probed with size=2048 and score=100
>> [avi @ 01E0699D86C0] use odml:1
>> st:1 removing common factor 36 from timebase
>> [avi @ 01E0699F8540] Before avformat_find_stream_info() pos: 4108
>> bytes read:142280 seeks:5 nb_streams:2
>> [mpeg4 @ 01E069A0AE00] Format yuv420p chosen by get_format().
>> [avi @ 01E0699F8540] All info found
>> [avi @ 01E0699F8540] After avformat_find_stream_info() pos: 16409
>> bytes read:142280 seeks:5 frames:21
>> Input #0, avi, from 'C:/Users/jcook/Desktop/sample.avi':
>>   Metadata:
>> encoder : MEncoder SVN-r33148-4.0.1
>>   Duration: 00:00:05.56, start: 0.00, bitrate: 540 kb/s
>> Stream #0:0, 1, 1/25: Video: mpeg4 (Simple Profile), 1 reference
>> frame (XVID / 0x44495658), yuv420p(left), 320x240 [SAR 1:1 DAR 4:3], 0/1,
>> 425 kb/s, 25 fps, 25 tbr, 25 tbn, 25 tbc
>> Stream #0:1, 20, 32/1225: Audio: mp3 (U[0][0][0] / 0x0055), 44100 Hz,
>> stereo, fltp, 100 kb/s
>> Successfully opened the file.
>> Parsing a group of options: output url output.mp4.
>> Applying option c:v (codec name) with argument mjpeg_qsv.
>> Successfully parsed a group of options.
>> Opening an output file: output.mp4.
>> [file @ 01E06B51B9C0] Setting default whitelist 'file,crypto'
>> Successfully opened the file.
>> detected 12 logical cores
>> Stream mapping:
>>   Stream #0:0 -> #0:0 (mpeg4 (native) -> mjp

Re: [FFmpeg-devel] [PATCH] libavcodec/qsvenc.c: Set height and width alignment when using mjpeg_qsv

[Jon Cook]

Updated patch with FrameInfo.Width and Height set as per review comments

Jon

On Tue, 21 Jan 2020 at 11:47, Jon Cook  wrote:

> Hi Zhong,
>
> Thanks for your reply.
>
> > IMHO If you set the aligned value, FrameInfo.Width/Height shoule
> > be changed to be:
> >
> >q->param.mfx.FrameInfo.Width  = FFALIGN(avctx->width, q->width_align);
> >q->param.mfx.FrameInfo.Height = FFALIGN(avctx->height,
> q->height_align);
>
> Makes sense. I'll update the patch. This is my first time making changes
> to the code so forgive any mistakes as I find my way around.
>
> > Could you help to provide detail steps (CLI and input video) to
> > reproduce the issue?
>
> The command line I'm using is below along with the output without the
> changes applied. Input video is not important, any video will do. There is
> a little bit more about the problem here
> http://ffmpeg.org/pipermail/ffmpeg-user/2019-November/046143.html
>
> Regards,
> Jon
>
> $ ./ffmpeg.exe -loglevel debug -y -i C:/Users/jcook/Desktop/sample.avi
> -c:v  mjpeg_qsv output.mp4
> ffmpeg version N-95883-ga2fbdc6898 Copyright (c) 2000-2019 the FFmpeg
> developers
>   built with Microsoft (R) C/C++ Optimizing Compiler Version 19.16.27031.1
> for x64
>   configuration: --toolchain=msvc --target-os=win64 --arch=x86_64
> --prefix=/ffmpeg --enable-shared --disable-static --enable-yasm
> --enable-asm --enable-libmfx --enable-nonfree
>   libavutil  56. 36.101 / 56. 36.101
>   libavcodec 58. 62.100 / 58. 62.100
>   libavformat58. 35.100 / 58. 35.100
>   libavdevice58.  9.101 / 58.  9.101
>   libavfilter 7. 67.100 /  7. 67.100
>   libswscale  5.  6.100 /  5.  6.100
>   libswresample   3.  6.100 /  3.  6.100
> Splitting the commandline.
> Reading option '-loglevel' ... matched as option 'loglevel' (set logging
> level) with argument 'debug'.
> Reading option '-y' ... matched as option 'y' (overwrite output files)
> with argument '1'.
> Reading option '-i' ... matched as input url with argument
> 'C:/Users/jcook/Desktop/sample.avi'.
> Reading option '-c:v' ... matched as option 'c' (codec name) with argument
> 'mjpeg_qsv'.
> Reading option 'output.mp4' ... matched as output url.
> Finished splitting the commandline.
> Parsing a group of options: global .
> Applying option loglevel (set logging level) with argument debug.
> Applying option y (overwrite output files) with argument 1.
> Successfully parsed a group of options.
> Parsing a group of options: input url C:/Users/jcook/Desktop/sample.avi.
> Successfully parsed a group of options.
> Opening an input file: C:/Users/jcook/Desktop/sample.avi.
> [NULL @ 01E0699F8540] Opening 'C:/Users/jcook/Desktop/sample.avi' for
> reading
> [file @ 01E0699D8280] Setting default whitelist 'file,crypto'
> [avi @ 01E0699F8540] Format avi probed with size=2048 and score=100
> [avi @ 01E0699D86C0] use odml:1
> st:1 removing common factor 36 from timebase
> [avi @ 01E0699F8540] Before avformat_find_stream_info() pos: 4108
> bytes read:142280 seeks:5 nb_streams:2
> [mpeg4 @ 01E069A0AE00] Format yuv420p chosen by get_format().
> [avi @ 01E0699F8540] All info found
> [avi @ 01E0699F8540] After avformat_find_stream_info() pos: 16409
> bytes read:142280 seeks:5 frames:21
> Input #0, avi, from 'C:/Users/jcook/Desktop/sample.avi':
>   Metadata:
> encoder : MEncoder SVN-r33148-4.0.1
>   Duration: 00:00:05.56, start: 0.00, bitrate: 540 kb/s
> Stream #0:0, 1, 1/25: Video: mpeg4 (Simple Profile), 1 reference frame
> (XVID / 0x44495658), yuv420p(left), 320x240 [SAR 1:1 DAR 4:3], 0/1, 425
> kb/s, 25 fps, 25 tbr, 25 tbn, 25 tbc
> Stream #0:1, 20, 32/1225: Audio: mp3 (U[0][0][0] / 0x0055), 44100 Hz,
> stereo, fltp, 100 kb/s
> Successfully opened the file.
> Parsing a group of options: output url output.mp4.
> Applying option c:v (codec name) with argument mjpeg_qsv.
> Successfully parsed a group of options.
> Opening an output file: output.mp4.
> [file @ 01E06B51B9C0] Setting default whitelist 'file,crypto'
> Successfully opened the file.
> detected 12 logical cores
> Stream mapping:
>   Stream #0:0 -> #0:0 (mpeg4 (native) -> mjpeg (mjpeg_qsv))
>   Stream #0:1 -> #0:1 (mp3 (mp3float) -> aac (native))
> Press [q] to stop, [?] for help
> cur_dts is invalid st:0 (0) [init:0 i_done:0 finish:0] (this is harmless
> if it occurs once at the start per stream)
> [graph_1_in_0_1 @ 01E069A12A40] Setting 'time_base' to value '1/44100'
> [graph_1_in_0_1 @ 01E069A12A40] Setting 'sample_rate' to value '44100'
> [graph_1_in_0_1 @ 01E069A12A40] Setting 'sample_fmt' to value 'fltp'
> [graph_1_in_0_1 @ 01E069A12A40] Setting 'channe

[FFmpeg-devel] [PATCH] libavcodec/qsvenc.c: Set height and width alignment when using mjpeg_qsv

[Jon Cook]

From: JonCookCubic 

Setting FrameInfo.Width and Height as per review comments

Signed-off-by: JonCookCubic 
---
 libavcodec/qsvenc.c | 6 ++
 1 file changed, 6 insertions(+)

diff --git a/libavcodec/qsvenc.c b/libavcodec/qsvenc.c
index 9e416500e9..40904af388 100644
--- a/libavcodec/qsvenc.c
+++ b/libavcodec/qsvenc.c
@@ -470,6 +470,12 @@ static int init_video_param_jpeg(AVCodecContext *avctx, 
QSVEncContext *q)
 q->param.mfx.Quality  = av_clip(avctx->global_quality, 1, 100);
 q->param.mfx.RestartInterval  = 0;
 
+q->width_align = 16;
+q->height_align = 16;
+
+q->param.mfx.FrameInfo.Width = FFALIGN(avctx->width, q->width_align);
+q->param.mfx.FrameInfo.Height = FFALIGN(avctx->height, q->height_align);
+
 return 0;
 }
 
-- 
2.20.1.windows.1

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Re: [FFmpeg-devel] [PATCH] libavcodec/qsvenc.c: Set height and width alignment when using mjpeg_qsv

[Jon Cook]

Hi Zhong,

Thanks for your reply.

> IMHO If you set the aligned value, FrameInfo.Width/Height shoule
> be changed to be:
>
>q->param.mfx.FrameInfo.Width  = FFALIGN(avctx->width, q->width_align);
>q->param.mfx.FrameInfo.Height = FFALIGN(avctx->height,
q->height_align);

Makes sense. I'll update the patch. This is my first time making changes to
the code so forgive any mistakes as I find my way around.

> Could you help to provide detail steps (CLI and input video) to
> reproduce the issue?

The command line I'm using is below along with the output without the
changes applied. Input video is not important, any video will do. There is
a little bit more about the problem here
http://ffmpeg.org/pipermail/ffmpeg-user/2019-November/046143.html

Regards,
Jon

$ ./ffmpeg.exe -loglevel debug -y -i C:/Users/jcook/Desktop/sample.avi -c:v
 mjpeg_qsv output.mp4
ffmpeg version N-95883-ga2fbdc6898 Copyright (c) 2000-2019 the FFmpeg
developers
  built with Microsoft (R) C/C++ Optimizing Compiler Version 19.16.27031.1
for x64
  configuration: --toolchain=msvc --target-os=win64 --arch=x86_64
--prefix=/ffmpeg --enable-shared --disable-static --enable-yasm
--enable-asm --enable-libmfx --enable-nonfree
  libavutil  56. 36.101 / 56. 36.101
  libavcodec 58. 62.100 / 58. 62.100
  libavformat58. 35.100 / 58. 35.100
  libavdevice58.  9.101 / 58.  9.101
  libavfilter 7. 67.100 /  7. 67.100
  libswscale  5.  6.100 /  5.  6.100
  libswresample   3.  6.100 /  3.  6.100
Splitting the commandline.
Reading option '-loglevel' ... matched as option 'loglevel' (set logging
level) with argument 'debug'.
Reading option '-y' ... matched as option 'y' (overwrite output files) with
argument '1'.
Reading option '-i' ... matched as input url with argument
'C:/Users/jcook/Desktop/sample.avi'.
Reading option '-c:v' ... matched as option 'c' (codec name) with argument '
mjpeg_qsv'.
Reading option 'output.mp4' ... matched as output url.
Finished splitting the commandline.
Parsing a group of options: global .
Applying option loglevel (set logging level) with argument debug.
Applying option y (overwrite output files) with argument 1.
Successfully parsed a group of options.
Parsing a group of options: input url C:/Users/jcook/Desktop/sample.avi.
Successfully parsed a group of options.
Opening an input file: C:/Users/jcook/Desktop/sample.avi.
[NULL @ 01E0699F8540] Opening 'C:/Users/jcook/Desktop/sample.avi' for
reading
[file @ 01E0699D8280] Setting default whitelist 'file,crypto'
[avi @ 01E0699F8540] Format avi probed with size=2048 and score=100
[avi @ 01E0699D86C0] use odml:1
st:1 removing common factor 36 from timebase
[avi @ 01E0699F8540] Before avformat_find_stream_info() pos: 4108 bytes
read:142280 seeks:5 nb_streams:2
[mpeg4 @ 01E069A0AE00] Format yuv420p chosen by get_format().
[avi @ 01E0699F8540] All info found
[avi @ 01E0699F8540] After avformat_find_stream_info() pos: 16409 bytes
read:142280 seeks:5 frames:21
Input #0, avi, from 'C:/Users/jcook/Desktop/sample.avi':
  Metadata:
encoder : MEncoder SVN-r33148-4.0.1
  Duration: 00:00:05.56, start: 0.00, bitrate: 540 kb/s
Stream #0:0, 1, 1/25: Video: mpeg4 (Simple Profile), 1 reference frame
(XVID / 0x44495658), yuv420p(left), 320x240 [SAR 1:1 DAR 4:3], 0/1, 425
kb/s, 25 fps, 25 tbr, 25 tbn, 25 tbc
Stream #0:1, 20, 32/1225: Audio: mp3 (U[0][0][0] / 0x0055), 44100 Hz,
stereo, fltp, 100 kb/s
Successfully opened the file.
Parsing a group of options: output url output.mp4.
Applying option c:v (codec name) with argument mjpeg_qsv.
Successfully parsed a group of options.
Opening an output file: output.mp4.
[file @ 01E06B51B9C0] Setting default whitelist 'file,crypto'
Successfully opened the file.
detected 12 logical cores
Stream mapping:
  Stream #0:0 -> #0:0 (mpeg4 (native) -> mjpeg (mjpeg_qsv))
  Stream #0:1 -> #0:1 (mp3 (mp3float) -> aac (native))
Press [q] to stop, [?] for help
cur_dts is invalid st:0 (0) [init:0 i_done:0 finish:0] (this is harmless if
it occurs once at the start per stream)
[graph_1_in_0_1 @ 01E069A12A40] Setting 'time_base' to value '1/44100'
[graph_1_in_0_1 @ 01E069A12A40] Setting 'sample_rate' to value '44100'
[graph_1_in_0_1 @ 01E069A12A40] Setting 'sample_fmt' to value 'fltp'
[graph_1_in_0_1 @ 01E069A12A40] Setting 'channel_layout' to value '0x3'
[graph_1_in_0_1 @ 01E069A12A40] tb:1/44100 samplefmt:fltp
samplerate:44100 chlayout:0x3
[format_out_0_1 @ 01E069A2F800] Setting 'sample_fmts' to value 'fltp'
[format_out_0_1 @ 01E069A2F800] Setting 'sample_rates' to value
'96000|88200|64000|48000|44100|32000|24000|22050|16000|12000|11025|8000|7350'
[AVFilterGraph @ 01E069AA7100] query_formats: 4 queried, 9 merged, 0
already done, 0 delayed
cur_dts is invalid st:0 (0) [init:0 i_done:0 finish:0] (this is harmless if
it occurs once at the start per stream)
Last message repeated 20 times
[mpeg4 @ 01E069A2E

[FFmpeg-devel] [PATCH] libavcodec/qsvenc.c: Set height and width alignment when using mjpeg_qsv

[Jon Cook]

From: JonCookCubic 

Currently width_align and height_align are zero which cases "Error submitting 
the frame for encoding". This change sets the alignments.

Signed-off-by: Jon Cook 
---
 libavcodec/qsvenc.c | 3 +++
 1 file changed, 3 insertions(+)

diff --git a/libavcodec/qsvenc.c b/libavcodec/qsvenc.c
index 9e416500e9..b18393532f 100644
--- a/libavcodec/qsvenc.c
+++ b/libavcodec/qsvenc.c
@@ -470,6 +470,9 @@ static int init_video_param_jpeg(AVCodecContext *avctx, 
QSVEncContext *q)
 q->param.mfx.Quality  = av_clip(avctx->global_quality, 1, 100);
 q->param.mfx.RestartInterval  = 0;
 
+q->width_align = 16;
+q->height_align = 16;
+
 return 0;
 }
 
-- 
2.20.1.windows.1

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[FFmpeg-devel] [PATCH] libavcodec/qsvenc.c: Set height and width alignment when using mjpeg_qsv

[Jon Cook]

From: JonCookCubic 

Currently width_align and height_align are zero which cases "Error submitting 
the frame for encoding". This change sets the alignments.

Signed-off-by: Jon Cook 
---
 libavcodec/qsvenc.c | 3 +++
 1 file changed, 3 insertions(+)

diff --git a/libavcodec/qsvenc.c b/libavcodec/qsvenc.c
index 9e416500e9..b18393532f 100644
--- a/libavcodec/qsvenc.c
+++ b/libavcodec/qsvenc.c
@@ -470,6 +470,9 @@ static int init_video_param_jpeg(AVCodecContext *avctx, 
QSVEncContext *q)
 q->param.mfx.Quality  = av_clip(avctx->global_quality, 1, 100);
 q->param.mfx.RestartInterval  = 0;
 
+q->width_align = 16;
+q->height_align = 16;
+
 return 0;
 }
 
-- 
2.20.1.windows.1

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[FFmpeg-devel] Patch vf_blackdetect

[Jon bae]

This is now the working version from my last patch.

Description:
This patch unify vf_blackdetect with af_silencedetect. Now the logging
prints black_start and black_end in separate lines. This is the same
behavior like af_silencedetect and it is also more useful for monitoring
streams.

Is it ok, as attachment?


0001-unify-blackdetect-with-af_silencedetect.patch
Description: Binary data
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[FFmpeg-devel] Last Patch: vf_blackdetect

[Jon bae]

Please forget the last patch! Something is not working right.
Sorry!
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[FFmpeg-devel] Patch vf_blackdetect

[Jon bae]

This patch unify vf_blackdetect with af_silencedetect. Now the logging
prints black_start and black_end in separate lines. This is the same
behavior like af_silencedetect and it is also more useful for monitoring
streams.


0001-unify-blackdetect-with-af_silencedetect.-Is-more-use.patch
Description: Binary data
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[FFmpeg-devel] [PATCH 0/2] Fix gapless encoding/remuxing for MP3

[Jon Toohill]

Round trip wav->mp3->wav now preserves the correct number of samples.
Remuxing mp3->mp3 with -c:a copy also preserves any existing gapless
metadata in the Info tag.

The code in libmp3lame.c to set AV_PKT_DATA_SKIP_SAMPLES was mostly
copied from libopusenc.c.

Jon Toohill (2):
  lavc/libmp3lame: send encoder delay/padding in packet side data
  lavf/mp3enc: write encoder delay/padding upon closing

 libavcodec/libmp3lame.c | 27 ++-
 libavformat/mp3enc.c| 34 --
 2 files changed, 54 insertions(+), 7 deletions(-)

-- 
2.8.0.rc3.226.g39d4020

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[FFmpeg-devel] [PATCH 1/2] lavc/libmp3lame: send encoder delay/padding in packet side data

[Jon Toohill]

---
 libavcodec/libmp3lame.c | 27 ++-
 1 file changed, 26 insertions(+), 1 deletion(-)

diff --git a/libavcodec/libmp3lame.c b/libavcodec/libmp3lame.c
index 5642264..e55aa85 100644
--- a/libavcodec/libmp3lame.c
+++ b/libavcodec/libmp3lame.c
@@ -50,6 +50,7 @@ typedef struct LAMEContext {
 int reservoir;
 int joint_stereo;
 int abr;
+int delay_sent;
 float *samples_flt[2];
 AudioFrameQueue afq;
 AVFloatDSPContext *fdsp;
@@ -185,7 +186,7 @@ static int mp3lame_encode_frame(AVCodecContext *avctx, 
AVPacket *avpkt,
 {
 LAMEContext *s = avctx->priv_data;
 MPADecodeHeader hdr;
-int len, ret, ch;
+int len, ret, ch, discard_padding;
 int lame_result;
 uint32_t h;
 
@@ -269,6 +270,30 @@ static int mp3lame_encode_frame(AVCodecContext *avctx, 
AVPacket *avpkt,
 ff_af_queue_remove(>afq, avctx->frame_size, >pts,
>duration);
 
+discard_padding = avctx->frame_size - avpkt->duration;
+// Check if subtraction resulted in an overflow
+if ((discard_padding < avctx->frame_size) != (avpkt->duration > 0)) {
+av_log(avctx, AV_LOG_ERROR, "discard padding overflow\n");
+av_packet_unref(avpkt);
+av_free(avpkt);
+return AVERROR(EINVAL);
+}
+if ((!s->delay_sent && avctx->initial_padding > 0) || discard_padding 
> 0) {
+uint8_t* side_data = av_packet_new_side_data(avpkt,
+ 
AV_PKT_DATA_SKIP_SAMPLES,
+ 10);
+if(!side_data) {
+av_packet_unref(avpkt);
+av_free(avpkt);
+return AVERROR(ENOMEM);
+}
+if (!s->delay_sent) {
+AV_WL32(side_data, avctx->initial_padding);
+s->delay_sent = 1;
+}
+AV_WL32(side_data + 4, discard_padding);
+}
+
 avpkt->size = len;
 *got_packet_ptr = 1;
 }
-- 
2.8.0.rc3.226.g39d4020

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[FFmpeg-devel] [PATCH 2/2] lavf/mp3enc: write encoder delay/padding upon closing

[Jon Toohill]

---
 libavformat/mp3enc.c | 34 --
 1 file changed, 28 insertions(+), 6 deletions(-)

diff --git a/libavformat/mp3enc.c b/libavformat/mp3enc.c
index de63401..4c97fa1 100644
--- a/libavformat/mp3enc.c
+++ b/libavformat/mp3enc.c
@@ -111,6 +111,8 @@ typedef struct MP3Context {
 uint64_t bag[XING_NUM_BAGS];
 int initial_bitrate;
 int has_variable_bitrate;
+int delay;
+int padding;
 
 /* index of the audio stream */
 int audio_stream_idx;
@@ -247,12 +249,7 @@ static int mp3_write_xing(AVFormatContext *s)
 ffio_fill(dyn_ctx, 0, 8); // empty replaygain fields
 avio_w8(dyn_ctx, 0);  // unknown encoding flags
 avio_w8(dyn_ctx, 0);  // unknown abr/minimal bitrate
-
-// encoder delay
-if (par->initial_padding - 528 - 1 >= 1 << 12) {
-av_log(s, AV_LOG_WARNING, "Too many samples of initial padding.\n");
-}
-avio_wb24(dyn_ctx, FFMAX(par->initial_padding - 528 - 1, 0)<<12);
+avio_wb24(dyn_ctx, 0);// empty encoder delay/padding
 
 avio_w8(dyn_ctx,   0); // misc
 avio_w8(dyn_ctx,   0); // mp3gain
@@ -345,10 +342,24 @@ static int mp3_write_audio_packet(AVFormatContext *s, 
AVPacket *pkt)
 #endif
 
 if (mp3->xing_offset) {
+uint8_t *side_data = NULL;
+int side_data_size = 0;
+
 mp3_xing_add_frame(mp3, pkt);
 mp3->audio_size += pkt->size;
 mp3->audio_crc   = av_crc(av_crc_get_table(AV_CRC_16_ANSI_LE),
   mp3->audio_crc, pkt->data, pkt->size);
+
+side_data = av_packet_get_side_data(pkt,
+AV_PKT_DATA_SKIP_SAMPLES,
+_data_size);
+if (side_data && side_data_size >= 10) {
+mp3->padding = FFMAX(AV_RL32(side_data + 4) + 528 + 1, 0);
+if (!mp3->delay)
+mp3->delay =  FFMAX(AV_RL32(side_data) - 528 - 1, 0);
+} else {
+mp3->padding = 0;
+}
 }
 }
 
@@ -422,6 +433,17 @@ static void mp3_update_xing(AVFormatContext *s)
 }
 }
 
+/* write encoder delay/padding */
+if (mp3->delay >= 1 << 12) {
+mp3->delay = (1 << 12) - 1;
+av_log(s, AV_LOG_WARNING, "Too many samples of initial padding.\n");
+}
+if (mp3->padding >= 1 << 12) {
+mp3->padding = (1 << 12) - 1;
+av_log(s, AV_LOG_WARNING, "Too many samples of trailing padding.\n");
+}
+AV_WB24(mp3->xing_frame + mp3->xing_offset + 141, (mp3->delay << 12) + 
mp3->padding);
+
 AV_WB32(mp3->xing_frame + mp3->xing_offset + XING_SIZE - 8, 
mp3->audio_size);
 AV_WB16(mp3->xing_frame + mp3->xing_offset + XING_SIZE - 4, 
mp3->audio_crc);
 
-- 
2.8.0.rc3.226.g39d4020

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[FFmpeg-devel] dh option in nnedi3

[Jon bae]

Hello All,
sorry for that question - I know is open source and we can add option by
ours self, but my knowledge is to limited for that.

Is it possible to integrate the dh (double high) option from vapoursynth
nnedi3 filter to the ffmpeg nnedi filter?

I have start copy parts from the sources, but I hang from line 709 on in
the ffmpeg version.

Regards

Jonathan
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[FFmpeg-devel] [PATCH 2/2] lavf/mp3enc: write encoder delay/padding upon closing

[Jon Toohill]

---
 libavformat/mp3enc.c | 34 +++---
 1 file changed, 27 insertions(+), 7 deletions(-)

diff --git a/libavformat/mp3enc.c b/libavformat/mp3enc.c
index de63401..ddf4b93 100644
--- a/libavformat/mp3enc.c
+++ b/libavformat/mp3enc.c
@@ -111,6 +111,8 @@ typedef struct MP3Context {
 uint64_t bag[XING_NUM_BAGS];
 int initial_bitrate;
 int has_variable_bitrate;
+int delay;
+int padding;
 
 /* index of the audio stream */
 int audio_stream_idx;
@@ -247,12 +249,7 @@ static int mp3_write_xing(AVFormatContext *s)
 ffio_fill(dyn_ctx, 0, 8); // empty replaygain fields
 avio_w8(dyn_ctx, 0);  // unknown encoding flags
 avio_w8(dyn_ctx, 0);  // unknown abr/minimal bitrate
-
-// encoder delay
-if (par->initial_padding - 528 - 1 >= 1 << 12) {
-av_log(s, AV_LOG_WARNING, "Too many samples of initial padding.\n");
-}
-avio_wb24(dyn_ctx, FFMAX(par->initial_padding - 528 - 1, 0)<<12);
+avio_wb24(dyn_ctx, 0);// empty encoder delay/padding
 
 avio_w8(dyn_ctx,   0); // misc
 avio_w8(dyn_ctx,   0); // mp3gain
@@ -345,10 +342,24 @@ static int mp3_write_audio_packet(AVFormatContext *s, 
AVPacket *pkt)
 #endif
 
 if (mp3->xing_offset) {
+uint8_t *side_data = NULL;
+int side_data_size = 0;
+
 mp3_xing_add_frame(mp3, pkt);
 mp3->audio_size += pkt->size;
 mp3->audio_crc   = av_crc(av_crc_get_table(AV_CRC_16_ANSI_LE),
   mp3->audio_crc, pkt->data, pkt->size);
+
+side_data = av_packet_get_side_data(pkt,
+AV_PKT_DATA_SKIP_SAMPLES,
+_data_size);
+if (side_data && side_data_size >= 10) {
+mp3->padding = AV_RL32(side_data + 4) + 528 + 1;
+if (!mp3->delay)
+mp3->delay =  FFMAX(AV_RL32(side_data) - 528 - 1, 0);
+} else {
+mp3->padding = 0;
+}
 }
 }
 
@@ -381,7 +392,7 @@ static void mp3_update_xing(AVFormatContext *s)
 AVReplayGain *rg;
 uint16_t tag_crc;
 uint8_t *toc;
-int i, rg_size;
+int i, rg_size, delay;
 
 /* replace "Xing" identification string with "Info" for CBR files. */
 if (!mp3->has_variable_bitrate)
@@ -422,6 +433,15 @@ static void mp3_update_xing(AVFormatContext *s)
 }
 }
 
+/* write encoder delay/padding */
+if (mp3->delay >= 1 << 12) {
+av_log(s, AV_LOG_WARNING, "Too many samples of initial padding.\n");
+}
+if (mp3->padding >= 1 << 12) {
+av_log(s, AV_LOG_WARNING, "Too many samples of trailing padding.\n");
+}
+AV_WB24(mp3->xing_frame + mp3->xing_offset + 141, (mp3->delay << 12) + 
mp3->padding);
+
 AV_WB32(mp3->xing_frame + mp3->xing_offset + XING_SIZE - 8, 
mp3->audio_size);
 AV_WB16(mp3->xing_frame + mp3->xing_offset + XING_SIZE - 4, 
mp3->audio_crc);
 
-- 
2.8.0.rc3.226.g39d4020

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[FFmpeg-devel] [PATCH 0/2] Fix gapless encoding/remuxing for MP3

[Jon Toohill]

Round trip wav->mp3->wav now preserves the correct number of samples.
Remuxing mp3->mp3 with -c:a copy also preserves any existing gapless
metadata in the Info tag.

The code in libmp3lame.c to set AV_PKT_DATA_SKIP_SAMPLES was mostly
copied from libopusenc.c.

Jon Toohill (2):
  lavc/libmp3lame: send encoder delay/padding in packet side data
  lavf/mp3enc: write encoder delay/padding upon closing

 libavcodec/libmp3lame.c | 26 +-
 libavformat/mp3enc.c| 34 +++---
 2 files changed, 52 insertions(+), 8 deletions(-)

-- 
2.8.0.rc3.226.g39d4020

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[FFmpeg-devel] [PATCH 1/2] lavc/libmp3lame: send encoder delay/padding in packet side data

[Jon Toohill]

---
 libavcodec/libmp3lame.c | 26 +-
 1 file changed, 25 insertions(+), 1 deletion(-)

diff --git a/libavcodec/libmp3lame.c b/libavcodec/libmp3lame.c
index 5642264..b3ba0d8 100644
--- a/libavcodec/libmp3lame.c
+++ b/libavcodec/libmp3lame.c
@@ -50,6 +50,7 @@ typedef struct LAMEContext {
 int reservoir;
 int joint_stereo;
 int abr;
+int delay_sent;
 float *samples_flt[2];
 AudioFrameQueue afq;
 AVFloatDSPContext *fdsp;
@@ -185,7 +186,7 @@ static int mp3lame_encode_frame(AVCodecContext *avctx, 
AVPacket *avpkt,
 {
 LAMEContext *s = avctx->priv_data;
 MPADecodeHeader hdr;
-int len, ret, ch;
+int len, ret, ch, discard_padding;
 int lame_result;
 uint32_t h;
 
@@ -269,6 +270,29 @@ static int mp3lame_encode_frame(AVCodecContext *avctx, 
AVPacket *avpkt,
 ff_af_queue_remove(>afq, avctx->frame_size, >pts,
>duration);
 
+discard_padding = avctx->frame_size - avpkt->duration;
+// Check if subtraction resulted in an overflow
+if ((discard_padding < avctx->frame_size) != (avpkt->duration > 0)) {
+av_packet_unref(avpkt);
+av_free(avpkt);
+return AVERROR(EINVAL);
+}
+if ((!s->delay_sent && avctx->initial_padding > 0) || discard_padding 
> 0) {
+uint8_t* side_data = av_packet_new_side_data(avpkt,
+ 
AV_PKT_DATA_SKIP_SAMPLES,
+ 10);
+if(!side_data) {
+av_packet_unref(avpkt);
+av_free(avpkt);
+return AVERROR(ENOMEM);
+}
+if (!s->delay_sent) {
+AV_WL32(side_data, avctx->initial_padding);
+s->delay_sent = 1;
+}
+AV_WL32(side_data + 4, discard_padding);
+}
+
 avpkt->size = len;
 *got_packet_ptr = 1;
 }
-- 
2.8.0.rc3.226.g39d4020

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Re: [FFmpeg-devel] [PATCH 3/3] lavf/mp3dec: read encoder delay/padding from Info tag

[Jon Toohill]

On Wed, Oct 5, 2016 at 11:03 AM, wm4 <nfx...@googlemail.com> wrote:

> On Wed, 5 Oct 2016 10:42:13 -0700
> Jon Toohill <jtoohill-at-google@ffmpeg.org> wrote:
>
> > On Wed, Oct 5, 2016 at 10:40 AM, Jon Toohill <jtooh...@google.com>
> wrote:
> >
> > > On Tue, Oct 4, 2016 at 7:19 AM, wm4 <nfx...@googlemail.com> wrote:
> > >
> > >> On Mon,  3 Oct 2016 17:45:08 -0700
> > >> Jon Toohill <jtoohill-at-google@ffmpeg.org> wrote:
> > >>
> > >> > Muxers can check AVCodecParameters.initial_padding for the
> > >> > encoder+decoder delay, and read the AV_PKT_DATA_SKIP_SAMPLES
> > >> > side data from the last packet for the encoder padding.
> > >> >
> > >> > This change also fixes the first_discard_sample calculation
> > >> > which erroneously included the decoder delay. Decoder delay
> > >> > is already accounted for in st->skip_samples. The affected
> > >> > FATE tests have been updated accordingly.
> > >> > ---
> > >> >  libavformat/mp3dec.c |  3 ++-
> > >> >  tests/ref/fate/audiomatch-square-mp3 |  2 +-
> > >> >  tests/ref/fate/gapless-mp3   | 10 +-
> > >> >  3 files changed, 8 insertions(+), 7 deletions(-)
> > >> >
> > >> > diff --git a/libavformat/mp3dec.c b/libavformat/mp3dec.c
> > >> > index 56c7f8c..e8b2428 100644
> > >> > --- a/libavformat/mp3dec.c
> > >> > +++ b/libavformat/mp3dec.c
> > >> > @@ -239,9 +239,10 @@ static void mp3_parse_info_tag(AVFormatContext
> > >> *s, AVStream *st,
> > >> >
> > >> >  mp3->start_pad = v>>12;
> > >> >  mp3->  end_pad = v&4095;
> > >> > +st->codecpar->initial_padding = mp3->start_pad + 528 + 1;
> > >> >  st->start_skip_samples = mp3->start_pad + 528 + 1;
> > >> >  if (mp3->frames) {
> > >> > -st->first_discard_sample = -mp3->end_pad + 528 + 1 +
> > >> mp3->frames * (int64_t)spf;
> > >> > +st->first_discard_sample = -mp3->end_pad + mp3->frames
> *
> > >> (int64_t)spf;
> > >>
> > >> How does mixing these even make sense?
> > >>
> > >
> > > mp3enc.c already uses initial_padding for the encoder delay, and as you
> > > previously pointed out, mp3dec.c already uses
> AV_PKT_START_SKIP_SAMPLES for
> > > the encoder delay. I'm not attempting to solve the inconsistency in
> this
> > > patch set.
> > >
> >
> > err, *mp3dec.c already uses AV_PKT_DATA_SKIP_SAMPLES for the encoder
> > padding. Sorry for the confusion.
> >
>
> Not solving the inconsistency is problematic - but the worse issue is
> that you seem to introduce inconsistencies. In my current opinion, the
> packet side data and the initial_padding fields do the same (i.e.
> duplicated API), so only one of them can or should be used. Your patch
> seems to require the decoder to use them both, though.
>

That's a fair point; I'll send a revised patch set that doesn't change
mp3dec.c.

Note that I don't think I can get away from having libmp3lame.c set
AVCodecContext.initial_padding, since that field is used by the
AudioFrameQueue (and other encoders rely on it as well).

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Re: [FFmpeg-devel] [PATCH 3/3] lavf/mp3dec: read encoder delay/padding from Info tag

[Jon Toohill]

On Tue, Oct 4, 2016 at 9:10 AM, Michael Niedermayer <mich...@niedermayer.cc>
wrote:

> On Mon, Oct 03, 2016 at 05:45:08PM -0700, Jon Toohill wrote:
> > Muxers can check AVCodecParameters.initial_padding for the
> > encoder+decoder delay, and read the AV_PKT_DATA_SKIP_SAMPLES
> > side data from the last packet for the encoder padding.
> >
> > This change also fixes the first_discard_sample calculation
> > which erroneously included the decoder delay. Decoder delay
> > is already accounted for in st->skip_samples. The affected
> > FATE tests have been updated accordingly.
> > ---
> >  libavformat/mp3dec.c |  3 ++-
> >  tests/ref/fate/audiomatch-square-mp3 |  2 +-
> >  tests/ref/fate/gapless-mp3   | 10 +-
> >  3 files changed, 8 insertions(+), 7 deletions(-)
> >
> > diff --git a/libavformat/mp3dec.c b/libavformat/mp3dec.c
> > index 56c7f8c..e8b2428 100644
> > --- a/libavformat/mp3dec.c
> > +++ b/libavformat/mp3dec.c
> > @@ -239,9 +239,10 @@ static void mp3_parse_info_tag(AVFormatContext *s,
> AVStream *st,
> >
> >  mp3->start_pad = v>>12;
> >  mp3->  end_pad = v&4095;
> > +st->codecpar->initial_padding = mp3->start_pad + 528 + 1;
> >  st->start_skip_samples = mp3->start_pad + 528 + 1;
> >  if (mp3->frames) {
> > -st->first_discard_sample = -mp3->end_pad + 528 + 1 +
> mp3->frames * (int64_t)spf;
> > +st->first_discard_sample = -mp3->end_pad + mp3->frames *
> (int64_t)spf;
> >  st->last_discard_sample = mp3->frames * (int64_t)spf;
> >  }
> >  if (!st->start_time)
> > diff --git a/tests/ref/fate/audiomatch-square-mp3
> b/tests/ref/fate/audiomatch-square-mp3
> > index 8de55c2..05176a0 100644
> > --- a/tests/ref/fate/audiomatch-square-mp3
> > +++ b/tests/ref/fate/audiomatch-square-mp3
> > @@ -1 +1 @@
> > -presig: 0 postsig:0 c: 0.9447 lenerr:0
> > +presig: 0 postsig:-529 c: 0.9334 lenerr:-529
>
> isnt this exactly correct before and wrong after this change ?
>
> zero signal before and zero signal after the original is what one would
> expect and equal length and high correlation
>
> every number that changes gets worse
>

Michael - you're right, this patch is incorrect currently. I had mistakenly
convinced myself that mp3dec.c was overcompensating for the decoder delay.

I also found that LAME itself writes the encoder padding + decoder delay
into the trailing padding field (e.g. for an input where 468 samples of
padding are added by the encoder, the Info tag contains the value 997).
I'll send a revised patch set.


> [...]
>
> --
> Michael GnuPG fingerprint: 9FF2128B147EF6730BADF133611EC787040B0FAB
>
> Never trust a computer, one day, it may think you are the virus. -- Compn
>
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Re: [FFmpeg-devel] [PATCH 3/3] lavf/mp3dec: read encoder delay/padding from Info tag

[Jon Toohill]

On Tue, Oct 4, 2016 at 7:19 AM, wm4 <nfx...@googlemail.com> wrote:

> On Mon,  3 Oct 2016 17:45:08 -0700
> Jon Toohill <jtoohill-at-google@ffmpeg.org> wrote:
>
> > Muxers can check AVCodecParameters.initial_padding for the
> > encoder+decoder delay, and read the AV_PKT_DATA_SKIP_SAMPLES
> > side data from the last packet for the encoder padding.
> >
> > This change also fixes the first_discard_sample calculation
> > which erroneously included the decoder delay. Decoder delay
> > is already accounted for in st->skip_samples. The affected
> > FATE tests have been updated accordingly.
> > ---
> >  libavformat/mp3dec.c |  3 ++-
> >  tests/ref/fate/audiomatch-square-mp3 |  2 +-
> >  tests/ref/fate/gapless-mp3   | 10 +-
> >  3 files changed, 8 insertions(+), 7 deletions(-)
> >
> > diff --git a/libavformat/mp3dec.c b/libavformat/mp3dec.c
> > index 56c7f8c..e8b2428 100644
> > --- a/libavformat/mp3dec.c
> > +++ b/libavformat/mp3dec.c
> > @@ -239,9 +239,10 @@ static void mp3_parse_info_tag(AVFormatContext *s,
> AVStream *st,
> >
> >  mp3->start_pad = v>>12;
> >  mp3->  end_pad = v&4095;
> > +st->codecpar->initial_padding = mp3->start_pad + 528 + 1;
> >  st->start_skip_samples = mp3->start_pad + 528 + 1;
> >  if (mp3->frames) {
> > -st->first_discard_sample = -mp3->end_pad + 528 + 1 +
> mp3->frames * (int64_t)spf;
> > +st->first_discard_sample = -mp3->end_pad + mp3->frames *
> (int64_t)spf;
>
> How does mixing these even make sense?
>

mp3enc.c already uses initial_padding for the encoder delay, and as you
previously pointed out, mp3dec.c already uses AV_PKT_START_SKIP_SAMPLES for
the encoder delay. I'm not attempting to solve the inconsistency in this
patch set.


> >  st->last_discard_sample = mp3->frames * (int64_t)spf;
> >  }
> >  if (!st->start_time)
> > diff --git a/tests/ref/fate/audiomatch-square-mp3
> b/tests/ref/fate/audiomatch-square-mp3
> > index 8de55c2..05176a0 100644
> > --- a/tests/ref/fate/audiomatch-square-mp3
> > +++ b/tests/ref/fate/audiomatch-square-mp3
> > @@ -1 +1 @@
> > -presig: 0 postsig:0 c: 0.9447 lenerr:0
> > +presig: 0 postsig:-529 c: 0.9334 lenerr:-529
> > diff --git a/tests/ref/fate/gapless-mp3 b/tests/ref/fate/gapless-mp3
> > index ebe7bfa..8b80bfc 100644
> > --- a/tests/ref/fate/gapless-mp3
> > +++ b/tests/ref/fate/gapless-mp3
> > @@ -1,5 +1,5 @@
> > -37534a3bcc3ef306e8c5ebfcfedfc41c *tests/data/fate/gapless-mp3.out-1
> > -c96c3ae7bd3300fd2f4debac222de5b7
> > -0cd1cdbcfd5cdbf6270cd98219bf31cd *tests/data/fate/gapless-mp3.out-2
> > -c96c3ae7bd3300fd2f4debac222de5b7
> > -9d3d8ba8a61b534f2d02ee648d6a8229 *tests/data/fate/gapless-mp3.out-3
> > +81695be427d45e8be4d527a6b2af2a85 *tests/data/fate/gapless-mp3.out-1
> > +c7879a827ab017364774069268d9a267
> > +62d074296f8c84a5f86a6afdd7bab459 *tests/data/fate/gapless-mp3.out-2
> > +c7879a827ab017364774069268d9a267
> > +e931f3fe1ba25e0d5eece4977c4061a9 *tests/data/fate/gapless-mp3.out-3
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[FFmpeg-devel] [PATCH 3/3] lavf/mp3dec: read encoder delay/padding from Info tag

[Jon Toohill]

Muxers can check AVCodecParameters.initial_padding for the
encoder+decoder delay, and read the AV_PKT_DATA_SKIP_SAMPLES
side data from the last packet for the encoder padding.

This change also fixes the first_discard_sample calculation
which erroneously included the decoder delay. Decoder delay
is already accounted for in st->skip_samples. The affected
FATE tests have been updated accordingly.
---
 libavformat/mp3dec.c |  3 ++-
 tests/ref/fate/audiomatch-square-mp3 |  2 +-
 tests/ref/fate/gapless-mp3   | 10 +-
 3 files changed, 8 insertions(+), 7 deletions(-)

diff --git a/libavformat/mp3dec.c b/libavformat/mp3dec.c
index 56c7f8c..e8b2428 100644
--- a/libavformat/mp3dec.c
+++ b/libavformat/mp3dec.c
@@ -239,9 +239,10 @@ static void mp3_parse_info_tag(AVFormatContext *s, 
AVStream *st,
 
 mp3->start_pad = v>>12;
 mp3->  end_pad = v&4095;
+st->codecpar->initial_padding = mp3->start_pad + 528 + 1;
 st->start_skip_samples = mp3->start_pad + 528 + 1;
 if (mp3->frames) {
-st->first_discard_sample = -mp3->end_pad + 528 + 1 + mp3->frames * 
(int64_t)spf;
+st->first_discard_sample = -mp3->end_pad + mp3->frames * 
(int64_t)spf;
 st->last_discard_sample = mp3->frames * (int64_t)spf;
 }
 if (!st->start_time)
diff --git a/tests/ref/fate/audiomatch-square-mp3 
b/tests/ref/fate/audiomatch-square-mp3
index 8de55c2..05176a0 100644
--- a/tests/ref/fate/audiomatch-square-mp3
+++ b/tests/ref/fate/audiomatch-square-mp3
@@ -1 +1 @@
-presig: 0 postsig:0 c: 0.9447 lenerr:0
+presig: 0 postsig:-529 c: 0.9334 lenerr:-529
diff --git a/tests/ref/fate/gapless-mp3 b/tests/ref/fate/gapless-mp3
index ebe7bfa..8b80bfc 100644
--- a/tests/ref/fate/gapless-mp3
+++ b/tests/ref/fate/gapless-mp3
@@ -1,5 +1,5 @@
-37534a3bcc3ef306e8c5ebfcfedfc41c *tests/data/fate/gapless-mp3.out-1
-c96c3ae7bd3300fd2f4debac222de5b7
-0cd1cdbcfd5cdbf6270cd98219bf31cd *tests/data/fate/gapless-mp3.out-2
-c96c3ae7bd3300fd2f4debac222de5b7
-9d3d8ba8a61b534f2d02ee648d6a8229 *tests/data/fate/gapless-mp3.out-3
+81695be427d45e8be4d527a6b2af2a85 *tests/data/fate/gapless-mp3.out-1
+c7879a827ab017364774069268d9a267
+62d074296f8c84a5f86a6afdd7bab459 *tests/data/fate/gapless-mp3.out-2
+c7879a827ab017364774069268d9a267
+e931f3fe1ba25e0d5eece4977c4061a9 *tests/data/fate/gapless-mp3.out-3
-- 
2.8.0.rc3.226.g39d4020

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[FFmpeg-devel] [PATCH 1/3] lavc/libmp3lame: send encoder padding in side data of final packet

[Jon Toohill]

---
 libavcodec/libmp3lame.c | 21 -
 1 file changed, 20 insertions(+), 1 deletion(-)

diff --git a/libavcodec/libmp3lame.c b/libavcodec/libmp3lame.c
index 5642264..a1bf122 100644
--- a/libavcodec/libmp3lame.c
+++ b/libavcodec/libmp3lame.c
@@ -185,7 +185,7 @@ static int mp3lame_encode_frame(AVCodecContext *avctx, 
AVPacket *avpkt,
 {
 LAMEContext *s = avctx->priv_data;
 MPADecodeHeader hdr;
-int len, ret, ch;
+int len, ret, ch, discard_padding;
 int lame_result;
 uint32_t h;
 
@@ -269,6 +269,25 @@ static int mp3lame_encode_frame(AVCodecContext *avctx, 
AVPacket *avpkt,
 ff_af_queue_remove(>afq, avctx->frame_size, >pts,
>duration);
 
+discard_padding = avctx->frame_size - avpkt->duration;
+// Check if subtraction resulted in an overflow
+if ((discard_padding < avctx->frame_size) != (avpkt->duration > 0)) {
+av_packet_unref(avpkt);
+av_free(avpkt);
+return AVERROR(EINVAL);
+}
+if (discard_padding > 0) {
+uint8_t* side_data = av_packet_new_side_data(avpkt,
+ 
AV_PKT_DATA_SKIP_SAMPLES,
+ 10);
+if(!side_data) {
+av_packet_unref(avpkt);
+av_free(avpkt);
+return AVERROR(ENOMEM);
+}
+AV_WL32(side_data + 4, discard_padding);
+}
+
 avpkt->size = len;
 *got_packet_ptr = 1;
 }
-- 
2.8.0.rc3.226.g39d4020

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[FFmpeg-devel] [PATCH 2/3] lavf/mp3enc: write encoder delay/padding upon closing

[Jon Toohill]

---
 libavformat/mp3enc.c | 32 +---
 1 file changed, 25 insertions(+), 7 deletions(-)

diff --git a/libavformat/mp3enc.c b/libavformat/mp3enc.c
index de63401..48cb0b4 100644
--- a/libavformat/mp3enc.c
+++ b/libavformat/mp3enc.c
@@ -111,6 +111,7 @@ typedef struct MP3Context {
 uint64_t bag[XING_NUM_BAGS];
 int initial_bitrate;
 int has_variable_bitrate;
+int trailing_padding;
 
 /* index of the audio stream */
 int audio_stream_idx;
@@ -247,12 +248,7 @@ static int mp3_write_xing(AVFormatContext *s)
 ffio_fill(dyn_ctx, 0, 8); // empty replaygain fields
 avio_w8(dyn_ctx, 0);  // unknown encoding flags
 avio_w8(dyn_ctx, 0);  // unknown abr/minimal bitrate
-
-// encoder delay
-if (par->initial_padding - 528 - 1 >= 1 << 12) {
-av_log(s, AV_LOG_WARNING, "Too many samples of initial padding.\n");
-}
-avio_wb24(dyn_ctx, FFMAX(par->initial_padding - 528 - 1, 0)<<12);
+avio_wb24(dyn_ctx, 0);// empty encoder delay/padding
 
 avio_w8(dyn_ctx,   0); // misc
 avio_w8(dyn_ctx,   0); // mp3gain
@@ -345,10 +341,22 @@ static int mp3_write_audio_packet(AVFormatContext *s, 
AVPacket *pkt)
 #endif
 
 if (mp3->xing_offset) {
+uint8_t *side_data = NULL;
+int side_data_size = 0;
+
 mp3_xing_add_frame(mp3, pkt);
 mp3->audio_size += pkt->size;
 mp3->audio_crc   = av_crc(av_crc_get_table(AV_CRC_16_ANSI_LE),
   mp3->audio_crc, pkt->data, pkt->size);
+
+side_data = av_packet_get_side_data(pkt,
+AV_PKT_DATA_SKIP_SAMPLES,
+_data_size);
+if (side_data && side_data_size >= 10) {
+mp3->trailing_padding = AV_RL32(side_data + 4);
+} else {
+mp3->trailing_padding = 0;
+}
 }
 }
 
@@ -381,7 +389,7 @@ static void mp3_update_xing(AVFormatContext *s)
 AVReplayGain *rg;
 uint16_t tag_crc;
 uint8_t *toc;
-int i, rg_size;
+int i, rg_size, delay;
 
 /* replace "Xing" identification string with "Info" for CBR files. */
 if (!mp3->has_variable_bitrate)
@@ -422,6 +430,16 @@ static void mp3_update_xing(AVFormatContext *s)
 }
 }
 
+/* write encoder delay/padding */
+delay = FFMAX(s->streams[0]->codecpar->initial_padding - 528 - 1, 0);
+if (delay >= 1 << 12) {
+av_log(s, AV_LOG_WARNING, "Too many samples of initial padding.\n");
+}
+if (mp3->trailing_padding >= 1 << 12) {
+av_log(s, AV_LOG_WARNING, "Too many samples of trailing padding.\n");
+}
+AV_WB24(mp3->xing_frame + mp3->xing_offset + 141, (delay << 12) + 
mp3->trailing_padding);
+
 AV_WB32(mp3->xing_frame + mp3->xing_offset + XING_SIZE - 8, 
mp3->audio_size);
 AV_WB16(mp3->xing_frame + mp3->xing_offset + XING_SIZE - 4, 
mp3->audio_crc);
 
-- 
2.8.0.rc3.226.g39d4020

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[FFmpeg-devel] [PATCH 0/3] Fix gapless encoding/decoding for MP3

[Jon Toohill]

Round trip wav->mp3->wav now preserves the correct number of samples.
Remuxing mp3->mp3 with -c:a copy also preserves any existing gapless
metadata in the Info tag.

The code in libmp3lame.c to set AV_PKT_DATA_SKIP_SAMPLES was mostly
copied from libopusenc.c.

Jon Toohill (3):
  lavc/libmp3lame: send encoder padding in side data of final packet
  lavf/mp3enc: write encoder delay/padding upon closing
  lavf/mp3dec: read encoder delay/padding from Info tag

 libavcodec/libmp3lame.c  | 21 -
 libavformat/mp3dec.c |  3 ++-
 libavformat/mp3enc.c | 32 +---
 tests/ref/fate/audiomatch-square-mp3 |  2 +-
 tests/ref/fate/gapless-mp3   | 10 +-
 5 files changed, 53 insertions(+), 15 deletions(-)

-- 
2.8.0.rc3.226.g39d4020

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Re: [FFmpeg-devel] [PATCH 1/4] ffmpeg: re-copy codec contexts after encoding

[Jon Toohill]

This was sent in error, please disregard.


Jon Toohill |  Google Play Music |  jtooh...@google.com |  (650) 215-0770

On Wed, Sep 28, 2016 at 11:28 AM, Jon Toohill <jtooh...@google.com> wrote:

> This preserves changes to fields of AVCodecContext that get
> updated during encoding, such as trailing_padding (which
> may not be known until encoding is complete).
> ---
>  ffmpeg.c | 15 +++
>  1 file changed, 15 insertions(+)
>
> diff --git a/ffmpeg.c b/ffmpeg.c
> index df55a49..1e973f5 100644
> --- a/ffmpeg.c
> +++ b/ffmpeg.c
> @@ -4243,6 +4243,21 @@ static int transcode(void)
>
>  term_exit();
>
> +/* update output codec contexts after encoding */
> +for (i = 0; i < nb_output_streams; i++) {
> +ost = output_streams[i];
> +if (ost->encoding_needed) {
> +ret = avcodec_copy_context(
> +output_files[ost->file_index]->ctx->streams[ost->index]->
> codec,
> +ost->enc_ctx);
> +if (ret < 0) {
> +av_log(ost, AV_LOG_ERROR, "Error copying final codec
> context: %s\n", av_err2str(ret));
> +if (exit_on_error)
> +exit_program(1);
> +}
> +}
> +}
> +
>  /* write the trailer if needed and close file */
>  for (i = 0; i < nb_output_files; i++) {
>  os = output_files[i]->ctx;
> --
> 2.8.0.rc3.226.g39d4020
>
>
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Re: [FFmpeg-devel] [PATCH 0/4] Handle trailing_padding in MP3 Info tag

[Jon Toohill]

Oops, forgot to send this in reply to the previous thread. Should I re-send?
http://ffmpeg.org/pipermail/ffmpeg-devel/2016-September/200092.html

On Wed, Sep 28, 2016 at 11:28 AM, Jon Toohill <jtooh...@google.com> wrote:

> Trimming trailing_padding samples from the end of the track is not yet
> implemented.
>
> Jon Toohill (4):
>   ffmpeg: re-copy codec parameters after encoding
>   lavc/libmp3lame: set trailing_padding after flushing encoder
>   lavf/mp3enc: write encoder delay/padding upon closing
>   lavf/mp3dec: read encoder delay/padding from Info tag
>
>  ffmpeg.c| 15 +++
>  libavcodec/libmp3lame.c |  1 +
>  libavformat/mp3dec.c|  2 ++
>  libavformat/mp3enc.c| 20 +---
>  4 files changed, 31 insertions(+), 7 deletions(-)
>
> --
> 2.8.0.rc3.226.g39d4020
>
>
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[FFmpeg-devel] [PATCH 4/4] lavf/mp3dec: read encoder delay/padding from Info tag

[Jon Toohill]

---
 libavformat/mp3dec.c | 2 ++
 1 file changed, 2 insertions(+)

diff --git a/libavformat/mp3dec.c b/libavformat/mp3dec.c
index 56c7f8c..9cc85a3 100644
--- a/libavformat/mp3dec.c
+++ b/libavformat/mp3dec.c
@@ -239,6 +239,8 @@ static void mp3_parse_info_tag(AVFormatContext *s, AVStream 
*st,
 
 mp3->start_pad = v>>12;
 mp3->  end_pad = v&4095;
+st->codecpar->initial_padding = mp3->start_pad + 528 + 1;
+st->codecpar->trailing_padding = mp3->end_pad;
 st->start_skip_samples = mp3->start_pad + 528 + 1;
 if (mp3->frames) {
 st->first_discard_sample = -mp3->end_pad + 528 + 1 + mp3->frames * 
(int64_t)spf;
-- 
2.8.0.rc3.226.g39d4020

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[FFmpeg-devel] [PATCH 3/4] lavf/mp3enc: write encoder delay/padding upon closing

[Jon Toohill]

trailing_padding is not known before encoding.
---
 libavformat/mp3enc.c | 20 +---
 1 file changed, 13 insertions(+), 7 deletions(-)

diff --git a/libavformat/mp3enc.c b/libavformat/mp3enc.c
index de63401..37608f1 100644
--- a/libavformat/mp3enc.c
+++ b/libavformat/mp3enc.c
@@ -247,12 +247,7 @@ static int mp3_write_xing(AVFormatContext *s)
 ffio_fill(dyn_ctx, 0, 8); // empty replaygain fields
 avio_w8(dyn_ctx, 0);  // unknown encoding flags
 avio_w8(dyn_ctx, 0);  // unknown abr/minimal bitrate
-
-// encoder delay
-if (par->initial_padding - 528 - 1 >= 1 << 12) {
-av_log(s, AV_LOG_WARNING, "Too many samples of initial padding.\n");
-}
-avio_wb24(dyn_ctx, FFMAX(par->initial_padding - 528 - 1, 0)<<12);
+avio_wb24(dyn_ctx, 0);// empty encoder delay/padding
 
 avio_w8(dyn_ctx,   0); // misc
 avio_w8(dyn_ctx,   0); // mp3gain
@@ -381,7 +376,7 @@ static void mp3_update_xing(AVFormatContext *s)
 AVReplayGain *rg;
 uint16_t tag_crc;
 uint8_t *toc;
-int i, rg_size;
+int i, rg_size, delay, padding;
 
 /* replace "Xing" identification string with "Info" for CBR files. */
 if (!mp3->has_variable_bitrate)
@@ -422,6 +417,17 @@ static void mp3_update_xing(AVFormatContext *s)
 }
 }
 
+/* write encoder delay/padding */
+delay = FFMAX(s->streams[0]->codecpar->initial_padding - 528 - 1, 0);
+padding = s->streams[0]->codecpar->trailing_padding;
+if (delay >= 1 << 12) {
+av_log(s, AV_LOG_WARNING, "Too many samples of initial padding.\n");
+}
+if (padding >= 1 << 12) {
+av_log(s, AV_LOG_WARNING, "Too many samples of initial padding.\n");
+}
+AV_WB24(mp3->xing_frame + mp3->xing_offset + 141, (delay << 12) + padding);
+
 AV_WB32(mp3->xing_frame + mp3->xing_offset + XING_SIZE - 8, 
mp3->audio_size);
 AV_WB16(mp3->xing_frame + mp3->xing_offset + XING_SIZE - 4, 
mp3->audio_crc);
 
-- 
2.8.0.rc3.226.g39d4020

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Re: [FFmpeg-devel] [PATCH 2/4] lavc/libmp3lame: set trailing_padding after flushing encoder

[Jon Toohill]

On Tue, Sep 27, 2016 at 1:04 AM, Carl Eugen Hoyos <ceffm...@gmail.com>
wrote:

> 2016-09-26 19:13 GMT+02:00 Jon Toohill <jtoohill-at-google@ffmpeg.org
> >:
>
> > +avctx->trailing_padding = FFMAX(lame_get_encoder_padding(s->gfp)
> - 528 - 1, 0);
>
> Can you confirm that this function exists in lame 3.98.3?
>

I downloaded the source tarball for lame 3.98 and found it exists there. Is
that sufficient?


>
> Thank you, Carl Eugen
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[FFmpeg-devel] [PATCH 1/4] ffmpeg: re-copy codec parameters after encoding

[Jon Toohill]

This preserves changes to fields of AVCodecParameters that
get updated during encoding, such as trailing_padding
(which may not be known until encoding is complete).
---
 ffmpeg.c | 15 +++
 1 file changed, 15 insertions(+)

diff --git a/ffmpeg.c b/ffmpeg.c
index d0f247e..0cdc762 100644
--- a/ffmpeg.c
+++ b/ffmpeg.c
@@ -4255,6 +4255,21 @@ static int transcode(void)
 
 term_exit();
 
+/* update output codec parameters after encoding */
+for (i = 0; i < nb_output_streams; i++) {
+ost = output_streams[i];
+if (ost->encoding_needed) {
+ret = avcodec_parameters_from_context(
+
output_files[ost->file_index]->ctx->streams[ost->index]->codecpar,
+ost->enc_ctx);
+if (ret < 0) {
+av_log(ost, AV_LOG_ERROR, "Error copying final codec context: 
%s\n", av_err2str(ret));
+if (exit_on_error)
+exit_program(1);
+}
+}
+}
+
 /* write the trailer if needed and close file */
 for (i = 0; i < nb_output_files; i++) {
 os = output_files[i]->ctx;
-- 
2.8.0.rc3.226.g39d4020

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[FFmpeg-devel] [PATCH 1/4] ffmpeg: re-copy codec contexts after encoding

[Jon Toohill]

This preserves changes to fields of AVCodecContext that get
updated during encoding, such as trailing_padding (which
may not be known until encoding is complete).
---
 ffmpeg.c | 15 +++
 1 file changed, 15 insertions(+)

diff --git a/ffmpeg.c b/ffmpeg.c
index df55a49..1e973f5 100644
--- a/ffmpeg.c
+++ b/ffmpeg.c
@@ -4243,6 +4243,21 @@ static int transcode(void)
 
 term_exit();
 
+/* update output codec contexts after encoding */
+for (i = 0; i < nb_output_streams; i++) {
+ost = output_streams[i];
+if (ost->encoding_needed) {
+ret = avcodec_copy_context(
+output_files[ost->file_index]->ctx->streams[ost->index]->codec,
+ost->enc_ctx);
+if (ret < 0) {
+av_log(ost, AV_LOG_ERROR, "Error copying final codec context: 
%s\n", av_err2str(ret));
+if (exit_on_error)
+exit_program(1);
+}
+}
+}
+
 /* write the trailer if needed and close file */
 for (i = 0; i < nb_output_files; i++) {
 os = output_files[i]->ctx;
-- 
2.8.0.rc3.226.g39d4020

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[FFmpeg-devel] [PATCH 2/4] lavc/libmp3lame: set trailing_padding after flushing encoder

[Jon Toohill]

---
 libavcodec/libmp3lame.c | 1 +
 1 file changed, 1 insertion(+)

diff --git a/libavcodec/libmp3lame.c b/libavcodec/libmp3lame.c
index 5642264..1566921 100644
--- a/libavcodec/libmp3lame.c
+++ b/libavcodec/libmp3lame.c
@@ -218,6 +218,7 @@ static int mp3lame_encode_frame(AVCodecContext *avctx, 
AVPacket *avpkt,
 } else {
 lame_result = lame_encode_flush(s->gfp, s->buffer + s->buffer_index,
 s->buffer_size - s->buffer_index);
+avctx->trailing_padding = FFMAX(lame_get_encoder_padding(s->gfp) - 528 
- 1, 0);
 }
 if (lame_result < 0) {
 if (lame_result == -1) {
-- 
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[FFmpeg-devel] [PATCH 0/4] Handle trailing_padding in MP3 Info tag

[Jon Toohill]

Trimming trailing_padding samples from the end of the track is not yet
implemented.

Jon Toohill (4):
  ffmpeg: re-copy codec parameters after encoding
  lavc/libmp3lame: set trailing_padding after flushing encoder
  lavf/mp3enc: write encoder delay/padding upon closing
  lavf/mp3dec: read encoder delay/padding from Info tag

 ffmpeg.c| 15 +++
 libavcodec/libmp3lame.c |  1 +
 libavformat/mp3dec.c|  2 ++
 libavformat/mp3enc.c| 20 +---
 4 files changed, 31 insertions(+), 7 deletions(-)

-- 
2.8.0.rc3.226.g39d4020

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Re: [FFmpeg-devel] [PATCH 4/4] lavf/mp3dec: read encoder delay/padding from Info tag

[Jon Toohill]

A similar concern was raised in a previous related patch:
http://ffmpeg.org/pipermail/ffmpeg-devel/2016-May/194690.html
I think the resolution at the time was to go ahead with using both, since
both are used widely and serve slightly different purposes, although I do
agree that it's confusing.

I think I could use AV_PKT_DATA_SKIP_SAMPLES here, and change mp3enc to use
that to fill out the Info tag. But that only seems worthwhile to me if
there is a general consensus to move to just AV_PKT_DATA_SKIP_SAMPLES (and
deprecate/remove initial_padding and trailing_padding). If there's a
concrete case where knowing trailing_padding at the start of a stream is
necessary, then that's not possible, but I'm pretty new to this and don't
know of one. Thoughts?


Jon Toohill |  Google Play Music |  jtooh...@google.com |  (650) 215-0770

On Mon, Sep 26, 2016 at 11:30 AM, wm4 <nfx...@googlemail.com> wrote:

> On Mon, 26 Sep 2016 10:13:39 -0700
> Jon Toohill <jtoohill-at-google@ffmpeg.org> wrote:
>
> > ---
> >  libavformat/mp3dec.c | 2 ++
> >  1 file changed, 2 insertions(+)
> >
> > diff --git a/libavformat/mp3dec.c b/libavformat/mp3dec.c
> > index 56c7f8c..9cc85a3 100644
> > --- a/libavformat/mp3dec.c
> > +++ b/libavformat/mp3dec.c
> > @@ -239,6 +239,8 @@ static void mp3_parse_info_tag(AVFormatContext *s,
> AVStream *st,
> >
> >  mp3->start_pad = v>>12;
> >  mp3->  end_pad = v&4095;
> > +st->codecpar->initial_padding = mp3->start_pad + 528 + 1;
> > +st->codecpar->trailing_padding = mp3->end_pad;
> >  st->start_skip_samples = mp3->start_pad + 528 + 1;
> >  if (mp3->frames) {
> >  st->first_discard_sample = -mp3->end_pad + 528 + 1 +
> mp3->frames * (int64_t)spf;
>
> I'm somewhat suspicious about this, because mp3dec.c uses
> AV_PKT_DATA_SKIP_SAMPLES to communicate delay/padding
> (libavformat/utils.c turns the start_skip_samples field into side
> data). So I'm not quite convinced is this mess of FFmpeg and Libav API
> mixture is healthy. Opinions welcome.
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[FFmpeg-devel] [PATCH 3/4] lavf/mp3enc: write encoder delay/padding upon closing

[Jon Toohill]

trailing_padding is not known before encoding.
---
 libavformat/mp3enc.c | 20 +---
 1 file changed, 13 insertions(+), 7 deletions(-)

diff --git a/libavformat/mp3enc.c b/libavformat/mp3enc.c
index de63401..433b070 100644
--- a/libavformat/mp3enc.c
+++ b/libavformat/mp3enc.c
@@ -247,12 +247,7 @@ static int mp3_write_xing(AVFormatContext *s)
 ffio_fill(dyn_ctx, 0, 8); // empty replaygain fields
 avio_w8(dyn_ctx, 0);  // unknown encoding flags
 avio_w8(dyn_ctx, 0);  // unknown abr/minimal bitrate
-
-// encoder delay
-if (par->initial_padding - 528 - 1 >= 1 << 12) {
-av_log(s, AV_LOG_WARNING, "Too many samples of initial padding.\n");
-}
-avio_wb24(dyn_ctx, FFMAX(par->initial_padding - 528 - 1, 0)<<12);
+avio_wb24(dyn_ctx, 0);// empty encoder delay/padding
 
 avio_w8(dyn_ctx,   0); // misc
 avio_w8(dyn_ctx,   0); // mp3gain
@@ -381,7 +376,7 @@ static void mp3_update_xing(AVFormatContext *s)
 AVReplayGain *rg;
 uint16_t tag_crc;
 uint8_t *toc;
-int i, rg_size;
+int i, rg_size, delay, padding;
 
 /* replace "Xing" identification string with "Info" for CBR files. */
 if (!mp3->has_variable_bitrate)
@@ -422,6 +417,17 @@ static void mp3_update_xing(AVFormatContext *s)
 }
 }
 
+/* write encoder delay/padding */
+delay = FFMAX(s->streams[0]->codec->initial_padding - 528 - 1, 0);
+padding = s->streams[0]->codec->trailing_padding;
+if (delay >= 1 << 12) {
+av_log(s, AV_LOG_WARNING, "Too many samples of initial padding.\n");
+}
+if (padding >= 1 << 12) {
+av_log(s, AV_LOG_WARNING, "Too many samples of initial padding.\n");
+}
+AV_WB24(mp3->xing_frame + mp3->xing_offset + 141, (delay << 12) + padding);
+
 AV_WB32(mp3->xing_frame + mp3->xing_offset + XING_SIZE - 8, 
mp3->audio_size);
 AV_WB16(mp3->xing_frame + mp3->xing_offset + XING_SIZE - 4, 
mp3->audio_crc);
 
-- 
2.8.0.rc3.226.g39d4020

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[FFmpeg-devel] [PATCH 4/4] lavf/mp3dec: read encoder delay/padding from Info tag

[Jon Toohill]

---
 libavformat/mp3dec.c | 2 ++
 1 file changed, 2 insertions(+)

diff --git a/libavformat/mp3dec.c b/libavformat/mp3dec.c
index 56c7f8c..9cc85a3 100644
--- a/libavformat/mp3dec.c
+++ b/libavformat/mp3dec.c
@@ -239,6 +239,8 @@ static void mp3_parse_info_tag(AVFormatContext *s, AVStream 
*st,
 
 mp3->start_pad = v>>12;
 mp3->  end_pad = v&4095;
+st->codecpar->initial_padding = mp3->start_pad + 528 + 1;
+st->codecpar->trailing_padding = mp3->end_pad;
 st->start_skip_samples = mp3->start_pad + 528 + 1;
 if (mp3->frames) {
 st->first_discard_sample = -mp3->end_pad + 528 + 1 + mp3->frames * 
(int64_t)spf;
-- 
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[FFmpeg-devel] [PATCH 1/4] ffmpeg: re-copy codec contexts after encoding

[Jon Toohill]

This preserves changes to fields of AVCodecContext that get
updated during encoding, such as trailing_padding (which
may not be known until encoding is complete).
---
 ffmpeg.c | 15 +++
 1 file changed, 15 insertions(+)

diff --git a/ffmpeg.c b/ffmpeg.c
index df55a49..1e973f5 100644
--- a/ffmpeg.c
+++ b/ffmpeg.c
@@ -4243,6 +4243,21 @@ static int transcode(void)
 
 term_exit();
 
+/* update output codec contexts after encoding */
+for (i = 0; i < nb_output_streams; i++) {
+ost = output_streams[i];
+if (ost->encoding_needed) {
+ret = avcodec_copy_context(
+output_files[ost->file_index]->ctx->streams[ost->index]->codec,
+ost->enc_ctx);
+if (ret < 0) {
+av_log(ost, AV_LOG_ERROR, "Error copying final codec context: 
%s\n", av_err2str(ret));
+if (exit_on_error)
+exit_program(1);
+}
+}
+}
+
 /* write the trailer if needed and close file */
 for (i = 0; i < nb_output_files; i++) {
 os = output_files[i]->ctx;
-- 
2.8.0.rc3.226.g39d4020

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[FFmpeg-devel] [PATCH 2/4] lavc/libmp3lame: set trailing_padding after flushing encoder

[Jon Toohill]

---
 libavcodec/libmp3lame.c | 1 +
 1 file changed, 1 insertion(+)

diff --git a/libavcodec/libmp3lame.c b/libavcodec/libmp3lame.c
index 5642264..1566921 100644
--- a/libavcodec/libmp3lame.c
+++ b/libavcodec/libmp3lame.c
@@ -218,6 +218,7 @@ static int mp3lame_encode_frame(AVCodecContext *avctx, 
AVPacket *avpkt,
 } else {
 lame_result = lame_encode_flush(s->gfp, s->buffer + s->buffer_index,
 s->buffer_size - s->buffer_index);
+avctx->trailing_padding = FFMAX(lame_get_encoder_padding(s->gfp) - 528 
- 1, 0);
 }
 if (lame_result < 0) {
 if (lame_result == -1) {
-- 
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[FFmpeg-devel] [PATCH 0/4] Handle trailing_padding in MP3 Info tag

[Jon Toohill]

Trimming trailing_padding samples from the end of the track is not yet
implemented.

Jon Toohill (4):
  ffmpeg: re-copy codec contexts after encoding
  lavc/libmp3lame: set trailing_padding after flushing encoder
  lavf/mp3enc: write encoder delay/padding upon closing
  lavf/mp3dec: read encoder delay/padding from Info tag

 ffmpeg.c| 15 +++
 libavcodec/libmp3lame.c |  1 +
 libavformat/mp3dec.c|  2 ++
 libavformat/mp3enc.c| 20 +---
 4 files changed, 31 insertions(+), 7 deletions(-)

-- 
2.8.0.rc3.226.g39d4020

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[FFmpeg-devel] [PATCH 2/2] ffmpeg: copy trailing_padding when using -acodec copy

[Jon Toohill]

---
 ffmpeg.c | 1 +
 1 file changed, 1 insertion(+)

diff --git a/ffmpeg.c b/ffmpeg.c
index bae515d..49a1b03 100644
--- a/ffmpeg.c
+++ b/ffmpeg.c
@@ -3001,6 +3001,7 @@ static int transcode_init(void)
 enc_ctx->audio_service_type = dec_ctx->audio_service_type;
 enc_ctx->block_align= dec_ctx->block_align;
 enc_ctx->initial_padding= dec_ctx->delay;
+enc_ctx->trailing_padding   = dec_ctx->trailing_padding;
 enc_ctx->profile= dec_ctx->profile;
 #if FF_API_AUDIOENC_DELAY
 enc_ctx->delay  = dec_ctx->delay;
-- 
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[FFmpeg-devel] [PATCH 1/2] lavc: add trailing_padding to AVCodecContext to match AVCodecParameters.

[Jon Toohill]

Shows encoder delay/padding in the stream summary if they are set.
---
 doc/APIchanges   |  4 
 libavcodec/avcodec.h | 11 +++
 libavcodec/utils.c   | 40 +++-
 libavcodec/version.h |  2 +-
 4 files changed, 39 insertions(+), 18 deletions(-)

diff --git a/doc/APIchanges b/doc/APIchanges
index 209ab41..74145b2 100644
--- a/doc/APIchanges
+++ b/doc/APIchanges
@@ -15,6 +15,10 @@ libavutil: 2015-08-28
 
 API changes, most recent first:
 
+2016-08-15 - xxx - lavc 57.53.100 - avcodec.h
+  Add trailing_padding to AVCodecContext to match the corresponding
+  field in AVCodecParameters.
+
 2016-08-04 - xxx - lavf 57.46.100 - avformat.h
   Add av_get_frame_filename2()
 
diff --git a/libavcodec/avcodec.h b/libavcodec/avcodec.h
index 06c2b89..b43ee5a 100644
--- a/libavcodec/avcodec.h
+++ b/libavcodec/avcodec.h
@@ -3524,6 +3524,17 @@ typedef struct AVCodecContext {
 #define FF_SUB_TEXT_FMT_ASS_WITH_TIMINGS 1
 #endif
 
+/**
+ * Audio only. The amount of padding (in samples) appended by the encoder 
to
+ * the end of the audio. I.e. this number of decoded samples must be
+ * discarded by the caller from the end of the stream to get the original
+ * audio without any trailing padding.
+ *
+ * - decoding: unused
+ * - encoding: unused
+ */
+int trailing_padding;
+
 } AVCodecContext;
 
 AVRational av_codec_get_pkt_timebase (const AVCodecContext *avctx);
diff --git a/libavcodec/utils.c b/libavcodec/utils.c
index 783f62c..6f4d553 100644
--- a/libavcodec/utils.c
+++ b/libavcodec/utils.c
@@ -3258,6 +3258,10 @@ void avcodec_string(char *buf, int buf_size, 
AVCodecContext *enc, int encode)
 && enc->bits_per_raw_sample != 
av_get_bytes_per_sample(enc->sample_fmt) * 8)
 snprintf(buf + strlen(buf), buf_size - strlen(buf),
  " (%d bit)", enc->bits_per_raw_sample);
+if (enc->initial_padding || enc->trailing_padding) {
+snprintf(buf + strlen(buf), buf_size - strlen(buf),
+ ", delay %d, padding %d", enc->initial_padding, 
enc->trailing_padding);
+}
 break;
 case AVMEDIA_TYPE_DATA:
 if (av_log_get_level() >= AV_LOG_DEBUG) {
@@ -4103,14 +4107,15 @@ int avcodec_parameters_from_context(AVCodecParameters 
*par,
 par->video_delay = codec->has_b_frames;
 break;
 case AVMEDIA_TYPE_AUDIO:
-par->format  = codec->sample_fmt;
-par->channel_layout  = codec->channel_layout;
-par->channels= codec->channels;
-par->sample_rate = codec->sample_rate;
-par->block_align = codec->block_align;
-par->frame_size  = codec->frame_size;
-par->initial_padding = codec->initial_padding;
-par->seek_preroll= codec->seek_preroll;
+par->format   = codec->sample_fmt;
+par->channel_layout   = codec->channel_layout;
+par->channels = codec->channels;
+par->sample_rate  = codec->sample_rate;
+par->block_align  = codec->block_align;
+par->frame_size   = codec->frame_size;
+par->initial_padding  = codec->initial_padding;
+par->trailing_padding = codec->trailing_padding;
+par->seek_preroll = codec->seek_preroll;
 break;
 case AVMEDIA_TYPE_SUBTITLE:
 par->width  = codec->width;
@@ -4157,15 +4162,16 @@ int avcodec_parameters_to_context(AVCodecContext *codec,
 codec->has_b_frames   = par->video_delay;
 break;
 case AVMEDIA_TYPE_AUDIO:
-codec->sample_fmt  = par->format;
-codec->channel_layout  = par->channel_layout;
-codec->channels= par->channels;
-codec->sample_rate = par->sample_rate;
-codec->block_align = par->block_align;
-codec->frame_size  = par->frame_size;
-codec->delay   =
-codec->initial_padding = par->initial_padding;
-codec->seek_preroll= par->seek_preroll;
+codec->sample_fmt   = par->format;
+codec->channel_layout   = par->channel_layout;
+codec->channels = par->channels;
+codec->sample_rate  = par->sample_rate;
+codec->block_align  = par->block_align;
+codec->frame_size   = par->frame_size;
+codec->delay=
+codec->initial_padding  = par->initial_padding;
+codec->trailing_padding = par->trailing_padding;
+codec->seek_preroll = par->seek_preroll;
 break;
 case AVMEDIA_TYPE_SUBTITLE:
 codec->width  = par->width;
diff --git a/libavcodec/version.h b/libavcodec/version.h
index a697261..cdfc4f9 100644
--- a/libavcodec/version.h
+++ b/libavcodec/version.h
@@ -28,7 +28,7 @@
 #include "libavutil/version.h"
 
 #define LIBAVCODEC_VERSION_MAJOR  57
-#define LIBAVCODEC_VERSION_MINOR  52
+#define LIBAVCODEC_VERSION_MINOR  53
 #define 

[FFmpeg-devel] [PATCH 0/2] Minor fixes for encoder padding

[Jon Toohill]

These patches backport the trailing_padding field to AVCodecContext,
add gapless information to the stream summary string, and fix some
cases where trailing_padding was being lost.

I'll continue working on supporting initial_padding and
trailing_padding for MP3 encoding/decoding in a separate patch set.

Jon Toohill (2):
  lavc: add trailing_padding to AVCodecContext to match
AVCodecParameters.
  ffmpeg: copy trailing_padding when using -acodec copy

 doc/APIchanges   |  4 
 ffmpeg.c |  1 +
 libavcodec/avcodec.h | 11 +++
 libavcodec/utils.c   | 40 +++-
 libavcodec/version.h |  2 +-
 5 files changed, 40 insertions(+), 18 deletions(-)

-- 
2.8.0.rc3.226.g39d4020

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[FFmpeg-devel] [PATCH 2/5] libmp3lame + mp3enc: removes decoder delay compensation

[Jon Toohill]

initial_padding specifies only encoder delay, decoder delay is
handled by start_skip_samples.
---
 doc/APIchanges  | 4 
 libavcodec/libmp3lame.c | 2 +-
 libavcodec/version.h| 2 +-
 libavformat/mp3enc.c| 4 ++--
 4 files changed, 8 insertions(+), 4 deletions(-)

diff --git a/doc/APIchanges b/doc/APIchanges
index d777dc0..ae450e1 100644
--- a/doc/APIchanges
+++ b/doc/APIchanges
@@ -15,6 +15,10 @@ libavutil: 2015-08-28
 
 API changes, most recent first:
 
+2016-07-12 - xxx - lavc 57.47.100 - libmp3lame.c
+  Removed MP3 decoder delay from initial_padding in AVCodecContext.
+  initial_padding only includes the encoder delay.
+
 2016-04-27 - xxx - lavu 55.23.100 - log.h
   Add a new function av_log_format_line2() which returns number of bytes
   written to the target buffer.
diff --git a/libavcodec/libmp3lame.c b/libavcodec/libmp3lame.c
index 5642264..198ac94 100644
--- a/libavcodec/libmp3lame.c
+++ b/libavcodec/libmp3lame.c
@@ -137,7 +137,7 @@ static av_cold int mp3lame_encode_init(AVCodecContext 
*avctx)
 }
 
 /* get encoder delay */
-avctx->initial_padding = lame_get_encoder_delay(s->gfp) + 528 + 1;
+avctx->initial_padding = lame_get_encoder_delay(s->gfp);
 ff_af_queue_init(avctx, >afq);
 
 avctx->frame_size  = lame_get_framesize(s->gfp);
diff --git a/libavcodec/version.h b/libavcodec/version.h
index 0852b43..37a6e17 100644
--- a/libavcodec/version.h
+++ b/libavcodec/version.h
@@ -28,7 +28,7 @@
 #include "libavutil/version.h"
 
 #define LIBAVCODEC_VERSION_MAJOR  57
-#define LIBAVCODEC_VERSION_MINOR  46
+#define LIBAVCODEC_VERSION_MINOR  47
 #define LIBAVCODEC_VERSION_MICRO 100
 
 #define LIBAVCODEC_VERSION_INT  AV_VERSION_INT(LIBAVCODEC_VERSION_MAJOR, \
diff --git a/libavformat/mp3enc.c b/libavformat/mp3enc.c
index de63401..3b77d29 100644
--- a/libavformat/mp3enc.c
+++ b/libavformat/mp3enc.c
@@ -249,10 +249,10 @@ static int mp3_write_xing(AVFormatContext *s)
 avio_w8(dyn_ctx, 0);  // unknown abr/minimal bitrate
 
 // encoder delay
-if (par->initial_padding - 528 - 1 >= 1 << 12) {
+if (par->initial_padding >= 1 << 12) {
 av_log(s, AV_LOG_WARNING, "Too many samples of initial padding.\n");
 }
-avio_wb24(dyn_ctx, FFMAX(par->initial_padding - 528 - 1, 0)<<12);
+avio_wb24(dyn_ctx, par->initial_padding << 12);
 
 avio_w8(dyn_ctx,   0); // misc
 avio_w8(dyn_ctx,   0); // mp3gain
-- 
2.8.0.rc3.226.g39d4020

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[FFmpeg-devel] [PATCH 4/5] ffmpeg: copy trailing_padding when using -acodec copy

[Jon Toohill]

---
 ffmpeg.c | 1 +
 1 file changed, 1 insertion(+)

diff --git a/ffmpeg.c b/ffmpeg.c
index 652774f..442f818 100644
--- a/ffmpeg.c
+++ b/ffmpeg.c
@@ -3001,6 +3001,7 @@ static int transcode_init(void)
 enc_ctx->audio_service_type = dec_ctx->audio_service_type;
 enc_ctx->block_align= dec_ctx->block_align;
 enc_ctx->initial_padding= dec_ctx->delay;
+enc_ctx->trailing_padding   = dec_ctx->trailing_padding;
 enc_ctx->profile= dec_ctx->profile;
 #if FF_API_AUDIOENC_DELAY
 enc_ctx->delay  = dec_ctx->delay;
-- 
2.8.0.rc3.226.g39d4020

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[FFmpeg-devel] [PATCH 5/5] lavf/mp3enc: write trailing_padding in Xing header

[Jon Toohill]

---
 libavformat/mp3enc.c | 7 +--
 1 file changed, 5 insertions(+), 2 deletions(-)

diff --git a/libavformat/mp3enc.c b/libavformat/mp3enc.c
index 3b77d29..da70d13 100644
--- a/libavformat/mp3enc.c
+++ b/libavformat/mp3enc.c
@@ -248,11 +248,14 @@ static int mp3_write_xing(AVFormatContext *s)
 avio_w8(dyn_ctx, 0);  // unknown encoding flags
 avio_w8(dyn_ctx, 0);  // unknown abr/minimal bitrate
 
-// encoder delay
+// encoder delay/padding
 if (par->initial_padding >= 1 << 12) {
 av_log(s, AV_LOG_WARNING, "Too many samples of initial padding.\n");
 }
-avio_wb24(dyn_ctx, par->initial_padding << 12);
+if (par->trailing_padding >= 1 << 12) {
+av_log(s, AV_LOG_WARNING, "Too many samples of trailing padding.\n");
+}
+avio_wb24(dyn_ctx, (par->initial_padding << 12) | (par->trailing_padding & 
0xFFF));
 
 avio_w8(dyn_ctx,   0); // misc
 avio_w8(dyn_ctx,   0); // mp3gain
-- 
2.8.0.rc3.226.g39d4020

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[FFmpeg-devel] [PATCH 3/5] lavc: show gapless info in stream summary

[Jon Toohill]

Also adds trailing_padding to AVCodecContext to match
AVCodecParameters so that it doesn't get lost when mapping
between them.
---
 doc/APIchanges   |  4 
 libavcodec/avcodec.h | 11 +++
 libavcodec/utils.c   | 40 +++-
 libavcodec/version.h |  2 +-
 4 files changed, 39 insertions(+), 18 deletions(-)

diff --git a/doc/APIchanges b/doc/APIchanges
index ae450e1..6e92cd4 100644
--- a/doc/APIchanges
+++ b/doc/APIchanges
@@ -15,6 +15,10 @@ libavutil: 2015-08-28
 
 API changes, most recent first:
 
+2016-07-12 - xxx - lavc 57.48.100 - avcodec.h
+  Add trailing_padding to AVCodecContext to match the corresponding
+  field in AVCodecParameters.
+
 2016-07-12 - xxx - lavc 57.47.100 - libmp3lame.c
   Removed MP3 decoder delay from initial_padding in AVCodecContext.
   initial_padding only includes the encoder delay.
diff --git a/libavcodec/avcodec.h b/libavcodec/avcodec.h
index 0181eb1..700c9b8 100644
--- a/libavcodec/avcodec.h
+++ b/libavcodec/avcodec.h
@@ -3504,6 +3504,17 @@ typedef struct AVCodecContext {
 #define FF_SUB_TEXT_FMT_ASS_WITH_TIMINGS 1
 #endif
 
+/**
+ * Audio only. The amount of padding (in samples) appended by the encoder 
to
+ * the end of the audio. I.e. this number of decoded samples must be
+ * discarded by the caller from the end of the stream to get the original
+ * audio without any trailing padding.
+ *
+ * - decoding: unused
+ * - encoding: unused
+ */
+int trailing_padding;
+
 } AVCodecContext;
 
 AVRational av_codec_get_pkt_timebase (const AVCodecContext *avctx);
diff --git a/libavcodec/utils.c b/libavcodec/utils.c
index 402a9d8..481e09c 100644
--- a/libavcodec/utils.c
+++ b/libavcodec/utils.c
@@ -3252,6 +3252,10 @@ void avcodec_string(char *buf, int buf_size, 
AVCodecContext *enc, int encode)
 && enc->bits_per_raw_sample != 
av_get_bytes_per_sample(enc->sample_fmt) * 8)
 snprintf(buf + strlen(buf), buf_size - strlen(buf),
  " (%d bit)", enc->bits_per_raw_sample);
+if (enc->initial_padding || enc->trailing_padding) {
+snprintf(buf + strlen(buf), buf_size - strlen(buf),
+ ", delay %d, padding %d", enc->initial_padding, 
enc->trailing_padding);
+}
 break;
 case AVMEDIA_TYPE_DATA:
 if (av_log_get_level() >= AV_LOG_DEBUG) {
@@ -4097,14 +4101,15 @@ int avcodec_parameters_from_context(AVCodecParameters 
*par,
 par->video_delay = codec->has_b_frames;
 break;
 case AVMEDIA_TYPE_AUDIO:
-par->format  = codec->sample_fmt;
-par->channel_layout  = codec->channel_layout;
-par->channels= codec->channels;
-par->sample_rate = codec->sample_rate;
-par->block_align = codec->block_align;
-par->frame_size  = codec->frame_size;
-par->initial_padding = codec->initial_padding;
-par->seek_preroll= codec->seek_preroll;
+par->format   = codec->sample_fmt;
+par->channel_layout   = codec->channel_layout;
+par->channels = codec->channels;
+par->sample_rate  = codec->sample_rate;
+par->block_align  = codec->block_align;
+par->frame_size   = codec->frame_size;
+par->initial_padding  = codec->initial_padding;
+par->trailing_padding = codec->trailing_padding;
+par->seek_preroll = codec->seek_preroll;
 break;
 case AVMEDIA_TYPE_SUBTITLE:
 par->width  = codec->width;
@@ -4151,15 +4156,16 @@ int avcodec_parameters_to_context(AVCodecContext *codec,
 codec->has_b_frames   = par->video_delay;
 break;
 case AVMEDIA_TYPE_AUDIO:
-codec->sample_fmt  = par->format;
-codec->channel_layout  = par->channel_layout;
-codec->channels= par->channels;
-codec->sample_rate = par->sample_rate;
-codec->block_align = par->block_align;
-codec->frame_size  = par->frame_size;
-codec->delay   =
-codec->initial_padding = par->initial_padding;
-codec->seek_preroll= par->seek_preroll;
+codec->sample_fmt   = par->format;
+codec->channel_layout   = par->channel_layout;
+codec->channels = par->channels;
+codec->sample_rate  = par->sample_rate;
+codec->block_align  = par->block_align;
+codec->frame_size   = par->frame_size;
+codec->delay=
+codec->initial_padding  = par->initial_padding;
+codec->trailing_padding = par->trailing_padding;
+codec->seek_preroll = par->seek_preroll;
 break;
 case AVMEDIA_TYPE_SUBTITLE:
 codec->width  = par->width;
diff --git a/libavcodec/version.h b/libavcodec/version.h
index 37a6e17..412fd01 100644
--- a/libavcodec/version.h
+++ b/libavcodec/version.h
@@ -28,7 +28,7 @@
 #include 

[FFmpeg-devel] [PATCH 1/5] lavf/mp3dec: pass Xing gapless metadata to AVCodecParameters

[Jon Toohill]

---
 libavformat/mp3dec.c | 2 ++
 1 file changed, 2 insertions(+)

diff --git a/libavformat/mp3dec.c b/libavformat/mp3dec.c
index 56c7f8c..345fa88 100644
--- a/libavformat/mp3dec.c
+++ b/libavformat/mp3dec.c
@@ -239,6 +239,8 @@ static void mp3_parse_info_tag(AVFormatContext *s, AVStream 
*st,
 
 mp3->start_pad = v>>12;
 mp3->  end_pad = v&4095;
+st->codecpar->initial_padding = mp3->start_pad;
+st->codecpar->trailing_padding = mp3->end_pad;
 st->start_skip_samples = mp3->start_pad + 528 + 1;
 if (mp3->frames) {
 st->first_discard_sample = -mp3->end_pad + 528 + 1 + mp3->frames * 
(int64_t)spf;
-- 
2.8.0.rc3.226.g39d4020

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[FFmpeg-devel] [PATCH 0/5] Pass Xing gapless metadata to users during mp3 parsing

[Jon Toohill]

These patches expose the encoder delay/padding parsed from an mp3's Xing header 
to users of lavc/lavf, and show gapless info in the stream summary string. They 
also change ffmpeg to pass Xing gapless metadata from input to output when 
using -acodec copy.

trailing_padding is still not set properly when encoding with libmp3lame, 
causing an encode/decode round trip to add trailing silence. This is not a 
regression from current behavior, and will be addressed in a separate patch set.

Jon Toohill (5):
  lavf/mp3dec: pass Xing gapless metadata to AVCodecParameters
  libmp3lame + mp3enc: removes decoder delay compensation
  lavc: show gapless info in stream summary
  ffmpeg: copy trailing_padding when using -acodec copy
  lavf/mp3enc: write trailing_padding in Xing header

 doc/APIchanges  |  8 
 ffmpeg.c|  1 +
 libavcodec/avcodec.h| 11 +++
 libavcodec/libmp3lame.c |  2 +-
 libavcodec/utils.c  | 40 +++-
 libavcodec/version.h|  2 +-
 libavformat/mp3dec.c|  2 ++
 libavformat/mp3enc.c|  9 ++---
 8 files changed, 53 insertions(+), 22 deletions(-)

-- 
2.8.0.rc3.226.g39d4020

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Re: [FFmpeg-devel] [PATCH 1/3] lavf/mp3dec: pass Xing gapless metadata to AVCodecParameters

[Jon Toohill]

I'm having a hard time finding software that uses this via simple GitHub
searches.

I think this should be considered a bugfix, since the struct field states
that it only represents encoder delay, not decoder delay. I'll send out an
updated patchset that splits this into more patches, including
documentation and another minor version bump.


Jon Toohill |  Google Play Music |  jtooh...@google.com |  (650) 215-0770

On Fri, Jun 17, 2016 at 5:32 PM, Michael Niedermayer <mich...@niedermayer.cc
> wrote:

> On Thu, Jun 16, 2016 at 11:16:05AM -0700, Jon Toohill wrote:
> > Also removes decoder delay compensation from libmp3lame and mp3enc.
> > initial_padding specifies only encoder delay, decoder delay is
> > handled by start_skip_samples.
> > ---
> >  libavcodec/libmp3lame.c | 2 +-
> >  libavformat/mp3dec.c| 2 ++
> >  libavformat/mp3enc.c| 9 ++---
> >  3 files changed, 9 insertions(+), 4 deletions(-)
> >
> > diff --git a/libavcodec/libmp3lame.c b/libavcodec/libmp3lame.c
> > index 5642264..198ac94 100644
> > --- a/libavcodec/libmp3lame.c
> > +++ b/libavcodec/libmp3lame.c
> > @@ -137,7 +137,7 @@ static av_cold int
> mp3lame_encode_init(AVCodecContext *avctx)
> >  }
> >
> >  /* get encoder delay */
> > -avctx->initial_padding = lame_get_encoder_delay(s->gfp) + 528 + 1;
> > +avctx->initial_padding = lame_get_encoder_delay(s->gfp);
> >  ff_af_queue_init(avctx, >afq);
> >
> >  avctx->frame_size  = lame_get_framesize(s->gfp);
>
> you are changing a field of the public API
> changing public API without major version bumps is tricky, we dont want
> to break applications linkng to a newer lib
>
> is there software that uses this?
> software that would break if this is applied ? (or maybe it wuld fix
> some software usig it)
>
> If this is a bugfix it should be documented in APIChanges with
> minor version bumps, any available references to specifications
> should be added too
>
> Such bugfix should also be seperate of other unrelated changes
>
>
> [...]
> --
> Michael GnuPG fingerprint: 9FF2128B147EF6730BADF133611EC787040B0FAB
>
> I have often repented speaking, but never of holding my tongue.
> -- Xenocrates
>
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>
>
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[FFmpeg-devel] [PATCH 3/3] ffmpeg: copy trailing_padding when using -acodec copy

[Jon Toohill]

---
 ffmpeg.c | 1 +
 1 file changed, 1 insertion(+)

diff --git a/ffmpeg.c b/ffmpeg.c
index 652774f..442f818 100644
--- a/ffmpeg.c
+++ b/ffmpeg.c
@@ -3001,6 +3001,7 @@ static int transcode_init(void)
 enc_ctx->audio_service_type = dec_ctx->audio_service_type;
 enc_ctx->block_align= dec_ctx->block_align;
 enc_ctx->initial_padding= dec_ctx->delay;
+enc_ctx->trailing_padding   = dec_ctx->trailing_padding;
 enc_ctx->profile= dec_ctx->profile;
 #if FF_API_AUDIOENC_DELAY
 enc_ctx->delay  = dec_ctx->delay;
-- 
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[FFmpeg-devel] [PATCH 1/3] lavf/mp3dec: pass Xing gapless metadata to AVCodecParameters

[Jon Toohill]

Also removes decoder delay compensation from libmp3lame and mp3enc.
initial_padding specifies only encoder delay, decoder delay is
handled by start_skip_samples.
---
 libavcodec/libmp3lame.c | 2 +-
 libavformat/mp3dec.c| 2 ++
 libavformat/mp3enc.c| 9 ++---
 3 files changed, 9 insertions(+), 4 deletions(-)

diff --git a/libavcodec/libmp3lame.c b/libavcodec/libmp3lame.c
index 5642264..198ac94 100644
--- a/libavcodec/libmp3lame.c
+++ b/libavcodec/libmp3lame.c
@@ -137,7 +137,7 @@ static av_cold int mp3lame_encode_init(AVCodecContext 
*avctx)
 }
 
 /* get encoder delay */
-avctx->initial_padding = lame_get_encoder_delay(s->gfp) + 528 + 1;
+avctx->initial_padding = lame_get_encoder_delay(s->gfp);
 ff_af_queue_init(avctx, >afq);
 
 avctx->frame_size  = lame_get_framesize(s->gfp);
diff --git a/libavformat/mp3dec.c b/libavformat/mp3dec.c
index 56c7f8c..345fa88 100644
--- a/libavformat/mp3dec.c
+++ b/libavformat/mp3dec.c
@@ -239,6 +239,8 @@ static void mp3_parse_info_tag(AVFormatContext *s, AVStream 
*st,
 
 mp3->start_pad = v>>12;
 mp3->  end_pad = v&4095;
+st->codecpar->initial_padding = mp3->start_pad;
+st->codecpar->trailing_padding = mp3->end_pad;
 st->start_skip_samples = mp3->start_pad + 528 + 1;
 if (mp3->frames) {
 st->first_discard_sample = -mp3->end_pad + 528 + 1 + mp3->frames * 
(int64_t)spf;
diff --git a/libavformat/mp3enc.c b/libavformat/mp3enc.c
index de63401..da70d13 100644
--- a/libavformat/mp3enc.c
+++ b/libavformat/mp3enc.c
@@ -248,11 +248,14 @@ static int mp3_write_xing(AVFormatContext *s)
 avio_w8(dyn_ctx, 0);  // unknown encoding flags
 avio_w8(dyn_ctx, 0);  // unknown abr/minimal bitrate
 
-// encoder delay
-if (par->initial_padding - 528 - 1 >= 1 << 12) {
+// encoder delay/padding
+if (par->initial_padding >= 1 << 12) {
 av_log(s, AV_LOG_WARNING, "Too many samples of initial padding.\n");
 }
-avio_wb24(dyn_ctx, FFMAX(par->initial_padding - 528 - 1, 0)<<12);
+if (par->trailing_padding >= 1 << 12) {
+av_log(s, AV_LOG_WARNING, "Too many samples of trailing padding.\n");
+}
+avio_wb24(dyn_ctx, (par->initial_padding << 12) | (par->trailing_padding & 
0xFFF));
 
 avio_w8(dyn_ctx,   0); // misc
 avio_w8(dyn_ctx,   0); // mp3gain
-- 
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[FFmpeg-devel] [PATCH 2/3] lavc: show gapless info in stream summary

[Jon Toohill]

Also adds trailing_padding to AVCodecContext to match
AVCodecParameters so that it doesn't get lost when mapping
between them.
---
 doc/APIchanges   |  4 
 libavcodec/avcodec.h | 11 +++
 libavcodec/utils.c   | 40 +++-
 libavcodec/version.h |  2 +-
 4 files changed, 39 insertions(+), 18 deletions(-)

diff --git a/doc/APIchanges b/doc/APIchanges
index d777dc0..6092419 100644
--- a/doc/APIchanges
+++ b/doc/APIchanges
@@ -15,6 +15,10 @@ libavutil: 2015-08-28
 
 API changes, most recent first:
 
+2016-05-25 - xxx - lavc 57.47.100 - avcodec.h
+  Add trailing_padding to AVCodecContext to match the corresponding
+  field in AVCodecParameters.
+
 2016-04-27 - xxx - lavu 55.23.100 - log.h
   Add a new function av_log_format_line2() which returns number of bytes
   written to the target buffer.
diff --git a/libavcodec/avcodec.h b/libavcodec/avcodec.h
index 0181eb1..700c9b8 100644
--- a/libavcodec/avcodec.h
+++ b/libavcodec/avcodec.h
@@ -3504,6 +3504,17 @@ typedef struct AVCodecContext {
 #define FF_SUB_TEXT_FMT_ASS_WITH_TIMINGS 1
 #endif
 
+/**
+ * Audio only. The amount of padding (in samples) appended by the encoder 
to
+ * the end of the audio. I.e. this number of decoded samples must be
+ * discarded by the caller from the end of the stream to get the original
+ * audio without any trailing padding.
+ *
+ * - decoding: unused
+ * - encoding: unused
+ */
+int trailing_padding;
+
 } AVCodecContext;
 
 AVRational av_codec_get_pkt_timebase (const AVCodecContext *avctx);
diff --git a/libavcodec/utils.c b/libavcodec/utils.c
index 402a9d8..481e09c 100644
--- a/libavcodec/utils.c
+++ b/libavcodec/utils.c
@@ -3252,6 +3252,10 @@ void avcodec_string(char *buf, int buf_size, 
AVCodecContext *enc, int encode)
 && enc->bits_per_raw_sample != 
av_get_bytes_per_sample(enc->sample_fmt) * 8)
 snprintf(buf + strlen(buf), buf_size - strlen(buf),
  " (%d bit)", enc->bits_per_raw_sample);
+if (enc->initial_padding || enc->trailing_padding) {
+snprintf(buf + strlen(buf), buf_size - strlen(buf),
+ ", delay %d, padding %d", enc->initial_padding, 
enc->trailing_padding);
+}
 break;
 case AVMEDIA_TYPE_DATA:
 if (av_log_get_level() >= AV_LOG_DEBUG) {
@@ -4097,14 +4101,15 @@ int avcodec_parameters_from_context(AVCodecParameters 
*par,
 par->video_delay = codec->has_b_frames;
 break;
 case AVMEDIA_TYPE_AUDIO:
-par->format  = codec->sample_fmt;
-par->channel_layout  = codec->channel_layout;
-par->channels= codec->channels;
-par->sample_rate = codec->sample_rate;
-par->block_align = codec->block_align;
-par->frame_size  = codec->frame_size;
-par->initial_padding = codec->initial_padding;
-par->seek_preroll= codec->seek_preroll;
+par->format   = codec->sample_fmt;
+par->channel_layout   = codec->channel_layout;
+par->channels = codec->channels;
+par->sample_rate  = codec->sample_rate;
+par->block_align  = codec->block_align;
+par->frame_size   = codec->frame_size;
+par->initial_padding  = codec->initial_padding;
+par->trailing_padding = codec->trailing_padding;
+par->seek_preroll = codec->seek_preroll;
 break;
 case AVMEDIA_TYPE_SUBTITLE:
 par->width  = codec->width;
@@ -4151,15 +4156,16 @@ int avcodec_parameters_to_context(AVCodecContext *codec,
 codec->has_b_frames   = par->video_delay;
 break;
 case AVMEDIA_TYPE_AUDIO:
-codec->sample_fmt  = par->format;
-codec->channel_layout  = par->channel_layout;
-codec->channels= par->channels;
-codec->sample_rate = par->sample_rate;
-codec->block_align = par->block_align;
-codec->frame_size  = par->frame_size;
-codec->delay   =
-codec->initial_padding = par->initial_padding;
-codec->seek_preroll= par->seek_preroll;
+codec->sample_fmt   = par->format;
+codec->channel_layout   = par->channel_layout;
+codec->channels = par->channels;
+codec->sample_rate  = par->sample_rate;
+codec->block_align  = par->block_align;
+codec->frame_size   = par->frame_size;
+codec->delay=
+codec->initial_padding  = par->initial_padding;
+codec->trailing_padding = par->trailing_padding;
+codec->seek_preroll = par->seek_preroll;
 break;
 case AVMEDIA_TYPE_SUBTITLE:
 codec->width  = par->width;
diff --git a/libavcodec/version.h b/libavcodec/version.h
index 0852b43..37a6e17 100644
--- a/libavcodec/version.h
+++ b/libavcodec/version.h
@@ -28,7 +28,7 @@
 #include "libavutil/version.h"
 
 #define 

[FFmpeg-devel] [PATCH 0/3] Pass Xing gapless metadata to users during mp3 parsing

[Jon Toohill]

These patches expose the encoder delay/padding parsed from an mp3's Xing header 
to users of lavc/lavf, and show gapless info in the stream summary string. They 
also change ffmpeg to pass Xing gapless metadata from input to output when 
using -acodec copy.

trailing_padding is still not set properly when encoding with libmp3lame, 
causing an encode/decode round trip to add trailing silence. This is not a 
regression from current behavior, and will be addressed in a separate patch set.

Jon Toohill (3):
  lavf/mp3dec: pass Xing gapless metadata to AVCodecParameters
  lavc: show gapless info in stream summary
  ffmpeg: copy trailing_padding when using -acodec copy

 doc/APIchanges  |  4 
 ffmpeg.c|  1 +
 libavcodec/avcodec.h| 11 +++
 libavcodec/libmp3lame.c |  2 +-
 libavcodec/utils.c  | 40 +++-
 libavcodec/version.h|  2 +-
 libavformat/mp3dec.c|  2 ++
 libavformat/mp3enc.c|  9 ++---
 8 files changed, 49 insertions(+), 22 deletions(-)

-- 
2.8.0.rc3.226.g39d4020

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Re: [FFmpeg-devel] [PATCH 1/2] lavf/mp3dec: pass Xing gapless metadata to AVCodecParameters

[Jon Toohill]

Michael et al., is this good to merge as-is? I just tested and a round trip
with ffmpeg from wav -> mp3 -> wav retains the correct number of samples.


Jon Toohill |  Google Play Music |  jtooh...@google.com |  (650) 215-0770

On Wed, Jun 1, 2016 at 5:58 PM, Jon Toohill <jtooh...@google.com> wrote:

> Based on my understanding of [1], these values in the Info tag specify
> only the encoder delay/padding, which matches the documentation for these
> fields. It looks like other formats are using the fields that way as well.
>
> I think the extra 528 + 1 samples are the decoder delay [2]. It looks like
> libmp3lame adds the 528 + 1 only to have mp3dec subtract it, so I'm not
> sure why that's done. IIUC start_skip_samples is the mechanism that
> actually accounts for the extra delay when decoding.
>
> [1]: http://gabriel.mp3-tech.org/mp3infotag.html#delays
> [2]: http://lame.sourceforge.net/tech-FAQ.txt
>
>
>
> Jon Toohill |  Google Play Music |  jtooh...@google.com |  (650) 215-0770
>
> On Thu, May 26, 2016 at 7:51 PM, Michael Niedermayer <
> mich...@niedermayer.cc> wrote:
>
>> On Wed, May 25, 2016 at 09:56:59AM -0700, Jon Toohill wrote:
>> > ---
>> >  libavformat/mp3dec.c | 2 ++
>> >  1 file changed, 2 insertions(+)
>> >
>> > diff --git a/libavformat/mp3dec.c b/libavformat/mp3dec.c
>> > index 3725d67..192f5ef 100644
>> > --- a/libavformat/mp3dec.c
>> > +++ b/libavformat/mp3dec.c
>> > @@ -234,6 +234,8 @@ static void mp3_parse_info_tag(AVFormatContext *s,
>> AVStream *st,
>> >
>> >  mp3->start_pad = v>>12;
>> >  mp3->  end_pad = v&4095;
>> > +st->codecpar->initial_padding = mp3->start_pad;
>> > +st->codecpar->trailing_padding = mp3->end_pad;
>> >  st->start_skip_samples = mp3->start_pad + 528 + 1;
>> >  if (mp3->frames) {
>> >  st->first_discard_sample = -mp3->end_pad + 528 + 1 +
>> mp3->frames * (int64_t)spf;
>>
>> is the 528 + 1 difference intended to
>> start_skip_samples/first_discard_sample
>> ?
>> mp3enc stores par->initial_padding - 528 - 1
>>
>> [...]
>>
>> --
>> Michael GnuPG fingerprint: 9FF2128B147EF6730BADF133611EC787040B0FAB
>>
>> If you drop bombs on a foreign country and kill a hundred thousand
>> innocent people, expect your government to call the consequence
>> "unprovoked inhuman terrorist attacks" and use it to justify dropping
>> more bombs and killing more people. The technology changed, the idea is
>> old.
>>
>> ___
>> ffmpeg-devel mailing list
>> ffmpeg-devel@ffmpeg.org
>> http://ffmpeg.org/mailman/listinfo/ffmpeg-devel
>>
>>
>
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Re: [FFmpeg-devel] [PATCH 1/2] lavf/mp3dec: pass Xing gapless metadata to AVCodecParameters

[Jon Toohill]

Based on my understanding of [1], these values in the Info tag specify only
the encoder delay/padding, which matches the documentation for these
fields. It looks like other formats are using the fields that way as well.

I think the extra 528 + 1 samples are the decoder delay [2]. It looks like
libmp3lame adds the 528 + 1 only to have mp3dec subtract it, so I'm not
sure why that's done. IIUC start_skip_samples is the mechanism that
actually accounts for the extra delay when decoding.

[1]: http://gabriel.mp3-tech.org/mp3infotag.html#delays
[2]: http://lame.sourceforge.net/tech-FAQ.txt



Jon Toohill |  Google Play Music |  jtooh...@google.com |  (650) 215-0770

On Thu, May 26, 2016 at 7:51 PM, Michael Niedermayer <mich...@niedermayer.cc
> wrote:

> On Wed, May 25, 2016 at 09:56:59AM -0700, Jon Toohill wrote:
> > ---
> >  libavformat/mp3dec.c | 2 ++
> >  1 file changed, 2 insertions(+)
> >
> > diff --git a/libavformat/mp3dec.c b/libavformat/mp3dec.c
> > index 3725d67..192f5ef 100644
> > --- a/libavformat/mp3dec.c
> > +++ b/libavformat/mp3dec.c
> > @@ -234,6 +234,8 @@ static void mp3_parse_info_tag(AVFormatContext *s,
> AVStream *st,
> >
> >  mp3->start_pad = v>>12;
> >  mp3->  end_pad = v&4095;
> > +st->codecpar->initial_padding = mp3->start_pad;
> > +st->codecpar->trailing_padding = mp3->end_pad;
> >  st->start_skip_samples = mp3->start_pad + 528 + 1;
> >  if (mp3->frames) {
> >  st->first_discard_sample = -mp3->end_pad + 528 + 1 +
> mp3->frames * (int64_t)spf;
>
> is the 528 + 1 difference intended to
> start_skip_samples/first_discard_sample
> ?
> mp3enc stores par->initial_padding - 528 - 1
>
> [...]
>
> --
> Michael GnuPG fingerprint: 9FF2128B147EF6730BADF133611EC787040B0FAB
>
> If you drop bombs on a foreign country and kill a hundred thousand
> innocent people, expect your government to call the consequence
> "unprovoked inhuman terrorist attacks" and use it to justify dropping
> more bombs and killing more people. The technology changed, the idea is
> old.
>
> ___
> ffmpeg-devel mailing list
> ffmpeg-devel@ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-devel
>
>
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[FFmpeg-devel] [PATCH 1/2] lavf/mp3dec: pass Xing gapless metadata to AVCodecParameters

[Jon Toohill]

---
 libavformat/mp3dec.c | 2 ++
 1 file changed, 2 insertions(+)

diff --git a/libavformat/mp3dec.c b/libavformat/mp3dec.c
index 3725d67..192f5ef 100644
--- a/libavformat/mp3dec.c
+++ b/libavformat/mp3dec.c
@@ -234,6 +234,8 @@ static void mp3_parse_info_tag(AVFormatContext *s, AVStream 
*st,
 
 mp3->start_pad = v>>12;
 mp3->  end_pad = v&4095;
+st->codecpar->initial_padding = mp3->start_pad;
+st->codecpar->trailing_padding = mp3->end_pad;
 st->start_skip_samples = mp3->start_pad + 528 + 1;
 if (mp3->frames) {
 st->first_discard_sample = -mp3->end_pad + 528 + 1 + mp3->frames * 
(int64_t)spf;
-- 
2.8.0.rc3.226.g39d4020

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[FFmpeg-devel] [PATCH 0/2] Pass Xing gapless metadata to users during mp3 parsing

[Jon Toohill]

These patches expose the encoder delay/padding parsed from an mp3's Xing header 
to users of lavc/lavf, and show gapless info in the stream summary string.

Jon Toohill (2):
  lavf/mp3dec: pass Xing gapless metadata to AVCodecParameters
  lavc: show gapless info in stream summary

 doc/APIchanges   |  4 
 libavcodec/avcodec.h | 11 +++
 libavcodec/utils.c   | 38 ++
 libavcodec/version.h |  2 +-
 libavformat/mp3dec.c |  2 ++
 5 files changed, 40 insertions(+), 17 deletions(-)

-- 
2.8.0.rc3.226.g39d4020

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[FFmpeg-devel] [PATCH 2/2] lavc: show gapless info in stream summary

[Jon Toohill]

Also adds trailing_padding to AVCodecContext to match
AVCodecParameters so that it doesn't get lost when mapping
between them.
---
 doc/APIchanges   |  4 
 libavcodec/avcodec.h | 11 +++
 libavcodec/utils.c   | 38 ++
 libavcodec/version.h |  2 +-
 4 files changed, 38 insertions(+), 17 deletions(-)

diff --git a/doc/APIchanges b/doc/APIchanges
index d777dc0..4720b70 100644
--- a/doc/APIchanges
+++ b/doc/APIchanges
@@ -15,6 +15,10 @@ libavutil: 2015-08-28
 
 API changes, most recent first:
 
+2016-05-25 - xxx - lavc 57.43.100 - avcodec.h
+  Add trailing_padding to AVCodecContext to match the corresponding
+  field in AVCodecParameters.
+
 2016-04-27 - xxx - lavu 55.23.100 - log.h
   Add a new function av_log_format_line2() which returns number of bytes
   written to the target buffer.
diff --git a/libavcodec/avcodec.h b/libavcodec/avcodec.h
index 9ec9adf..554e1ee 100644
--- a/libavcodec/avcodec.h
+++ b/libavcodec/avcodec.h
@@ -3497,6 +3497,17 @@ typedef struct AVCodecContext {
 #define FF_SUB_TEXT_FMT_ASS_WITH_TIMINGS 1
 #endif
 
+/**
+ * Audio only. The amount of padding (in samples) appended by the encoder 
to
+ * the end of the audio. I.e. this number of decoded samples must be
+ * discarded by the caller from the end of the stream to get the original
+ * audio without any trailing padding.
+ *
+ * - decoding: unused
+ * - encoding: unused
+ */
+int trailing_padding;
+
 } AVCodecContext;
 
 AVRational av_codec_get_pkt_timebase (const AVCodecContext *avctx);
diff --git a/libavcodec/utils.c b/libavcodec/utils.c
index e5a832b..51f50b0 100644
--- a/libavcodec/utils.c
+++ b/libavcodec/utils.c
@@ -3251,6 +3251,10 @@ void avcodec_string(char *buf, int buf_size, 
AVCodecContext *enc, int encode)
 && enc->bits_per_raw_sample != 
av_get_bytes_per_sample(enc->sample_fmt) * 8)
 snprintf(buf + strlen(buf), buf_size - strlen(buf),
  " (%d bit)", enc->bits_per_raw_sample);
+if (enc->initial_padding || enc->trailing_padding) {
+snprintf(buf + strlen(buf), buf_size - strlen(buf),
+ ", delay %d, padding %d", enc->initial_padding, 
enc->trailing_padding);
+}
 break;
 case AVMEDIA_TYPE_DATA:
 if (av_log_get_level() >= AV_LOG_DEBUG) {
@@ -4094,14 +4098,15 @@ int avcodec_parameters_from_context(AVCodecParameters 
*par,
 par->video_delay = codec->has_b_frames;
 break;
 case AVMEDIA_TYPE_AUDIO:
-par->format  = codec->sample_fmt;
-par->channel_layout  = codec->channel_layout;
-par->channels= codec->channels;
-par->sample_rate = codec->sample_rate;
-par->block_align = codec->block_align;
-par->frame_size  = codec->frame_size;
-par->initial_padding = codec->initial_padding;
-par->seek_preroll= codec->seek_preroll;
+par->format   = codec->sample_fmt;
+par->channel_layout   = codec->channel_layout;
+par->channels = codec->channels;
+par->sample_rate  = codec->sample_rate;
+par->block_align  = codec->block_align;
+par->frame_size   = codec->frame_size;
+par->initial_padding  = codec->initial_padding;
+par->trailing_padding = codec->trailing_padding;
+par->seek_preroll = codec->seek_preroll;
 break;
 case AVMEDIA_TYPE_SUBTITLE:
 par->width  = codec->width;
@@ -4148,14 +4153,15 @@ int avcodec_parameters_to_context(AVCodecContext *codec,
 codec->has_b_frames   = par->video_delay;
 break;
 case AVMEDIA_TYPE_AUDIO:
-codec->sample_fmt  = par->format;
-codec->channel_layout  = par->channel_layout;
-codec->channels= par->channels;
-codec->sample_rate = par->sample_rate;
-codec->block_align = par->block_align;
-codec->frame_size  = par->frame_size;
-codec->initial_padding = par->initial_padding;
-codec->seek_preroll= par->seek_preroll;
+codec->sample_fmt   = par->format;
+codec->channel_layout   = par->channel_layout;
+codec->channels = par->channels;
+codec->sample_rate  = par->sample_rate;
+codec->block_align  = par->block_align;
+codec->frame_size   = par->frame_size;
+codec->initial_padding  = par->initial_padding;
+codec->trailing_padding = par->trailing_padding;
+codec->seek_preroll = par->seek_preroll;
 break;
 case AVMEDIA_TYPE_SUBTITLE:
 codec->width  = par->width;
diff --git a/libavcodec/version.h b/libavcodec/version.h
index 0916f81..ee3006c 100644
--- a/libavcodec/version.h
+++ b/libavcodec/version.h
@@ -28,7 +28,7 @@
 #include "libavutil/version.h"
 
 #define LIBAVCODEC_VERSION_MAJOR  57
-#define LIBAVCODEC_VERSION_MINOR  42

Re: [FFmpeg-devel] [PATCH 0/2] Pass Xing gapless metadata to users during mp3 parsing

[Jon Toohill]

Forgot to fix the author line in those patches, it should read "Jon Toohill
<jtooh...@google.com>" instead. I can fix it when sending updated patches
if there are revisions to make, otherwise can you change it before merging
(or should I send updated patches)?


Jon Toohill |  Google Play Music |  jtooh...@google.com |  (650) 215-0770

On Tue, May 24, 2016 at 3:52 PM, Jon Toohill <jtooh...@google.com> wrote:

> These patches expose the encoder delay/padding parsed from an mp3's Xing
> header to users of lavc/lavf, and show gapless info in the stream summary
> string.
>
> Jon Toohill (2):
>   lavf/mp3dec: pass Xing gapless metadata to AVCodecParameters
>   lavc: show gapless info in stream summary
>
>  libavcodec/avcodec.h | 11 +++
>  libavcodec/utils.c   | 38 ++
>  libavformat/mp3dec.c |  2 ++
>  3 files changed, 35 insertions(+), 16 deletions(-)
>
> --
> 2.8.0.rc3.226.g39d4020
>
>
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[FFmpeg-devel] [PATCH 2/2] lavc: show gapless info in stream summary

[Jon Toohill]

From: Jon Toohill <jon.tooh...@gmail.com>

Also adds trailing_padding to AVCodecContext to match
AVCodecParameters so that it doesn't get lost when mapping
between them.
---
 libavcodec/avcodec.h | 11 +++
 libavcodec/utils.c   | 38 ++
 2 files changed, 33 insertions(+), 16 deletions(-)

diff --git a/libavcodec/avcodec.h b/libavcodec/avcodec.h
index 9ec9adf..408efe1 100644
--- a/libavcodec/avcodec.h
+++ b/libavcodec/avcodec.h
@@ -3321,6 +3321,17 @@ typedef struct AVCodecContext {
 int initial_padding;
 
 /**
+ * Audio only. The amount of padding (in samples) appended by the encoder 
to
+ * the end of the audio. I.e. this number of decoded samples must be
+ * discarded by the caller from the end of the stream to get the original
+ * audio without any trailing padding.
+ *
+ * - decoding: unused
+ * - encoding: unused
+ */
+int trailing_padding;
+
+/**
  * - decoding: For codecs that store a framerate value in the compressed
  * bitstream, the decoder may export it here. { 0, 1} when
  * unknown.
diff --git a/libavcodec/utils.c b/libavcodec/utils.c
index e5a832b..51f50b0 100644
--- a/libavcodec/utils.c
+++ b/libavcodec/utils.c
@@ -3251,6 +3251,10 @@ void avcodec_string(char *buf, int buf_size, 
AVCodecContext *enc, int encode)
 && enc->bits_per_raw_sample != 
av_get_bytes_per_sample(enc->sample_fmt) * 8)
 snprintf(buf + strlen(buf), buf_size - strlen(buf),
  " (%d bit)", enc->bits_per_raw_sample);
+if (enc->initial_padding || enc->trailing_padding) {
+snprintf(buf + strlen(buf), buf_size - strlen(buf),
+ ", delay %d, padding %d", enc->initial_padding, 
enc->trailing_padding);
+}
 break;
 case AVMEDIA_TYPE_DATA:
 if (av_log_get_level() >= AV_LOG_DEBUG) {
@@ -4094,14 +4098,15 @@ int avcodec_parameters_from_context(AVCodecParameters 
*par,
 par->video_delay = codec->has_b_frames;
 break;
 case AVMEDIA_TYPE_AUDIO:
-par->format  = codec->sample_fmt;
-par->channel_layout  = codec->channel_layout;
-par->channels= codec->channels;
-par->sample_rate = codec->sample_rate;
-par->block_align = codec->block_align;
-par->frame_size  = codec->frame_size;
-par->initial_padding = codec->initial_padding;
-par->seek_preroll= codec->seek_preroll;
+par->format   = codec->sample_fmt;
+par->channel_layout   = codec->channel_layout;
+par->channels = codec->channels;
+par->sample_rate  = codec->sample_rate;
+par->block_align  = codec->block_align;
+par->frame_size   = codec->frame_size;
+par->initial_padding  = codec->initial_padding;
+par->trailing_padding = codec->trailing_padding;
+par->seek_preroll = codec->seek_preroll;
 break;
 case AVMEDIA_TYPE_SUBTITLE:
 par->width  = codec->width;
@@ -4148,14 +4153,15 @@ int avcodec_parameters_to_context(AVCodecContext *codec,
 codec->has_b_frames   = par->video_delay;
 break;
 case AVMEDIA_TYPE_AUDIO:
-codec->sample_fmt  = par->format;
-codec->channel_layout  = par->channel_layout;
-codec->channels= par->channels;
-codec->sample_rate = par->sample_rate;
-codec->block_align = par->block_align;
-codec->frame_size  = par->frame_size;
-codec->initial_padding = par->initial_padding;
-codec->seek_preroll= par->seek_preroll;
+codec->sample_fmt   = par->format;
+codec->channel_layout   = par->channel_layout;
+codec->channels = par->channels;
+codec->sample_rate  = par->sample_rate;
+codec->block_align  = par->block_align;
+codec->frame_size   = par->frame_size;
+codec->initial_padding  = par->initial_padding;
+codec->trailing_padding = par->trailing_padding;
+codec->seek_preroll = par->seek_preroll;
 break;
 case AVMEDIA_TYPE_SUBTITLE:
 codec->width  = par->width;
-- 
2.8.0.rc3.226.g39d4020

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[FFmpeg-devel] [PATCH 1/2] lavf/mp3dec: pass Xing gapless metadata to AVCodecParameters

[Jon Toohill]

From: Jon Toohill <jon.tooh...@gmail.com>

---
 libavformat/mp3dec.c | 2 ++
 1 file changed, 2 insertions(+)

diff --git a/libavformat/mp3dec.c b/libavformat/mp3dec.c
index 3725d67..192f5ef 100644
--- a/libavformat/mp3dec.c
+++ b/libavformat/mp3dec.c
@@ -234,6 +234,8 @@ static void mp3_parse_info_tag(AVFormatContext *s, AVStream 
*st,
 
 mp3->start_pad = v>>12;
 mp3->  end_pad = v&4095;
+st->codecpar->initial_padding = mp3->start_pad;
+st->codecpar->trailing_padding = mp3->end_pad;
 st->start_skip_samples = mp3->start_pad + 528 + 1;
 if (mp3->frames) {
 st->first_discard_sample = -mp3->end_pad + 528 + 1 + mp3->frames * 
(int64_t)spf;
-- 
2.8.0.rc3.226.g39d4020

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[FFmpeg-devel] [PATCH 0/2] Pass Xing gapless metadata to users during mp3 parsing

[Jon Toohill]

These patches expose the encoder delay/padding parsed from an mp3's Xing header 
to users of lavc/lavf, and show gapless info in the stream summary string.

Jon Toohill (2):
  lavf/mp3dec: pass Xing gapless metadata to AVCodecParameters
  lavc: show gapless info in stream summary

 libavcodec/avcodec.h | 11 +++
 libavcodec/utils.c   | 38 ++
 libavformat/mp3dec.c |  2 ++
 3 files changed, 35 insertions(+), 16 deletions(-)

-- 
2.8.0.rc3.226.g39d4020

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[FFmpeg-devel] libavformat/movenc.c: Correct color range when writing DNxHD atoms

[jon morley]

From 0097277471810ab1d9d737c64a57c2278a039153 Mon Sep 17 00:00:00 2001
From: Jon Morley j...@tweaksoftware.com
Date: Tue, 27 Jan 2015 11:10:27 -0800
Subject: [PATCH] libavformat/movenc.c: Correct color range when writing DNxHD
 atoms

The meaning of the color range values in the AVdn.ACLR atom was swapped.
This change makes the selection explicit and correctly mapped.
---
 libavformat/movenc.c | 9 ++---
 1 file changed, 6 insertions(+), 3 deletions(-)

diff --git a/libavformat/movenc.c b/libavformat/movenc.c
index d7ae5f0..dfe4c27 100644
--- a/libavformat/movenc.c
+++ b/libavformat/movenc.c
@@ -1039,10 +1039,13 @@ static int mov_write_avid_tag(AVIOContext *pb, MOVTrack 
*track)
 ffio_wfourcc(pb, ACLR);
 ffio_wfourcc(pb, 0001);
 if (track-enc-color_range == AVCOL_RANGE_MPEG) { /* Legal range (16-235) 
*/
-avio_wb32(pb, 1); /* Corresponds to 709 in official encoder */
-} else { /* Full range (0-255) */
-avio_wb32(pb, 2); /* Corresponds to RGB in official encoder */
+avio_wb32(pb, 2); /* Corresponds to 709 in official encoder */
+} else if (track-enc-color_range == AVCOL_RANGE_JPEG) { /* Full range 
(0-255) */
+avio_wb32(pb, 1); /* Corresponds to RGB in official encoder */
+} else {
+avio_wb32(pb, 0); /* Unspecified */
 }
+
 avio_wb32(pb, 0); /* unknown */
 
 avio_wb32(pb, 24); /* size */
-- 
1.8.5.2 (Apple Git-48)
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Re: [FFmpeg-devel] Question about supported_fps in libavutil/timecode.c::check_fps

[jon morley]

Hi Clément,

I am sorry I was rude. That was not my intention. I was attempting to 
follow these directions from the ffmpeg.org page:


You can use the FFmpeg libraries in your commercial program, but you 
are encouraged to publish any patch you make. In this case the best way 
to proceed is to send your patches to the ffmpeg-devel mailing list 
following the guidelines illustrated in the remainder of this document.


I will stick to mailing patches exclusively in the future.

Patches reflecting your suggestions attached.

Sincerely,
Jon

On 1/24/15 8:21 AM, Clément Bœsch wrote:

On Sat, Jan 24, 2015 at 07:40:38AM -0800, jon morley wrote:

Hi Clément,



Hi,


That is a good point! I am attaching an additional patch to remove those
cases even before entering the mod test loop.

Now the logic should look like this:

static int check_fps(int fps)
{



 if (fps = 0) return -1;

 int i;
 static const int supported_fps_bases[] = {24, 25, 30};


You can't put statements before declarations, some compilers will choke on
it.

Also, please squash it with the previous patch since it wasn't applied
yet.



 for (i = 0; i  FF_ARRAY_ELEMS(supported_fps_bases); i++)
 if (fps % supported_fps_bases[i] == 0)
 return 0;
 return -1;
}

I am still really curious to know if switching to this division (modulo)
test breaks the spirit of check_fps. I could not find anywhere that
benefited from the explicit list the method currently used, but that doesn't
mean it isn't out there.


I'm more concerned about how the rest of the code will behave. Typically,
av_timecode_adjust_ntsc_framenum2() could benefit from some improvements
(checking if fps % 30, and deducing drop_frames and frames_per_10mins
accordingly) if you allow such thing. Then you might need to adjust
check_timecode() as well to allow the drop frame for the other % 30.



Thanks,
Jon



[...]

Note: please do not top post on this mailing list, it is considered rude.

Regards,



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From 95f1fb3695f086de1baa301015985742d688a159 Mon Sep 17 00:00:00 2001
From: Jon Morley j...@tweaksoftware.com
Date: Sat, 24 Jan 2015 12:18:50 -0800
Subject: [PATCH] libavutil/timecode.c: Add support for frame rates beyond 60
 fps

Instead of looking for specifically supported frame rates this
collection of changes checks to see if the given rates are evenly
divisible by supported common factors.
---
 libavutil/timecode.c | 33 -
 1 file changed, 16 insertions(+), 17 deletions(-)

diff --git a/libavutil/timecode.c b/libavutil/timecode.c
index 1dfd040..c2469c0 100644
--- a/libavutil/timecode.c
+++ b/libavutil/timecode.c
@@ -33,18 +33,15 @@
 
 int av_timecode_adjust_ntsc_framenum2(int framenum, int fps)
 {
-/* only works for NTSC 29.97 and 59.94 */
+int factor = 1;
 int drop_frames = 0;
 int d, m, frames_per_10mins;
 
-if (fps == 30) {
-drop_frames = 2;
-frames_per_10mins = 17982;
-} else if (fps == 60) {
-drop_frames = 4;
-frames_per_10mins = 35964;
-} else
-return framenum;
+if (fps  30 || fps % 30 != 0) return framenum;
+
+factor = fps / 30;
+drop_frames = factor * 2;
+frames_per_10mins = factor * 17982;
 
 d = framenum / frames_per_10mins;
 m = framenum % frames_per_10mins;
@@ -141,10 +138,12 @@ char *av_timecode_make_mpeg_tc_string(char *buf, uint32_t 
tc25bit)
 static int check_fps(int fps)
 {
 int i;
-static const int supported_fps[] = {24, 25, 30, 48, 50, 60};
+static const int supported_fps_multiples[] = {24, 25, 30};
+
+if (fps = 0) return -1;
 
-for (i = 0; i  FF_ARRAY_ELEMS(supported_fps); i++)
-if (fps == supported_fps[i])
+for (i = 0; i  FF_ARRAY_ELEMS(supported_fps_multiples); i++)
+if (fps % supported_fps_multiples[i] == 0)
 return 0;
 return -1;
 }
@@ -155,8 +154,8 @@ static int check_timecode(void *log_ctx, AVTimecode *tc)
 av_log(log_ctx, AV_LOG_ERROR, Timecode frame rate must be 
specified\n);
 return AVERROR(EINVAL);
 }
-if ((tc-flags  AV_TIMECODE_FLAG_DROPFRAME)  tc-fps != 30  tc-fps 
!= 60) {
-av_log(log_ctx, AV_LOG_ERROR, Drop frame is only allowed with 
3/1001 or 6/1001 FPS\n);
+if ((tc-flags  AV_TIMECODE_FLAG_DROPFRAME)  tc-fps % 30 != 0) {
+av_log(log_ctx, AV_LOG_ERROR, Drop frame is only allowed in frame 
rates evenly divisible by 30 FPS\n);
 return AVERROR(EINVAL);
 }
 if (check_fps(tc-fps)  0) {
@@ -201,9 +200,9 @@ int av_timecode_init_from_string(AVTimecode *tc, AVRational 
rate, const char *st
 }
 
 memset(tc, 0, sizeof(*tc));
-tc-flags = c != ':' ? AV_TIMECODE_FLAG_DROPFRAME : 0; // drop if ';', 
'.', ...
-tc-rate  = rate;
-tc-fps   = fps_from_frame_rate(rate);
+tc-flags  = c != ':' ? AV_TIMECODE_FLAG_DROPFRAME : 0

Re: [FFmpeg-devel] Question about supported_fps in libavutil/timecode.c::check_fps

[jon morley]

Hi Clément,

That is a good point! I am attaching an additional patch to remove those 
cases even before entering the mod test loop.


Now the logic should look like this:

static int check_fps(int fps)
{
if (fps = 0) return -1;

int i;
static const int supported_fps_bases[] = {24, 25, 30};

for (i = 0; i  FF_ARRAY_ELEMS(supported_fps_bases); i++)
if (fps % supported_fps_bases[i] == 0)
return 0;
return -1;
}

I am still really curious to know if switching to this division (modulo) 
test breaks the spirit of check_fps. I could not find anywhere that 
benefited from the explicit list the method currently used, but that 
doesn't mean it isn't out there.


Thanks,
Jon

On 1/24/15 2:27 AM, Clément Bœsch wrote:

On Fri, Jan 23, 2015 at 08:48:37AM -0800, jon morley wrote:

Patch attached for consideration.

On 1/23/15 8:03 AM, jon morley wrote:

Currently check_fps has the following logic:

static int check_fps(int fps)
{
 int i;
 static const int supported_fps[] = {24, 25, 30, 48, 50, 60};

 for (i = 0; i  FF_ARRAY_ELEMS(supported_fps); i++)
 if (fps == supported_fps[i])
 return 0;
 return -1;
}

I am starting to see more and more movies with fps rates in excess of
this list from modified GoPro files and other raw camera sources.

I was originally adding more entries as the sources came rolling in
because I could not see any issue in how this was getting called later
with that approach.

I still don't see the drawback of adding more, but I am tired of adding
a new rate every time I encounter one in the wild. I was curious if it
wouldn't make more sense to change the logic to the following:

static int check_fps(int fps)
{
 int i;
 static const int supported_fps_bases[] = {24, 25, 30};

 for (i = 0; i  FF_ARRAY_ELEMS(supported_fps_bases); i++)
 if (fps % supported_fps_bases[i] == 0)
 return 0;
 return -1;
}

If that makes sense to you, then I will submit a patch. Please let me
know if I have overlooked some other usage/meaning of check_fps that I
am overlooking.

Thanks,
Jon
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 From 73e5339ec76305d34214b5e84dc5a38673f784b7 Mon Sep 17 00:00:00 2001
From: Jon Morley j...@tweaksoftware.com
Date: Fri, 23 Jan 2015 08:43:33 -0800
Subject: [PATCH] libavutil/timecode.c: Extend check_fps logic to handle high
  frame rates

QuickTime sources continue to push higher and higher frame rates. This
change moves away from explictly testing incoming fps values towards
ensuring the incoming value is evenly divisable by 24, 25, or 30.
---
  libavutil/timecode.c | 6 +++---
  1 file changed, 3 insertions(+), 3 deletions(-)

diff --git a/libavutil/timecode.c b/libavutil/timecode.c
index 1dfd040..c10895c 100644
--- a/libavutil/timecode.c
+++ b/libavutil/timecode.c
@@ -141,10 +141,10 @@ char *av_timecode_make_mpeg_tc_string(char *buf, uint32_t 
tc25bit)
  static int check_fps(int fps)
  {
  int i;
-static const int supported_fps[] = {24, 25, 30, 48, 50, 60};
+static const int supported_fps_bases[] = {24, 25, 30};

-for (i = 0; i  FF_ARRAY_ELEMS(supported_fps); i++)
-if (fps == supported_fps[i])
+for (i = 0; i  FF_ARRAY_ELEMS(supported_fps_bases); i++)
+if (fps % supported_fps_bases[i] == 0)
  return 0;
  return -1;


I don't think you want to accept fps ≤ 0

[...]



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From 0a72d78992bbeb6c2536285397149cceb64b05d8 Mon Sep 17 00:00:00 2001
From: Jon Morley j...@tweaksoftware.com
Date: Sat, 24 Jan 2015 07:28:40 -0800
Subject: [PATCH 2/2] libavutil/timecode.c: check_fps must reject rates at or
 below zero

An earlier change to check_fps's logic which now confirms that the
incoming evaluation fps is evenly divisable by a list of supported
rates leaves open the possibility of accepting zero and negative frame
rates. This change removes that posibility.
---
 libavutil/timecode.c | 2 ++
 1 file changed, 2 insertions(+)

diff --git a/libavutil/timecode.c b/libavutil/timecode.c
index c10895c..446e2d5 100644
--- a/libavutil/timecode.c
+++ b/libavutil/timecode.c
@@ -140,6 +140,8 @@ char *av_timecode_make_mpeg_tc_string(char *buf, uint32_t 
tc25bit)
 
 static int check_fps(int fps)
 {
+if (fps = 0) return -1;
+
 int i;
 static const int supported_fps_bases[] = {24, 25, 30};
 
-- 
1.8.5.2 (Apple Git-48)

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Re: [FFmpeg-devel] Question about supported_fps in libavutil/timecode.c::check_fps

[jon morley]

Patch attached for consideration.

On 1/23/15 8:03 AM, jon morley wrote:

Currently check_fps has the following logic:

static int check_fps(int fps)
{
 int i;
 static const int supported_fps[] = {24, 25, 30, 48, 50, 60};

 for (i = 0; i  FF_ARRAY_ELEMS(supported_fps); i++)
 if (fps == supported_fps[i])
 return 0;
 return -1;
}

I am starting to see more and more movies with fps rates in excess of
this list from modified GoPro files and other raw camera sources.

I was originally adding more entries as the sources came rolling in
because I could not see any issue in how this was getting called later
with that approach.

I still don't see the drawback of adding more, but I am tired of adding
a new rate every time I encounter one in the wild. I was curious if it
wouldn't make more sense to change the logic to the following:

static int check_fps(int fps)
{
 int i;
 static const int supported_fps_bases[] = {24, 25, 30};

 for (i = 0; i  FF_ARRAY_ELEMS(supported_fps_bases); i++)
 if (fps % supported_fps_bases[i] == 0)
 return 0;
 return -1;
}

If that makes sense to you, then I will submit a patch. Please let me
know if I have overlooked some other usage/meaning of check_fps that I
am overlooking.

Thanks,
Jon
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From 73e5339ec76305d34214b5e84dc5a38673f784b7 Mon Sep 17 00:00:00 2001
From: Jon Morley j...@tweaksoftware.com
Date: Fri, 23 Jan 2015 08:43:33 -0800
Subject: [PATCH] libavutil/timecode.c: Extend check_fps logic to handle high
 frame rates

QuickTime sources continue to push higher and higher frame rates. This
change moves away from explictly testing incoming fps values towards
ensuring the incoming value is evenly divisable by 24, 25, or 30.
---
 libavutil/timecode.c | 6 +++---
 1 file changed, 3 insertions(+), 3 deletions(-)

diff --git a/libavutil/timecode.c b/libavutil/timecode.c
index 1dfd040..c10895c 100644
--- a/libavutil/timecode.c
+++ b/libavutil/timecode.c
@@ -141,10 +141,10 @@ char *av_timecode_make_mpeg_tc_string(char *buf, uint32_t 
tc25bit)
 static int check_fps(int fps)
 {
 int i;
-static const int supported_fps[] = {24, 25, 30, 48, 50, 60};
+static const int supported_fps_bases[] = {24, 25, 30};
 
-for (i = 0; i  FF_ARRAY_ELEMS(supported_fps); i++)
-if (fps == supported_fps[i])
+for (i = 0; i  FF_ARRAY_ELEMS(supported_fps_bases); i++)
+if (fps % supported_fps_bases[i] == 0)
 return 0;
 return -1;
 }
-- 
1.8.5.2 (Apple Git-48)

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Re: [FFmpeg-devel] Patch for device list error in decklink_common.cpp

[Jon bae]

Thanks Ramiro for the correction!
Here is the new patch. (Is it better to post directly the patch, or is ok
as a attachment?)

Regards
Jonathan

2014-12-02 22:19 GMT+01:00 Ramiro Polla ramiro.po...@gmail.com:


 On 02.12.2014 20:30, Jon bae wrote:

 Here is the other patch for decklink_common.cpp. It fix the error:

 COM initialization failed
 [decklink @ 02e5b520] Could not create DeckLink iterator
 dummy: Immediate exit request


  From 203eba2fad14dd6d84552d6c22899792e80b53bb Mon Sep 17 00:00:00 2001
 From: Jonathan Baecker jonba...@gmail.com
 Date: Tue, 2 Dec 2014 20:12:38 +0100
 Subject: [PATCH 2/2] device list error in decklink_common

 Signed-off-by: Jonathan Baecker jonba...@gmail.com
 ---
  libavdevice/decklink_common.cpp | 24 ++--
  1 file changed, 14 insertions(+), 10 deletions(-)

 diff --git a/libavdevice/decklink_common.cpp
 b/libavdevice/decklink_common.cpp
 index 8eff910..8f7e32a 100644
 --- a/libavdevice/decklink_common.cpp
 +++ b/libavdevice/decklink_common.cpp
 @@ -42,16 +42,20 @@ IDeckLinkIterator *CreateDeckLinkIteratorInstance
 (void)
  {
  IDeckLinkIterator *iter;

 -if (CoInitialize(NULL) != S_OK) {
 -av_log(NULL, AV_LOG_ERROR, COM initialization failed.\n);
 -return NULL;
 -}
 -
 -if (CoCreateInstance(CLSID_CDeckLinkIterator, NULL, CLSCTX_ALL,
 - IID_IDeckLinkIterator, (void**) iter) != S_OK)
 {
 -av_log(NULL, AV_LOG_ERROR, DeckLink drivers not installed.\n);
 -return NULL;
 -}
 +HRESULT result;
 +/* Initialize COM on this thread */
 +result = CoInitialize(NULL);
 +if (FAILED(result)) {
 +av_log(NULL, AV_LOG_ERROR, COM initialization failed.\n);
 +return NULL;
 +}
 +
 +/* Create an IDeckLinkIterator object to enumerate all DeckLink
 cards in the system */
 +result = CoCreateInstance(CLSID_CDeckLinkIterator, NULL,
 CLSCTX_ALL, IID_IDeckLinkIterator, (void**)iter);
 +if (FAILED(result)) {
 +av_log(NULL, AV_LOG_ERROR, DeckLink drivers not installed.\n);
 +return NULL;
 +}

  return iter;
  }
 --
 2.2.0


 This code is Copyright (c) Blackmagic Design. Try just changing the check
 for CoInitialize(NULL) from != S_OK to  0.
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0001-fix-COM-initialization-failed.patch
Description: Binary data
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[FFmpeg-devel] fix for decklink_common.cpp

[Jon bae]

Hello everybody,
on the Zeranoe forum we find out that there is two bugs in the
decklink_common.cpp file. I build I patch and want to send them now to you.

I hope is all right, I never did before a git format-patch...

Regards!
Jonathan


fix-device_list-and-COM-initialization-failed_in decklink_common.patch
Description: Binary data
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Re: [FFmpeg-devel] fix for decklink_common.cpp

[Jon bae]

Hello Carl,
that means that the patch is at the moment not useful?! This code comes
from the decklink sdk and there was also the  FAILED() defined, I think. I
have the code from here:
http://ffmpeg.zeranoe.com/forum/viewtopic.php?f=10t=1823start=10#p7580
and there they say it works.
When is ok for you, I copy your answer to the zeranoe forum, then they can
help me to fix it.

2014-12-02 16:45 GMT+01:00 Carl Eugen Hoyos ceho...@ag.or.at:

 Jon bae jonbae77 at gmail.com writes:

  I hope is all right, I never did before a
  git format-patch...

 You used git format-patch perfectly, thank you.

 Your patch introduces tabs, they cannot be committed
 to FFmpeg. You can find tabs with tools/patcheck,
 a script that is part of the source.

 Please do not replace av_log() with fprintf().
 Please do not move opening braces, just leave them
 where they are now.
 Please keep the indentation at four spaces.

 Assuming that the free fix is unrelated to the
 remaining patch, please make it separate.

 (I don't know where FAILED() is defined and
 therefore cannot comment on the actual patch and
 if it can be simplified.)

 Thank you, Carl Eugen

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[FFmpeg-devel] Patch for heap corruption run time error in decklink_common.cpp

[Jon bae]

Ok here a second run, I try to follow the instruction from Carl Eugen. 
This is the first patch for decklink_common.cpp. It fix this error:


   Unhandled exception at 0x76FA4102 (ntdll.dll) in ffmpeg.exe:
   0xC374: A heap has been corrupted (parameters: 0x7701B4B0).

Regards

Jonathan
From e9bc8e910f515af4030054df3e6feb308f3208aa Mon Sep 17 00:00:00 2001
From: Jonathan Baecker jonba...@gmail.com
Date: Tue, 2 Dec 2014 20:10:41 +0100
Subject: [PATCH 1/2] heap corruption run time error in decklink_common

Signed-off-by: Jonathan Baecker jonba...@gmail.com
---
 libavdevice/decklink_common.cpp | 6 --
 1 file changed, 4 insertions(+), 2 deletions(-)

diff --git a/libavdevice/decklink_common.cpp b/libavdevice/decklink_common.cpp
index 9a9e44b..8eff910 100644
--- a/libavdevice/decklink_common.cpp
+++ b/libavdevice/decklink_common.cpp
@@ -69,9 +69,12 @@ static char *dup_wchar_to_utf8(wchar_t *w)
 }
 #define DECKLINK_STROLECHAR *
 #define DECKLINK_STRDUP dup_wchar_to_utf8
+#define DECKLINK_FREE(s) SysFreeString(s)
 #else
 #define DECKLINK_STRconst char *
 #define DECKLINK_STRDUP av_strdup
+/* free() is needed for a string returned by the DeckLink SDL. */
+#define DECKLINK_FREE(s) free((void *) s)
 #endif
 
 HRESULT ff_decklink_get_display_name(IDeckLink *This, const char **displayName)
@@ -81,8 +84,7 @@ HRESULT ff_decklink_get_display_name(IDeckLink *This, const 
char **displayName)
 if (hr != S_OK)
 return hr;
 *displayName = DECKLINK_STRDUP(tmpDisplayName);
-/* free() is needed for a string returned by the DeckLink SDL. */
-free((void *) tmpDisplayName);
+DECKLINK_FREE(tmpDisplayName);
 return hr;
 }
 
-- 
2.2.0

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[FFmpeg-devel] fix for colorspace minimum option value

[jon]

From 616d5017d8d2e566db3deb2696cc1672f2019777 Mon Sep 17 00:00:00 2001
From: Jon Morley j...@tweaksoftware.com
Date: Mon, 10 Nov 2014 10:43:42 -0800
Subject: [PATCH] options_table.h: min value for colorspace is 0
 (AVCOL_SPC_RGB)

The min value for colorspace should be zero and not one since the first
valid index into the frame colorspace array is AVCOL_SPC_RGB which is 0.
---
 libavcodec/options_table.h | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

diff --git a/libavcodec/options_table.h b/libavcodec/options_table.h
index d617ae7..b72fbb1 100644
--- a/libavcodec/options_table.h
+++ b/libavcodec/options_table.h
@@ -432,7 +432,7 @@ static const AVOption avcodec_options[] = {
 {iec61966_2_1, IEC 61966-2-1,0, AV_OPT_TYPE_CONST, {.i64 = 
AVCOL_TRC_IEC61966_2_1 }, INT_MIN, INT_MAX, V|E|D, color_trc_type},
 {bt2020_10bit, BT.2020 - 10 bit, 0, AV_OPT_TYPE_CONST, {.i64 = 
AVCOL_TRC_BT2020_10 },INT_MIN, INT_MAX, V|E|D, color_trc_type},
 {bt2020_12bit, BT.2020 - 12 bit, 0, AV_OPT_TYPE_CONST, {.i64 = 
AVCOL_TRC_BT2020_12 },INT_MIN, INT_MAX, V|E|D, color_trc_type},
-{colorspace, color space, OFFSET(colorspace), AV_OPT_TYPE_INT, 
{.i64 = AVCOL_SPC_UNSPECIFIED }, 1, AVCOL_SPC_NB-1, V|E|D, 
colorspace_type},
+{colorspace, color space, OFFSET(colorspace), AV_OPT_TYPE_INT, 
{.i64 = AVCOL_SPC_UNSPECIFIED }, 0, AVCOL_SPC_NB-1, V|E|D, 
colorspace_type},
 {rgb, RGB, 0, AV_OPT_TYPE_CONST, {.i64 = 
AVCOL_SPC_RGB }, INT_MIN, INT_MAX, V|E|D, colorspace_type},
 {bt709,   BT.709,  0, AV_OPT_TYPE_CONST, {.i64 = 
AVCOL_SPC_BT709 },   INT_MIN, INT_MAX, V|E|D, colorspace_type},
 {unspecified, Unspecified, 0, AV_OPT_TYPE_CONST, {.i64 = 
AVCOL_SPC_UNSPECIFIED }, INT_MIN, INT_MAX, V|E|D, colorspace_type},

--
1.8.5.2 (Apple Git-48)

From 616d5017d8d2e566db3deb2696cc1672f2019777 Mon Sep 17 00:00:00 2001
From: Jon Morley j...@tweaksoftware.com
Date: Mon, 10 Nov 2014 10:43:42 -0800
Subject: [PATCH] options_table.h: min value for colorspace is 0
 (AVCOL_SPC_RGB)

The min value for colorspace should be zero and not one since the first
valid index into the frame colorspace array is AVCOL_SPC_RGB which is 0.
---
 libavcodec/options_table.h | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

diff --git a/libavcodec/options_table.h b/libavcodec/options_table.h
index d617ae7..b72fbb1 100644
--- a/libavcodec/options_table.h
+++ b/libavcodec/options_table.h
@@ -432,7 +432,7 @@ static const AVOption avcodec_options[] = {
 {iec61966_2_1, IEC 61966-2-1,0, AV_OPT_TYPE_CONST, {.i64 = 
AVCOL_TRC_IEC61966_2_1 }, INT_MIN, INT_MAX, V|E|D, color_trc_type},
 {bt2020_10bit, BT.2020 - 10 bit, 0, AV_OPT_TYPE_CONST, {.i64 = 
AVCOL_TRC_BT2020_10 },INT_MIN, INT_MAX, V|E|D, color_trc_type},
 {bt2020_12bit, BT.2020 - 12 bit, 0, AV_OPT_TYPE_CONST, {.i64 = 
AVCOL_TRC_BT2020_12 },INT_MIN, INT_MAX, V|E|D, color_trc_type},
-{colorspace, color space, OFFSET(colorspace), AV_OPT_TYPE_INT, {.i64 = 
AVCOL_SPC_UNSPECIFIED }, 1, AVCOL_SPC_NB-1, V|E|D, colorspace_type},
+{colorspace, color space, OFFSET(colorspace), AV_OPT_TYPE_INT, {.i64 = 
AVCOL_SPC_UNSPECIFIED }, 0, AVCOL_SPC_NB-1, V|E|D, colorspace_type},
 {rgb, RGB, 0, AV_OPT_TYPE_CONST, {.i64 = AVCOL_SPC_RGB },  
   INT_MIN, INT_MAX, V|E|D, colorspace_type},
 {bt709,   BT.709,  0, AV_OPT_TYPE_CONST, {.i64 = AVCOL_SPC_BT709 
},   INT_MIN, INT_MAX, V|E|D, colorspace_type},
 {unspecified, Unspecified, 0, AV_OPT_TYPE_CONST, {.i64 = 
AVCOL_SPC_UNSPECIFIED }, INT_MIN, INT_MAX, V|E|D, colorspace_type},
-- 
1.8.5.2 (Apple Git-48)

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[FFmpeg-devel] Suggested Patch for timecode timebase calculation in mov.c

[jon]

Hi ffmpeg developers.

I am still new to attempting contributing here, so please let me know if 
there is a better approach for this. I am attaching a patch I would like 
to incorporate for calculating the stream time_base of the timecode data 
track in mov's. Please advise.


Thanks!
From 5ae0b5a9cf9c37e11d5a3fea05c80c66b7c00c3e Mon Sep 17 00:00:00 2001
From: Jon Morley j...@tweaksoftware.com
Date: Tue, 9 Sep 2014 11:48:02 -0700
Subject: [PATCH] libavformat/mov.c: Set stream time_base from
 frameduration/timescale

Use tmcd atom's timescale and frameduration to for stream time base
instead of 1/framenum. For timecode streams that have an entry for each
frame 1/framenum is less accurate than frameduration/timescale.
---
 libavformat/mov.c | 6 --
 1 file changed, 4 insertions(+), 2 deletions(-)

diff --git a/libavformat/mov.c b/libavformat/mov.c
index ae48c02..c300dd2 100644
--- a/libavformat/mov.c
+++ b/libavformat/mov.c
@@ -1530,8 +1530,10 @@ static int mov_parse_stsd_data(MOVContext *c, 
AVIOContext *pb,
 tmcd_ctx-tmcd_flags = val;
 if (val  1)
 st-codec-flags2 |= CODEC_FLAG2_DROP_FRAME_TIMECODE;
-st-codec-time_base.den = st-codec-extradata[16]; /* number of 
frame */
-st-codec-time_base.num = 1;
+int timescale = AV_RB32(st-codec-extradata + 8);
+int framedur  =  AV_RB32(st-codec-extradata + 12);
+st-codec-time_base.den = timescale;
+st-codec-time_base.num = framedur;
 if (size  30) {
 uint32_t len = AV_RB32(st-codec-extradata + 18); /* name 
atom length */
 uint32_t format = AV_RB32(st-codec-extradata + 22);
-- 
1.8.5.2 (Apple Git-48)

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[FFmpeg-devel] Set AV_TIMECODE_FLAG_ALLOWNEGATIVE in avutil/timecode.c av_timecode_init_from_string

[jon]

If the timecode value is negative it certainly seems safe to set the 
AV_TIMECODE_FLAG_ALLOWNEGATIVE on the AVTimecode object.
From 35ae38f9557e1ff01040d0e681e843d1727f979b Mon Sep 17 00:00:00 2001
From: Jon Morley j...@tweaksoftware.com
Date: Tue, 9 Sep 2014 12:12:17 -0700
Subject: [PATCH 2/2] libavutil/timecode.c: improve flag setting when init tc
 from string

If the string begins with a negative HH (hour) value then make sure to
set the AV_TIMECODE_FLAG_ALLOW_NEGATIVE flag for the resulting
tc-flags.
---
 libavutil/timecode.c | 7 ---
 1 file changed, 4 insertions(+), 3 deletions(-)

diff --git a/libavutil/timecode.c b/libavutil/timecode.c
index 1dfd040..b96a4d1 100644
--- a/libavutil/timecode.c
+++ b/libavutil/timecode.c
@@ -201,9 +201,10 @@ int av_timecode_init_from_string(AVTimecode *tc, 
AVRational rate, const char *st
 }
 
 memset(tc, 0, sizeof(*tc));
-tc-flags = c != ':' ? AV_TIMECODE_FLAG_DROPFRAME : 0; // drop if ';', 
'.', ...
-tc-rate  = rate;
-tc-fps   = fps_from_frame_rate(rate);
+tc-flags  = c != ':' ? AV_TIMECODE_FLAG_DROPFRAME : 0; // drop if ';', 
'.', ...
+tc-flags |= (hh  0) ? AV_TIMECODE_FLAG_ALLOWNEGATIVE : 0; // neg if 
starts with '-'
+tc-rate   = rate;
+tc-fps= fps_from_frame_rate(rate);
 
 ret = check_timecode(log_ctx, tc);
 if (ret  0)
-- 
1.8.5.2 (Apple Git-48)

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