[FFmpeg-devel] Encode2() function in libavcodec/avcodec.h

2014-10-30 Thread greeshma 
Hi,

the attributes has been changed to (AVCodecContext *avctx, AVPacket *avpkt,
const AVFrame *frame, int *got_packet_ptr);

what is the exact change made in the function.
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Re: [FFmpeg-devel] OPW Qualification Task: Enable daemon mode for FFserver

2014-10-30 Thread Binathi Bingi 
I see your point.
Changing the ffserver_config.c would help in addressing config files having
NoDaemon option.
You were right! Thanks.


From 476c8605fab4d6c575c38796dd9dccaf854cf536 Mon Sep 17 00:00:00 2001
From: Binathi Bingi binti...@gmail.com
Date: Thu, 30 Oct 2014 13:43:13 +0530
Subject: [PATCH] Restoring back NoDaemon option in ffserver.conf

---
 ffserver_config.c | 2 ++
 1 file changed, 2 insertions(+)

diff --git a/ffserver_config.c b/ffserver_config.c
index e2c78d8..ab33ec5 100644
--- a/ffserver_config.c
+++ b/ffserver_config.c
@@ -358,6 +358,8 @@ static int ffserver_parse_config_global(FFServerConfig
*config, const char *cmd,
 ffserver_get_arg(arg, sizeof(arg), p);
 if (resolve_host(config-http_addr.sin_addr, arg) != 0)
 ERROR(%s:%d: Invalid host/IP address: %s\n, arg);
+} else if (!av_strcasecmp(cmd, NoDaemon)) {
+WARNING(NoDaemon option has no effect, you should remove
it\n);
 } else if (!av_strcasecmp(cmd, RTSPPort)) {
 ffserver_get_arg(arg, sizeof(arg), p);
 val = atoi(arg);
-- 
1.9.1


Binathi

On Thu, Oct 30, 2014 at 8:04 AM, Michael Niedermayer michae...@gmx.at
wrote:

 On Thu, Oct 30, 2014 at 07:49:10AM +0530, Binathi Bingi wrote:
  If you check the latest ffserver.conf file on GIT [
  http://git.videolan.org/?p=ffmpeg.git
  
 http://www.google.com/url?q=http%3A%2F%2Fgit.videolan.org%2F%3Fp%3Dffmpeg.gitsa=Dsntz=1usg=AFQjCNEA5QH18TtMxcLhGx4b04pMUwSgYA
 ],
  there is NoDaemon option in it. So the patch is written as per latest
  version. But if desired can include NoDameon option in ffserver.conf and
  make futher changes.

 ffserver must work with any config file it worked with previously
 not only the ffserver.conf file from git

 [...]
 --
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 Those who are best at talking, realize last or never when they are wrong.

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[FFmpeg-devel] [PATCH] libavformat/assenc: Add ignore_gaps option

2014-10-30 Thread Rodger Combs 
This patch lets the user ignore ReadOrder when writing ASS subtitles, which is 
useful when e.g. streaming output.

From d3d4cb4da2382a1d762fa1e9bfafbaf3d18cf5c5 Mon Sep 17 00:00:00 2001
From: Rodger Combs rodger.co...@gmail.com
Date: Thu, 30 Oct 2014 04:33:17 -0500
Subject: [PATCH] libavformat/assenc: Add ignore_gaps option

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

diff --git a/libavformat/assenc.c b/libavformat/assenc.c
index 8225967..c7a1d1e 100644
--- a/libavformat/assenc.c
+++ b/libavformat/assenc.c
@@ -23,6 +23,8 @@
 #include avformat.h
 #include internal.h
 
+#include libavutil/opt.h
+
 typedef struct DialogueLine {
 int readorder;
 char *line;
@@ -30,12 +32,14 @@ typedef struct DialogueLine {
 } DialogueLine;
 
 typedef struct ASSContext{
+const AVClass *class;
 int write_ts; // 0: ssa (timing in payload), 1: ass (matroska like)
 int expected_readorder;
 DialogueLine *dialogue_cache;
 DialogueLine *last_added_dialogue;
 int cache_size;
 int ssa_mode;
+int ignore_gaps;
 }ASSContext;
 
 static int write_header(AVFormatContext *s)
@@ -178,7 +182,7 @@ static int write_packet(AVFormatContext *s, AVPacket *pkt)
 return AVERROR(ENOMEM);
 }
 insert_dialogue(ass, dialogue);
-purge_dialogues(s, 0);
+purge_dialogues(s, ass-ignore_gaps);
 } else {
 avio_write(s-pb, pkt-data, pkt-size);
 }
@@ -192,6 +196,20 @@ static int write_trailer(AVFormatContext *s)
 return 0;
 }
 
+#define OFFSET(x) offsetof(ASSContext, x)
+#define E AV_OPT_FLAG_ENCODING_PARAM
+static const AVOption options[] = {
+{ ignore_gaps, write events immediately, even if they're out-of-order, 
OFFSET(ignore_gaps), AV_OPT_TYPE_INT, {.i64 = 0}, 0, 1, E },
+{ NULL },
+};
+
+static const AVClass ass_class = {
+.class_name = ass muxer,
+.item_name  = av_default_item_name,
+.option = options,
+.version= LIBAVUTIL_VERSION_INT,
+};
+
 AVOutputFormat ff_ass_muxer = {
 .name   = ass,
 .long_name  = NULL_IF_CONFIG_SMALL(SSA (SubStation Alpha) subtitle),
@@ -203,4 +221,5 @@ AVOutputFormat ff_ass_muxer = {
 .write_packet   = write_packet,
 .write_trailer  = write_trailer,
 .flags  = AVFMT_GLOBALHEADER | AVFMT_NOTIMESTAMPS | 
AVFMT_TS_NONSTRICT,
+.priv_class = ass_class,
 };
-- 
2.1.0.rc1.4.g764c739

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Re: [FFmpeg-devel] Evolution of lavfi's design and API

2014-10-30 Thread Stefano Sabatini 
Sorry for the slow reply.

On date Wednesday 2014-10-22 23:45:42 +0200, Nicolas George encoded: 
 [ CCing Anton, as most that is written here also apply to libav too, and
 this would be a good occasion to try a cross-fork cooperation; if that is
 not wanted, please let us know so we can drop the cc. ]
 
 1. Problems with the current design
 
   1.1. Mixed input-/output-driven model
 
 Currently, lavfi is designed to work in a mixed input-driven and
 output-driven model. That means the application needs sometimes to add
 input to buffersources and sometimes request output to buffersinks. This
 is a bit of a nuisance, because it requires the application to do it
 properly: adding input on the wrong input or requesting a frame on the
 wrong output will cause extra memory consumption or latency.
 
 With the libav API, it can not work at all since there is no mechanism
 to determine which input needs a frame in order to proceed.
 
 The libav API is clearly designed for a more output-driven
 implementation, with FIFOs anywhere to prevent input-driven frames to
 reach unready filters. Unfortunately, since it is impossible from the
 outside to guess what output will get a frame next, that can cause
 frames to accumulate anywhere in the filter graph, eating a lot of
 memory unnecessarily.
 
 FFmpeg's API has eliminated FIFOs in favour of queues in filters that
 need it, but these queues can not be controlled for unusual filter
 graphs with extreme needs. Also, there still is an implicit FIFO inside
 buffersink.
 
   1.2. Recursive implementation
 
 All work in a filter graph is triggered by recursive invocations of the
 filters' methods. It makes debugging harder. It also can lead to large
 stack usage and makes frame- and filter-level multithreading harder to
 implement. It also prevents some diagnosis from working reliably.
 
   1.3. EOF handling
 
 Currently, EOF is propagated only through the return value of the
 request_frame() method. That means it only works in an output-driven
 scheme. It also means that it has no timestamp attached to it; this is
 an issue for filters where the duration of the last frame is relevant,
 like vf_fps.
 
   1.4. Latency
 
 Some filters need to know the timestamp of the next frame in order to
 know when the current frame will stop and be able to process it:
 overlay, fps are two examples. These filters will introduce a latency of
 one input frame that could otherwise be avoided.
 
   1.5. Timestamps
 
 Some filters do not care about timestamps at all. Some check and have a
 proper handling of NOPTS values. Some filters just assume the frames
 will have timestamps, and possibly make extra assumptions on that:
 monotony, consistency, etc. That is an inconsistent mess.
 
   1.6. Sparse streams
 
 There is a more severe instance of the latency issue when the input
 comes from an interleaved sparse stream: in that case, waiting for the
 next frame in order to find the end of the current one may require
 demuxing a large chunk of input, in turn provoking a lot of activity on
 other inputs of the graph.

Other issues.

S1. the filtergraph can't properly readapt to mid-stream
changes involving assumed invariants (aspect ratio, size, timebase,
pixel format, sample_rate). Indeed the framework was designed as
though some of these properties (the ones defined by query_formats)
were not allowed to change.

S2. Another problem is that we initialize the filter before the
filtergraph, so for example the single filter can't readapt to the
filtergraph topology. For example it would be useful to have the split
filter to change the number of outputs depending on the number of
outputs specified, but this can't be easily achieved. (That's in my
opinion a minor problem though).

S3. It is not possible to direct commands towards a specific
filter. For this we can add an ID to each filter instance. We could
have something has:
color:left_color=c=red   [left]
color:right_color=c=blue [right]

then you can send commands (e.g. with zmq) with:
echo left_color c yellow | tools/zmqsend

S4. We should support output encoding movie. We got stuck designing
the interface for that.

...

About fifos and queues, we could add some options to control fifo
filters to limit their size.

For example we could specify the maximum number of allowed queued
frames, or the total allowed size, and the dropping policy (drop last,
drop first, drop random frame in the midst).

 2. Proposed API changes
 
   To fix/enhance all these issues, I believe a complete rethink of the
   scheduling design of the library is necessary. I propose the following
   changes.
 
   Note: some of these changes are not 100% related to the issues I raised,
   but looked like a good idea while thinking on an API rework.
 
   2.1. AVFrame.duration
 
 Add a duration field to AVFrame; if set,

Re: [FFmpeg-devel] [PATCH] avformat/subtitles: reduce log level of UTF-16 warning

2014-10-30 Thread Nicolas George 
L'octidi 8 brumaire, an CCXXIII, wm4 a écrit :
 Would a patch adding a this is UTF-8 flag to the subtitle codec
 headers be accepted?

No.

   What should it be named?

sub_charenc

Regards,

-- 
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Re: [FFmpeg-devel] [PATCH] ffprobe.xsd: Add packets_and_frames element definition

2014-10-30 Thread Stefano Sabatini 
On date Thursday 2014-10-23 15:17:55 +0200, Tobias Rapp encoded:
 Attached patch allows validation of ffprobe XML output if
 -show_packets and -show_frames are both set.

 From 0c10cf28993f61cbb8365caf8d06f5d5b6111a80 Mon Sep 17 00:00:00 2001
 From: Tobias Rapp t.r...@noa-audio.com
 Date: Thu, 23 Oct 2014 15:10:28 +0200
 Subject: [PATCH] ffprobe.xsd: Add packets_and_frames element definition
 
 This allows validation of ffprobe XML output if -show_packets and
 -show_frames are both set.
 ---
  doc/ffprobe.xsd | 11 +++
  1 file changed, 11 insertions(+)

Applied, thanks.
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Re: [FFmpeg-devel] [PATCH] libavformat/assenc: Add ignore_gaps option

2014-10-30 Thread wm4 
On Thu, 30 Oct 2014 04:36:28 -0500
Rodger Combs rodger.co...@gmail.com wrote:

 This patch lets the user ignore ReadOrder when writing ASS subtitles, which 
 is useful when e.g. streaming output.
 
 From d3d4cb4da2382a1d762fa1e9bfafbaf3d18cf5c5 Mon Sep 17 00:00:00 2001
 From: Rodger Combs rodger.co...@gmail.com
 Date: Thu, 30 Oct 2014 04:33:17 -0500
 Subject: [PATCH] libavformat/assenc: Add ignore_gaps option
 
 ---
  libavformat/assenc.c | 21 -
  1 file changed, 20 insertions(+), 1 deletion(-)
 
 diff --git a/libavformat/assenc.c b/libavformat/assenc.c
 index 8225967..c7a1d1e 100644
 --- a/libavformat/assenc.c
 +++ b/libavformat/assenc.c
 @@ -23,6 +23,8 @@
  #include avformat.h
  #include internal.h
  
 +#include libavutil/opt.h
 +
  typedef struct DialogueLine {
  int readorder;
  char *line;
 @@ -30,12 +32,14 @@ typedef struct DialogueLine {
  } DialogueLine;
  
  typedef struct ASSContext{
 +const AVClass *class;
  int write_ts; // 0: ssa (timing in payload), 1: ass (matroska like)
  int expected_readorder;
  DialogueLine *dialogue_cache;
  DialogueLine *last_added_dialogue;
  int cache_size;
  int ssa_mode;
 +int ignore_gaps;
  }ASSContext;
  
  static int write_header(AVFormatContext *s)
 @@ -178,7 +182,7 @@ static int write_packet(AVFormatContext *s, AVPacket *pkt)
  return AVERROR(ENOMEM);
  }
  insert_dialogue(ass, dialogue);
 -purge_dialogues(s, 0);
 +purge_dialogues(s, ass-ignore_gaps);
  } else {
  avio_write(s-pb, pkt-data, pkt-size);
  }
 @@ -192,6 +196,20 @@ static int write_trailer(AVFormatContext *s)
  return 0;
  }
  
 +#define OFFSET(x) offsetof(ASSContext, x)
 +#define E AV_OPT_FLAG_ENCODING_PARAM
 +static const AVOption options[] = {
 +{ ignore_gaps, write events immediately, even if they're 
 out-of-order, OFFSET(ignore_gaps), AV_OPT_TYPE_INT, {.i64 = 0}, 0, 1, E },
 +{ NULL },
 +};
 +
 +static const AVClass ass_class = {
 +.class_name = ass muxer,
 +.item_name  = av_default_item_name,
 +.option = options,
 +.version= LIBAVUTIL_VERSION_INT,
 +};
 +
  AVOutputFormat ff_ass_muxer = {
  .name   = ass,
  .long_name  = NULL_IF_CONFIG_SMALL(SSA (SubStation Alpha) 
 subtitle),
 @@ -203,4 +221,5 @@ AVOutputFormat ff_ass_muxer = {
  .write_packet   = write_packet,
  .write_trailer  = write_trailer,
  .flags  = AVFMT_GLOBALHEADER | AVFMT_NOTIMESTAMPS | 
 AVFMT_TS_NONSTRICT,
 +.priv_class = ass_class,
  };

The option name is a bit unspecific - how about ignore_readorder?
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Re: [FFmpeg-devel] [PATCH] lavfi: add xbr filter

2014-10-30 Thread arwa arif 
On Wed, Oct 29, 2014 at 3:36 AM, Clément Bœsch u...@pkh.me wrote:

 On Tue, Oct 28, 2014 at 10:51:27PM +0100, Michael Niedermayer wrote:
  On Tue, Oct 28, 2014 at 07:16:45PM +0100, Clément Bœsch wrote:
   On Tue, Oct 28, 2014 at 06:30:34PM +0100, Stefano Sabatini wrote:
   [...]
How much effort would it take to implement the remaining scaling
 modes?
   
  
   According to
   https://ffmpeg.org/pipermail/ffmpeg-devel/2014-October/164574.html
  
   I think 4x can be done fast enough, but 3x will take time.
  
   [...]
 +typedef struct {
 +uint32_t rgbtoyuv[124];
   
We should avoid this 64MiB. Also the table should be possibly static,
so you don't have to fill it per each xBR instance.
   
  
   So, I requested to do it exactly the same as HQx because this part is
   common according to the specifications. This should be kept the same
   vf_hqx, and then factorized.
  
 
   Now about removing this allocation, I did benchmark this LUT vs
   computation (see attached patch for comp. version). And the problem is
   that it's slightly slower, probably due to the /1000.
 
  why do you divide at all ?
  cant you do the computations with full precission ?

 I wasn't able to... but I was probably doing it wrong.

 And anyway, so far I observed this:
   lut: 127fps
   nolut+div:   119fps
   nolut+nodiv: 123fps

 So even with fast computation, it's still slower than the LUT. It
 probably
 doesn't matter that much in practice, and dropping that huge table might be
 worth the performance impact (feel free to discuss).

 Note that looking at the original code (which was working on rgb565 only),
 it was bitexact. The rgb 24-bit was added in the modern hqx with float
 point. So we can probably tolerate the inaccuracy. Still, if you find a
 way of keeping full accuracy with the modern implementation...

 Typically, I tried stuff like this:

   const uint32_t y = (uint32_t)((1225*r + 2404*g +  467*b + (111)) 
 12);
   const uint32_t u = (uint32_t)((-692*r - 1356*g + 2048*b + (111)) 
 12) + 128;
   const uint32_t v = (uint32_t)((2048*r - 1716*g -  332*b + (111)) 
 12) + 128;

 ...but I'm probably doing it very wrong somewhere (sign issue maybe?),
 haven't
 looked deeper. I went up to 15 bits, still didn't match, so I was probably
 doing something stupid.

  also instead of doing 2 rgb2yuv and then taking their difference you
  can do the difference in rgb space and convert the rgb difference to
  a yuv difference
  its just aM - bM = (a-b)M

 Ah, sounds like a good idea, I guess I'll try that.

 [...]

 --
 Clément B.

 Updated the patch. How should I finally go about converting rgb to yuv?


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From 905fb0f15033e447fc344097dd649ca671fa1074 Mon Sep 17 00:00:00 2001
From: Arwa Arif arwaarif1...@gmail.com
Date: Thu, 30 Oct 2014 18:05:45 +0530
Subject: [PATCH] [PATCH]lvafi: add xbr filter

---
 doc/filters.texi |   81 ++--
 libavfilter/Makefile |1 +
 libavfilter/allfilters.c |1 +
 libavfilter/vf_xbr.c |  319 ++
 4 files changed, 335 insertions(+), 67 deletions(-)
 create mode 100644 libavfilter/vf_xbr.c

diff --git a/doc/filters.texi b/doc/filters.texi
index 7be29de..253384b 100644
--- a/doc/filters.texi
+++ b/doc/filters.texi
+@section xbr
+Apply high-quality magnification filter which is designed for pixel art. It follows a set
+of edge-detection rules @url{http://www.libretro.com/forums/viewtopic.php?f=6t=134}.
+This filter was originally created by Hyllian.
+
 @anchor{yadif}
 @section yadif
 
diff --git a/libavfilter/Makefile b/libavfilter/Makefile
index 6d868e7..2c56e38 100644
--- a/libavfilter/Makefile
+++ b/libavfilter/Makefile
@@ -198,6 +198,7 @@ OBJS-$(CONFIG_VIDSTABDETECT_FILTER)  += vidstabutils.o vf_vidstabdetect.
 OBJS-$(CONFIG_VIDSTABTRANSFORM_FILTER)   += vidstabutils.o vf_vidstabtransform.o
 OBJS-$(CONFIG_VIGNETTE_FILTER)   += vf_vignette.o
 OBJS-$(CONFIG_W3FDIF_FILTER) += vf_w3fdif.o
+OBJS-$(CONFIG_XBR_FILTER)+= vf_xbr.o
 OBJS-$(CONFIG_YADIF_FILTER)  += vf_yadif.o
 OBJS-$(CONFIG_ZMQ_FILTER)+= f_zmq.o
 OBJS-$(CONFIG_ZOOMPAN_FILTER)+= vf_zoompan.o
diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c
index d88a9ad..2352d44 100644
--- a/libavfilter/allfilters.c
+++ b/libavfilter/allfilters.c
@@ -213,6 +213,7 @@ void avfilter_register_all(void)
 REGISTER_FILTER(VIDSTABTRANSFORM, vidstabtransform, vf);
 REGISTER_FILTER(VIGNETTE,   vignette,   vf);
 REGISTER_FILTER(W3FDIF, w3fdif, vf);
+REGISTER_FILTER(XBR,xbr,vf);
 REGISTER_FILTER(YADIF,  yadif,  vf);
 REGISTER_FILTER(ZMQ,zmq,vf);
 REGISTER_FILTER(ZOOMPAN,zoompan,

Re: [FFmpeg-devel] Evolution of lavfi's design and API

2014-10-30 Thread Michael Niedermayer 
On Thu, Oct 30, 2014 at 11:50:46AM +0100, Stefano Sabatini wrote:
 Sorry for the slow reply.
 
 On date Wednesday 2014-10-22 23:45:42 +0200, Nicolas George encoded: 
  [ CCing Anton, as most that is written here also apply to libav too, and
  this would be a good occasion to try a cross-fork cooperation; if that is
  not wanted, please let us know so we can drop the cc. ]
  
  1. Problems with the current design
  
1.1. Mixed input-/output-driven model
  
  Currently, lavfi is designed to work in a mixed input-driven and
  output-driven model. That means the application needs sometimes to add
  input to buffersources and sometimes request output to buffersinks. This
  is a bit of a nuisance, because it requires the application to do it
  properly: adding input on the wrong input or requesting a frame on the
  wrong output will cause extra memory consumption or latency.
  
  With the libav API, it can not work at all since there is no mechanism
  to determine which input needs a frame in order to proceed.
  
  The libav API is clearly designed for a more output-driven
  implementation, with FIFOs anywhere to prevent input-driven frames to
  reach unready filters. Unfortunately, since it is impossible from the
  outside to guess what output will get a frame next, that can cause
  frames to accumulate anywhere in the filter graph, eating a lot of
  memory unnecessarily.
  
  FFmpeg's API has eliminated FIFOs in favour of queues in filters that
  need it, but these queues can not be controlled for unusual filter
  graphs with extreme needs. Also, there still is an implicit FIFO inside
  buffersink.
  
1.2. Recursive implementation
  
  All work in a filter graph is triggered by recursive invocations of the
  filters' methods. It makes debugging harder. It also can lead to large
  stack usage and makes frame- and filter-level multithreading harder to
  implement. It also prevents some diagnosis from working reliably.
  
1.3. EOF handling
  
  Currently, EOF is propagated only through the return value of the
  request_frame() method. That means it only works in an output-driven
  scheme. It also means that it has no timestamp attached to it; this is
  an issue for filters where the duration of the last frame is relevant,
  like vf_fps.
  
1.4. Latency
  
  Some filters need to know the timestamp of the next frame in order to
  know when the current frame will stop and be able to process it:
  overlay, fps are two examples. These filters will introduce a latency of
  one input frame that could otherwise be avoided.
  
1.5. Timestamps
  
  Some filters do not care about timestamps at all. Some check and have a
  proper handling of NOPTS values. Some filters just assume the frames
  will have timestamps, and possibly make extra assumptions on that:
  monotony, consistency, etc. That is an inconsistent mess.
  
1.6. Sparse streams
  
  There is a more severe instance of the latency issue when the input
  comes from an interleaved sparse stream: in that case, waiting for the
  next frame in order to find the end of the current one may require
  demuxing a large chunk of input, in turn provoking a lot of activity on
  other inputs of the graph.
 
 Other issues.
 

 S1. the filtergraph can't properly readapt to mid-stream
 changes involving assumed invariants (aspect ratio, size, timebase,
 pixel format, sample_rate). Indeed the framework was designed as
 though some of these properties (the ones defined by query_formats)
 were not allowed to change.

no, no and no :)
the filtergraph should be able adapt to some changes like aspect,
resolution and pixel / sample format. These are not invariants, some
of this definitly worked when i tried it long ago
i posted a (incomplete) patchset that fixes bugs in this relation
long ago There are filters that can perfectly well handle changes in
resolution, aspect, formats, ...
and there are filters which are buggy but could when they are fixed
also equally well support such changes
and there are filters which fundamentally do not support some changes,
these need to either be reinited and loose state/history or a
scale/aresample be inserted before them when changes on their input
occur
for some filters reinit is not appropriate, examples are things that
are intended to collect global statistics
Also scale / aresample filters can serve as
parameter change barriers, filters afterwards do not need to deal
with such changes


 
 S2. Another problem is that we initialize the filter before the
 filtergraph, so for example the single filter can't readapt to the
 filtergraph topology. For example it would be useful to have the split
 filter to change the number of outputs depending on the number of
 outputs specified, but this can't be easily achieved. (That's in my
 opinion a minor problem though).
 

 S3.

Re: [FFmpeg-devel] OPW Qualification Task: Enable daemon mode for FFserver

2014-10-30 Thread Nicolas George 
Le nonidi 9 brumaire, an CCXXIII, Binathi Bingi a écrit :
 From 0fb7dcf1f126bd137e2b2025c5cd6cff4af65801 Mon Sep 17 00:00:00 2001
 From: Binathi Bingi binti...@gmail.com
 Date: Thu, 30 Oct 2014 01:14:08 +0530
 Subject: [PATCH] ffserver: enable back deamon mode
 
 ---
  ffserver.c| 34 ++
  ffserver_config.c |  2 --
  2 files changed, 30 insertions(+), 6 deletions(-)

Thanks for the submission.

I believe you forgot to update the docs (doc/ffserver.texi).

 
 diff --git a/ffserver.c b/ffserver.c
 index ea2a2ae..96e312e 100644
 --- a/ffserver.c
 +++ b/ffserver.c
 @@ -236,7 +236,7 @@ static int rtp_new_av_stream(HTTPContext *c,
   HTTPContext *rtsp_c);
 
  static const char *my_program_name;

 -

I think the empty line could stay.

 +static int ffserver_daemon;
  static int no_launch;
  static int need_to_start_children;
 
 @@ -3671,6 +3671,7 @@ static void handle_child_exit(int sig)
  static void opt_debug(void)
  {
  config.debug = 1;

 +ffserver_daemon = 1;

This is strange: debugging is precisely when you do not want ffserver to
fork into background. If daemon mode is enabled by default, debug mode
should probably disable it, but not the other way around.

  snprintf(config.logfilename, sizeof(config.logfilename), -);
  }
 
 @@ -3694,7 +3695,7 @@ int main(int argc, char **argv)
  {
  struct sigaction sigact = { { 0 } };
  int ret = 0;

 -

The empty line can stay.

 +ffserver_daemon = 1;

This is changing the default behaviour again, I am not sure this is a good
idea nowadays. We can probably leave that line out for the time being.

  config.filename = av_strdup(/etc/ffserver.conf);
 
  parse_loglevel(argc, argv, options);
 @@ -3736,10 +3737,35 @@ int main(int argc, char **argv)
  build_feed_streams();
 
  compute_bandwidth();

 -

Ditto for the empty line.

 +if(ffserver_daemon){

Style nitpick: space between if/for/while/switch and parentheses and before
the brace.

 +int ffserver_id = 0;
 +pid_t sid = 0;
 +

 +ffserver_id = fork(); // Create child process

Small nitpick: the comment seems useless.

 +
 +if (ffserver_id  0){

 +printf(fork failed!\n); //Indication of failure

This kind of message should go to stderr, never stdout. In FFmpeg code, use
av_log() for that.

Also, please include the error reason, using errno (probably through AVERROR
and av_err2str()).

And the comment seems useless too.

 +exit(1);
 +}
 +

 +if(ffserver_id  0){ //Parent process need to kill

I do not understand the comment. And ditto for spacing.

 +exit(0);
 +}
 +
 +sid = setsid(); //set new session

 +if(sid  0){
 +exit(1); //return failure

Ditto for spacing and comment.

 +}
 +

 +open (/dev/null, O_RDWR);
 +
 +if (strcmp(config.logfilename, -) != 0) {
 +close(1);
 +}

This looks strange. If the purpose is to sanitize stdio after detaching,
there are some bits missing.

 +}
  /* signal init */
  signal(SIGPIPE, SIG_IGN);

 -
 +

I have no idea why Git proposes to replace an empty line with an identical
empty line, but that does not matter.

  if (http_server()  0) {
  http_log(Could not start server\n);
  exit(1);
 diff --git a/ffserver_config.c b/ffserver_config.c
 index e44cdf7..e2c78d8 100644
 --- a/ffserver_config.c
 +++ b/ffserver_config.c
 @@ -358,8 +358,6 @@ static int ffserver_parse_config_global(FFServerConfig
 *config, const char *cmd,
  ffserver_get_arg(arg, sizeof(arg), p);
  if (resolve_host(config-http_addr.sin_addr, arg) != 0)
  ERROR(%s:%d: Invalid host/IP address: %s\n, arg);

 -} else if (!av_strcasecmp(cmd, NoDaemon)) {
 -WARNING(NoDaemon option has no effect, you should remove it\n);

Superseded by:

 +} else if (!av_strcasecmp(cmd, NoDaemon)) {
 +WARNING(NoDaemon option has no effect, you should remove
 it\n);

First, you may notice that your mail user agent has mangled the patch by
adding a line break before the end of the last line. That happens when
pasting a patch directly in the body of a mail with a bad mail user agent.
This is of no matter here because of the next point.

Second, this patch needs to be squashed with the previous one, so that they
are a single commit. If you had not yet committed, you could do that using
the --amend option to git commit. Since you have already committed, you must
use git rebase -i master (assuming you created a branch from the branch
called master).

Third, I do not think this exact version is correct. If you make the daemon
mode the default, then NoDaemon must not be ignored, it must have its
specified effect: turn daemon off; if you do not make the daemon mode the
default, then there must be an option to turn it on.

IMHO, the best is to have both Daemon and NoDaemon mode.

  } else if (!av_strcasecmp(cmd, RTSPPort)) {
  ffserver_get_arg(arg,

Re: [FFmpeg-devel] OPW Qualification Task: Enable daemon mode for FFserver

2014-10-30 Thread Reynaldo H. Verdejo Pinochet 
Hello

On 10/30/2014 10:50 AM, Nicolas George wrote:
 [..] Third, I do not think this exact version is correct. If you
 make the daemon mode the default, then NoDaemon must not be
 ignored, it must have its specified effect: turn daemon off; if you
 do not make the daemon mode the default, then there must be an
 option to turn it on.
 
 IMHO, the best is to have both Daemon and NoDaemon mode. [..]

I would prefer no-deamon mode been the default to not
break current behavior till deamon-mode has had a good
deal of production testing. I also think having a single
Daemonize yes/no toggle should be enough but I'm OK
either way in the context of this patch.

Binathi: As an added bonus, try adding an -s to your
commit command line so you get the  Signed-off-by:
line added, this is customary.

Bests,

-- 
Reynaldo H. Verdejo Pinochet
Open Source Group
Samsung Research America / Silicon Valley
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Re: [FFmpeg-devel] [PATCH 4/6] dv: fix weight table for 2x4x8 transform

2014-10-30 Thread Michael Niedermayer 
On Thu, Oct 30, 2014 at 02:49:30PM +0100, Christophe Gisquet wrote:
 Hi,
 
 2014-10-26 19:20 GMT+01:00 Michael Niedermayer michae...@gmx.at:
  I took the liberty to apply the patch and fix the bug instead of
  leaving it open until someone succeeds writing a fate test for it
 
 Hi, I wonder whether this should be reverted for now (and ticket #2970
 reopened at the same time).

The patch fixes decoder artifacts, i dont think we should just revert
it unless it causes artifacts in some other (known to be good) file


 
 I have been checking how to produce interlaced content through the
 encoder, and the current table produces worse results (PSNR-wise, but
 it was big enough to not bother checking visually).
 

 I don't know if that's an encoder issue yet or if the decoder table is
 still incorrect.

did you try to decode with a known to be correct decoder ?

[...]
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Re: [FFmpeg-devel] [PATCH] lavfi: add xbr filter

2014-10-30 Thread Stefano Sabatini 
On date Thursday 2014-10-30 18:23:35 +0530, arwa arif encoded:
 On Wed, Oct 29, 2014 at 3:36 AM, Clément Bœsch u...@pkh.me wrote:
 
  On Tue, Oct 28, 2014 at 10:51:27PM +0100, Michael Niedermayer wrote:
   On Tue, Oct 28, 2014 at 07:16:45PM +0100, Clément Bœsch wrote:
On Tue, Oct 28, 2014 at 06:30:34PM +0100, Stefano Sabatini wrote:
[...]
 How much effort would it take to implement the remaining scaling
  modes?

   
According to
https://ffmpeg.org/pipermail/ffmpeg-devel/2014-October/164574.html
   
I think 4x can be done fast enough, but 3x will take time.
   
[...]
  +typedef struct {
  +uint32_t rgbtoyuv[124];

 We should avoid this 64MiB. Also the table should be possibly static,
 so you don't have to fill it per each xBR instance.

   
So, I requested to do it exactly the same as HQx because this part is
common according to the specifications. This should be kept the same
vf_hqx, and then factorized.
   
  
Now about removing this allocation, I did benchmark this LUT vs
computation (see attached patch for comp. version). And the problem is
that it's slightly slower, probably due to the /1000.
  
   why do you divide at all ?
   cant you do the computations with full precission ?
 
  I wasn't able to... but I was probably doing it wrong.
 
  And anyway, so far I observed this:
lut: 127fps
nolut+div:   119fps
nolut+nodiv: 123fps
 
  So even with fast computation, it's still slower than the LUT. It
  probably
  doesn't matter that much in practice, and dropping that huge table might be
  worth the performance impact (feel free to discuss).
 
  Note that looking at the original code (which was working on rgb565 only),
  it was bitexact. The rgb 24-bit was added in the modern hqx with float
  point. So we can probably tolerate the inaccuracy. Still, if you find a
  way of keeping full accuracy with the modern implementation...
 
  Typically, I tried stuff like this:
 
const uint32_t y = (uint32_t)((1225*r + 2404*g +  467*b + (111)) 
  12);
const uint32_t u = (uint32_t)((-692*r - 1356*g + 2048*b + (111)) 
  12) + 128;
const uint32_t v = (uint32_t)((2048*r - 1716*g -  332*b + (111)) 
  12) + 128;
 
  ...but I'm probably doing it very wrong somewhere (sign issue maybe?),
  haven't
  looked deeper. I went up to 15 bits, still didn't match, so I was probably
  doing something stupid.
 
   also instead of doing 2 rgb2yuv and then taking their difference you
   can do the difference in rgb space and convert the rgb difference to
   a yuv difference
   its just aM - bM = (a-b)M
 
  Ah, sounds like a good idea, I guess I'll try that.
 
  [...]
 
  --
  Clément B.
 
  Updated the patch. How should I finally go about converting rgb to yuv?
 
 
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 From 905fb0f15033e447fc344097dd649ca671fa1074 Mon Sep 17 00:00:00 2001
 From: Arwa Arif arwaarif1...@gmail.com
 Date: Thu, 30 Oct 2014 18:05:45 +0530
 Subject: [PATCH] [PATCH]lvafi: add xbr filter

lavfi: add xbr filter

 
 ---

  doc/filters.texi |   81 ++--

what happened with this? Did you edit the patch by hand? I can't apply
the patch...

  libavfilter/Makefile |1 +
  libavfilter/allfilters.c |1 +
  libavfilter/vf_xbr.c |  319 
 ++
  4 files changed, 335 insertions(+), 67 deletions(-)
  create mode 100644 libavfilter/vf_xbr.c
 
[...]
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[FFmpeg-devel] [PATCH] psy_snr:Psychoacoustic SNR for audio files

2014-10-30 Thread Senjuti Kundu 
Implemented an SNR for audio files which takes into a account the 
psychoacoustic masking. This results in an SNR which is closer to how humans 
percieve sound, compared to tiny_psnr which directly compares audio signals

Signed-off-by: Senjuti Kundu senjutikund...@gmail.com
---
 tests/psy_snr.c | 421 
 1 file changed, 421 insertions(+)
 create mode 100644 tests/psy_snr.c

diff --git a/tests/psy_snr.c b/tests/psy_snr.c
new file mode 100644
index 000..94041ed
--- /dev/null
+++ b/tests/psy_snr.c
@@ -0,0 +1,421 @@
+/*
+ * Copyright (c) 2003 Michael Niedermayer michae...@gmx.at
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+/*
+ * Psy-SNR - Psychoacoustic SNR for audio files.
+ * Author - Senjuti Kundu senjutikund...@gmail.com
+ * Input format - tiny_psysnr file1 file2
+ *  [elem size|u8|s16|f32|f64 [shift [skip bytes [shift 
search range
+ * WAV headers are skipped automatically.
+ * SIZE can be changed to adjust window size as need be.
+ * compile using gcc psysnr.c $(pkg-config --cflags --libs libavformat 
libavcodec)
+ * -I /usr/local/include/libavcodec/
+ */
+
+#include stdio.h
+#include stdlib.h
+#include string.h
+#include inttypes.h
+#include math.h
+#include float.h
+#include limits.h
+
+#include libavutil/intfloat.h
+#include libavutil/intreadwrite.h
+#include libavcodec/avfft.h
+#include libavutil/mem.h
+
+#define FFMIN(a, b) ((a)  (b) ? (b) : (a))
+#define FFMAX(a, b) ((a)  (b) ? (a) : (b))
+#define F 100
+//size should be close to 20k
+#define SIZE 1024
+#define db_fw_rollof 4
+#define db_bw_rollof 4
+#define db_attenuation 0.1
+
+uint64_t exp16_table[21] = {
+   65537,
+   65538,
+   65540,
+   65544,
+   65552,
+   65568,
+   65600,
+   65664,
+   65793,
+   66050,
+   66568,
+   67616,
+   69763,
+   74262,
+   84150,
+  108051,
+  178145,
+  484249,
+ 3578144,
+   195360063,
+582360139072LL,
+};
+
+#if 0
+// 16.16 fixpoint exp()
+static unsigned int exp16(unsigned int a){
+int i;
+int out= 116;
+
+for(i=19;i=0;i--){
+if(a(1i))
+out= (out*exp16_table[i] + (115))16;
+}
+
+return out;
+}
+#endif
+
+// 16.16 fixpoint log()
+static int64_t log16(uint64_t a)
+{
+int i;
+int out = 0;
+
+if (a  1  16)
+return -log16((1LL  32) / a);
+a = 16;
+
+for (i = 20; i = 0; i--) {
+int64_t b = exp16_table[i];
+if (a  (b  16))
+continue;
+out |= 1  i;
+a= ((a / b)  16) + (((a % b)  16) + b / 2) / b;
+}
+return out;
+}
+
+static uint64_t int_sqrt(uint64_t a)
+{
+uint64_t ret= 0;
+uint64_t ret_sq = 0;
+int s;
+
+for (s = 31; s = 0; s--) {
+uint64_t b = ret_sq + (1ULL  (s * 2)) + (ret  s) * 2;
+if (b = a) {
+ret_sq = b;
+ret   += 1ULL  s;
+}
+}
+return ret;
+}
+
+static int16_t get_s16l(uint8_t *p)
+{
+union {
+uint16_t u;
+int16_t  s;
+} v;
+v.u = p[0] | p[1]  8;
+return v.s;
+}
+
+static float get_f32l(uint8_t *p)
+{
+union av_intfloat32 v;
+v.i = p[0] | p[1]  8 | p[2]  16 | p[3]  24;
+return v.f;
+}
+
+static double get_f64l(uint8_t *p)
+{
+return av_int2double(AV_RL64(p));
+}
+
+static float* get_mask_array(int tempsize){
+//modelling the mask function as a parabole. Others can be
+//explored as need be. y=(-(x-mid)2+c)/c
+int i = 0;
+float* maskingfunc = malloc(tempsize*sizeof(float));
+maskingfunc[tempsize/2] = exp(-db_attenuation*log(10));
+for (i = (tempsize/2)+1; itempsize; i++){
+maskingfunc[i] = 
maskingfunc[i-1]*exp(-(db_fw_rollof*log(10))/(20*(i-(tempsize/2;;
+}
+for(i = (tempsize/2)-1; i = 0; i--){
+maskingfunc[i] = 
maskingfunc[i+1]*exp(-(2*db_bw_rollof*log(10))/(20*((tempsize/2)-i)));
+}
+return maskingfunc;
+}
+
+static float* get_mask(FFTComplex* a, int tempsize, float* maskingfunc){
+int i = 0;
+int j = 0;
+float* mask = malloc(tempsize*sizeof(float));
+float self = 0;
+float next = 0;
+float prev = 0;
+
+for (i =

Re: [FFmpeg-devel] Encode2() function in libavcodec/avcodec.h

2014-10-30 Thread Michael Niedermayer 
On Thu, Oct 30, 2014 at 11:31:25AM +0530, greeshma wrote:
 Hi,
 
 the attributes has been changed to (AVCodecContext *avctx, AVPacket *avpkt,
 const AVFrame *frame, int *got_packet_ptr);
 
 what is the exact change made in the function.

if you want to know which commit introduced a change there is
git blame and git log -S / -G

[...]
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Re: [FFmpeg-devel] Use ffmpeg as an Android stagefright plugin

2014-10-30 Thread Fabio Fumi 
Also after adding new ffmpeg codecs to I still get a crash on H264 (avc) 
playback.

Added ffmpeg codecs to device/renesas/emev/media_codecs.xml 

Rebuilt... test... still crashing:

V/AwesomePlayer(  758): track of type 'video/avc' does not publish bitrate
V/AwesomePlayer(  758): mBitrate = -1 bits/sec
V/AwesomePlayer(  758): haveAudio:1, haveVideo:1
V/AwesomePlayer(  758): initVideoDecoder flags=0x0
I/EV2OMXPlugin(  758): ***run OMFPlugin --- makeComponentInstance 

I/OMXCodec(  758): [OMX.RENESAS.VIDEO.DECODER.H264] AVC profile = 66 
(Baseline), level = 30
I/OMXCodec(  758): [OMX.RENESAS.VIDEO.DECODER.H264] video dimensions are 
640 x 344
E/OMXNodeInstance(  758): OMX_GetExtensionIndex (index:'7fff') failed
I/EV2OMXPlugin(  758): ***run OMFPlugin --- destroyComponentInstance 

E/OMXMaster(  758): Invalid OMX component name 'OMX.ffmpeg.h264.decoder'
E/MediaPlayer(  996): error (1, -2147483648)
E/MediaPlayer(  996): Error (1,-2147483648)
W/YouTube (  996): (unknown) MediaPlayer error during prepare [what=1, 
extra=-2147483648]
W/YouTube (  996): (unknown) Retrying MediaPlayer error [retry=1, max=3]
F/libc(  758): @@@ ABORTING: INVALID HEAP ADDRESS IN dlfree 
addr=0x2a023ad0
F/libc(  758): Fatal signal 11 (SIGSEGV) at 0xdeadbaad (code=1), thread 
794 (Binder_2)
...
I/DEBUG   (   70): #00  pc 000137b8  /system/lib/libc.so
I/DEBUG   (   70): #01  pc 00015b95  /system/lib/libc.so (dlfree+1628)
I/DEBUG   (   70): #02  pc 00016d43  /system/lib/libc.so (free+10)
I/DEBUG   (   70): #03  pc 0007a853  /system/lib/libstagefright.so
I/DEBUG   (   70): #04  pc 0007ac0d  /system/lib/libstagefright.so 
(android::TimedEventQueue::~TimedEventQueue()+36)
I/DEBUG   (   70): #05  pc 000538ad  /system/lib/libstagefright.so 
(android::AwesomePlayer::~AwesomePlayer()+288)

On Thursday, 30 October 2014 03:09:35 UTC-7, Fabio Fumi wrote:

 Hi Chih,

 I've completed the merge of frameworks/av and framewroks/native and added 
 the stagefright-plugins and ffmpeg to my project. I enabld the USES_NAM and 
 built new system.

 Startup wnet fine, but on YouTube test playback (which went just fine 
 before), I get a crash (below).

 Note how I haven't enabled he new ffmpeg codecs in media_codecs.xml and 
 I'm using Google SW codec for h264:

 MediaCodec name=OMX.google.h264.decoder type=video/avc/

 Suggestions?

 thanks
 Fabio


 ...
 D/H264Dec (  707): H264SwDecInit#
 D/H264Dec (  707): H264SwDecInit# decInst 0x2a055c08 noOutputReordering 0
 D/H264Dec (  707): H264SwDecInit# OK: return 0x2a0b1238
 I/OMXCodec(  707): [OMX.google.h264.decoder] AVC profile = 66 (Baseline), 
 level = 30
 I/OMXCodec(  707): [OMX.google.h264.decoder] video dimensions are 320 x 240
 I/OMXCodec(  707): [OMX.google.h264.decoder] Crop rect is 320 x 240 @ (0, 
 0)
 ...
 D/H264Dec (  707): H264SwDecDecode#
 D/H264Dec (  707): H264SwDecDecode# decInst 0x2a0b1238  pInput 0x4201ed88  
 pOutput 0x4201ed7c
 D/H264_decoder(  707): Access unit boundary
 D/H264_decoder(  707): SEQ PARAM SET
 D/H264Dec (  707): H264SwDecDecode# OK: DecResult 1
 D/H264Dec (  707): H264SwDecDecode#
 D/H264Dec (  707): H264SwDecDecode# decInst 0x2a0b1238  pInput 0x4201ed88  
 pOutput 0x4201ed7c
 D/H264_decoder(  707): Access unit boundary
 D/H264_decoder(  707): PIC PARAM SET
 D/H264Dec (  707): H264SwDecDecode# OK: DecResult 1
 D/H264Dec (  707): H264SwDecDecode#
 D/H264Dec (  707): H264SwDecDecode# decInst 0x2a0b1238  pInput 0x4201ed88  
 pOutput 0x4201ed7c
 D/H264_decoder(  707): Access unit boundary
 D/H264_decoder(  707): IDR 
 D/H264_decoder(  707): SLICE HEADER
 D/H264Dec (  707): H264SwDecDecode# OK: DecResult 4
 D/H264Dec (  707): H264SwDecGetInfo#
 D/H264Dec (  707): H264SwDecGetInfo# decInst 0x2a0b1238  pDecInfo 
 0x4201eda8
 D/H264Dec (  707): H264SwDecGetInfo# OK
 D/H264Dec (  707): H264SwDecDecode#
 D/H264Dec (  707): H264SwDecDecode# decInst 0x2a0b1238  pInput 0x4201ed88  
 pOutput 0x4201ed7c
 D/H264_decoder(  707): IDR 
 D/H264_decoder(  707): SLICE HEADER
 D/H264_decoder(  707): (null)
 D/H264Dec (  707): H264SwDecDecode# OK: DecResult 2
 D/H264Dec (  707): H264SwDecNextPicture#
 D/H264Dec (  707): H264SwDecNextPicture# decInst 0x2a0b1238 pOutput 
 0x4201ed98 flushBuffer 0
 D/H264Dec (  707): H264SwDecNextPicture# OK: return H264SWDEC_PIC_RDY
 F/libc(  707): Fatal signal 11 (SIGSEGV) at 0x0008 (code=1), 
 thread 1818 (le.h264.decoder)
 ...
 I/DEBUG   (   70): #00  pc 00011ed0  /system/lib/libc.so
 I/DEBUG   (   70): #01  pc ad6b  /system/lib/libstagefright_omx.so 
 (android::OMX::CallbackDispatcher::post(android::omx_message const)+14)
 I/DEBUG   (   70): #02  pc b5e5  /system/lib/libstagefright_omx.so 
 (android::OMX::OnEmptyBufferDone(void*, OMX_BUFFERHEADERTYPE*)+30)
 I/DEBUG   (   70): #03  pc ebd9  /system/lib/libstagefright_omx.so 
 (android::SoftOMXComponent::notifyEmptyBufferDone(OMX_BUFFERHEADERTYPE*)+16)
 I/DEBUG   (   70): #04  pc 9f60

Re: [FFmpeg-devel] Use ffmpeg as an Android stagefright plugin

2014-10-30 Thread Fabio Fumi 
Hi Chih,

I've completed the merge of frameworks/av and framewroks/native and added 
the stagefright-plugins and ffmpeg to my project. I enabld the USES_NAM and 
built new system.

Startup wnet fine, but on YouTube test playback (which went just fine 
before), I get a crash (below).

Note how I haven't enabled he new ffmpeg codecs in media_codecs.xml and I'm 
using Google SW codec for h264:

MediaCodec name=OMX.google.h264.decoder type=video/avc/

Suggestions?

thanks
Fabio


...
D/H264Dec (  707): H264SwDecInit#
D/H264Dec (  707): H264SwDecInit# decInst 0x2a055c08 noOutputReordering 0
D/H264Dec (  707): H264SwDecInit# OK: return 0x2a0b1238
I/OMXCodec(  707): [OMX.google.h264.decoder] AVC profile = 66 (Baseline), 
level = 30
I/OMXCodec(  707): [OMX.google.h264.decoder] video dimensions are 320 x 240
I/OMXCodec(  707): [OMX.google.h264.decoder] Crop rect is 320 x 240 @ (0, 0)
...
D/H264Dec (  707): H264SwDecDecode#
D/H264Dec (  707): H264SwDecDecode# decInst 0x2a0b1238  pInput 0x4201ed88  
pOutput 0x4201ed7c
D/H264_decoder(  707): Access unit boundary
D/H264_decoder(  707): SEQ PARAM SET
D/H264Dec (  707): H264SwDecDecode# OK: DecResult 1
D/H264Dec (  707): H264SwDecDecode#
D/H264Dec (  707): H264SwDecDecode# decInst 0x2a0b1238  pInput 0x4201ed88  
pOutput 0x4201ed7c
D/H264_decoder(  707): Access unit boundary
D/H264_decoder(  707): PIC PARAM SET
D/H264Dec (  707): H264SwDecDecode# OK: DecResult 1
D/H264Dec (  707): H264SwDecDecode#
D/H264Dec (  707): H264SwDecDecode# decInst 0x2a0b1238  pInput 0x4201ed88  
pOutput 0x4201ed7c
D/H264_decoder(  707): Access unit boundary
D/H264_decoder(  707): IDR 
D/H264_decoder(  707): SLICE HEADER
D/H264Dec (  707): H264SwDecDecode# OK: DecResult 4
D/H264Dec (  707): H264SwDecGetInfo#
D/H264Dec (  707): H264SwDecGetInfo# decInst 0x2a0b1238  pDecInfo 0x4201eda8
D/H264Dec (  707): H264SwDecGetInfo# OK
D/H264Dec (  707): H264SwDecDecode#
D/H264Dec (  707): H264SwDecDecode# decInst 0x2a0b1238  pInput 0x4201ed88  
pOutput 0x4201ed7c
D/H264_decoder(  707): IDR 
D/H264_decoder(  707): SLICE HEADER
D/H264_decoder(  707): (null)
D/H264Dec (  707): H264SwDecDecode# OK: DecResult 2
D/H264Dec (  707): H264SwDecNextPicture#
D/H264Dec (  707): H264SwDecNextPicture# decInst 0x2a0b1238 pOutput 
0x4201ed98 flushBuffer 0
D/H264Dec (  707): H264SwDecNextPicture# OK: return H264SWDEC_PIC_RDY
F/libc(  707): Fatal signal 11 (SIGSEGV) at 0x0008 (code=1), thread 
1818 (le.h264.decoder)
...
I/DEBUG   (   70): #00  pc 00011ed0  /system/lib/libc.so
I/DEBUG   (   70): #01  pc ad6b  /system/lib/libstagefright_omx.so 
(android::OMX::CallbackDispatcher::post(android::omx_message const)+14)
I/DEBUG   (   70): #02  pc b5e5  /system/lib/libstagefright_omx.so 
(android::OMX::OnEmptyBufferDone(void*, OMX_BUFFERHEADERTYPE*)+30)
I/DEBUG   (   70): #03  pc ebd9  /system/lib/libstagefright_omx.so 
(android::SoftOMXComponent::notifyEmptyBufferDone(OMX_BUFFERHEADERTYPE*)+16)
I/DEBUG   (   70): #04  pc 9f60  
/system/lib/libstagefright_soft_h264dec.so 
(android::SoftAVC::onQueueFilled(unsigned long)+508)



On Sunday, 11 August 2013 01:25:25 UTC-7, Chih-Wei Huang wrote:

 Hello ffmpeg list, 
 I'm planning to integrate the latest ffmpeg as 
 a stagefright OMXPlugin with Android-x86 4.3. 
 I found the stagefright-plugins developed by 
 Michael Chen is a good start. 
 (repo: https://github.com/omxcodec/stagefright-plugins.git) 
 But it's based on android 4.0 (ICS). 

 My plan is: 
 * Update stagefright-plugins to work with 
   stagefright of android 4.3. 
 * Make it to be a libstagefrighthw.so plugin. 
 * Create Android.mk to build ffmpeg libs. 

 Before I go ahead, I'd like to know 
 if anyone is doing a similar project 
 to avoid duplicate work. 
 Thanks a lot! 


 -- 
 Chih-Wei 
 Android-x86 project 
 http://www.android-x86.org 

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Re: [FFmpeg-devel] [PATCH] psy_snr:Psychoacoustic SNR for audio files

2014-10-30 Thread Michael Niedermayer 
On Thu, Oct 30, 2014 at 11:33:51PM +0530, Senjuti Kundu wrote:
 Implemented an SNR for audio files which takes into a account the 
 psychoacoustic masking. This results in an SNR which is closer to how humans 
 percieve sound, compared to tiny_psnr which directly compares audio signals
 
 Signed-off-by: Senjuti Kundu senjutikund...@gmail.com
 ---
  tests/psy_snr.c | 421 
 
  1 file changed, 421 insertions(+)
  create mode 100644 tests/psy_snr.c
 
 diff --git a/tests/psy_snr.c b/tests/psy_snr.c
 new file mode 100644
 index 000..94041ed
 --- /dev/null
 +++ b/tests/psy_snr.c
 @@ -0,0 +1,421 @@
 +/*
 + * Copyright (c) 2003 Michael Niedermayer michae...@gmx.at
 + *
 + * This file is part of FFmpeg.
 + *
 + * FFmpeg is free software; you can redistribute it and/or
 + * modify it under the terms of the GNU Lesser General Public
 + * License as published by the Free Software Foundation; either
 + * version 2.1 of the License, or (at your option) any later version.
 + *
 + * FFmpeg is distributed in the hope that it will be useful,
 + * but WITHOUT ANY WARRANTY; without even the implied warranty of
 + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
 + * Lesser General Public License for more details.
 + *
 + * You should have received a copy of the GNU Lesser General Public
 + * License along with FFmpeg; if not, write to the Free Software
 + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 
 USA
 + */
 +
 +/*
 + * Psy-SNR - Psychoacoustic SNR for audio files.
 + * Author - Senjuti Kundu senjutikund...@gmail.com
 + * Input format - tiny_psysnr file1 file2
 + *  [elem size|u8|s16|f32|f64 [shift [skip bytes [shift 
 search range
 + * WAV headers are skipped automatically.
 + * SIZE can be changed to adjust window size as need be.
 + * compile using gcc psysnr.c $(pkg-config --cflags --libs libavformat 
 libavcodec)
 + * -I /usr/local/include/libavcodec/
 + */
 +
 +#include stdio.h
 +#include stdlib.h
 +#include string.h
 +#include inttypes.h
 +#include math.h
 +#include float.h
 +#include limits.h
 +
 +#include libavutil/intfloat.h
 +#include libavutil/intreadwrite.h
 +#include libavcodec/avfft.h
 +#include libavutil/mem.h
 +
 +#define FFMIN(a, b) ((a)  (b) ? (b) : (a))
 +#define FFMAX(a, b) ((a)  (b) ? (a) : (b))
 +#define F 100
 +//size should be close to 20k
 +#define SIZE 1024

 +#define db_fw_rollof 4
 +#define db_bw_rollof 4
 +#define db_attenuation 0.1

#defines
 should be all uppercase


 +
 +uint64_t exp16_table[21] = {
 +   65537,
 +   65538,
 +   65540,
 +   65544,
 +   65552,
 +   65568,
 +   65600,
 +   65664,
 +   65793,
 +   66050,
 +   66568,
 +   67616,
 +   69763,
 +   74262,
 +   84150,
 +  108051,
 +  178145,
 +  484249,
 + 3578144,
 +   195360063,
 +582360139072LL,
 +};
 +
 +#if 0
 +// 16.16 fixpoint exp()
 +static unsigned int exp16(unsigned int a){
 +int i;
 +int out= 116;
 +
 +for(i=19;i=0;i--){
 +if(a(1i))
 +out= (out*exp16_table[i] + (115))16;
 +}
 +
 +return out;
 +}
 +#endif
 +
 +// 16.16 fixpoint log()
 +static int64_t log16(uint64_t a)
 +{
 +int i;
 +int out = 0;
 +
 +if (a  1  16)
 +return -log16((1LL  32) / a);
 +a = 16;
 +
 +for (i = 20; i = 0; i--) {
 +int64_t b = exp16_table[i];
 +if (a  (b  16))
 +continue;
 +out |= 1  i;
 +a= ((a / b)  16) + (((a % b)  16) + b / 2) / b;
 +}
 +return out;
 +}
 +
 +static uint64_t int_sqrt(uint64_t a)
 +{
 +uint64_t ret= 0;
 +uint64_t ret_sq = 0;
 +int s;
 +
 +for (s = 31; s = 0; s--) {
 +uint64_t b = ret_sq + (1ULL  (s * 2)) + (ret  s) * 2;
 +if (b = a) {
 +ret_sq = b;
 +ret   += 1ULL  s;
 +}
 +}
 +return ret;
 +}
 +

these functions look duplicated from tiny_psnr, it would be better
to share them


 +static int16_t get_s16l(uint8_t *p)
 +{
 +union {
 +uint16_t u;
 +int16_t  s;
 +} v;
 +v.u = p[0] | p[1]  8;
 +return v.s;
 +}
 +
 +static float get_f32l(uint8_t *p)
 +{
 +union av_intfloat32 v;
 +v.i = p[0] | p[1]  8 | p[2]  16 | p[3]  24;
 +return v.f;
 +}
 +
 +static double get_f64l(uint8_t *p)
 +{
 +return av_int2double(AV_RL64(p));
 +}
 +
 +static float* get_mask_array(int tempsize){
 +//modelling the mask function as a parabole. Others can be
 +//explored as need be. y=(-(x-mid)2+c)/c
 +int i = 0;
 +float* maskingfunc = malloc(tempsize*sizeof(float));

 +maskingfunc[tempsize/2] = exp(-db_attenuation*log(10));


 +for (i = (tempsize/2)+1; itempsize; i++){
 +maskingfunc[i] = 
 maskingfunc[i-1]*exp(-(db_fw_rollof*log(10))/(20*(i-(tempsize/2;;

double ;



 +}
 +for(i = (tempsize/2)-1;

Re: [FFmpeg-devel] [PATCH] lavfi: add xbr filter

2014-10-30 Thread arwa arif 
On Thu, Oct 30, 2014 at 9:34 PM, Stefano Sabatini stefa...@gmail.com
wrote:

 On date Thursday 2014-10-30 18:23:35 +0530, arwa arif encoded:
  On Wed, Oct 29, 2014 at 3:36 AM, Clément Bœsch u...@pkh.me wrote:
 
   On Tue, Oct 28, 2014 at 10:51:27PM +0100, Michael Niedermayer wrote:
On Tue, Oct 28, 2014 at 07:16:45PM +0100, Clément Bœsch wrote:
 On Tue, Oct 28, 2014 at 06:30:34PM +0100, Stefano Sabatini wrote:
 [...]
  How much effort would it take to implement the remaining scaling
   modes?
 

 According to
 https://ffmpeg.org/pipermail/ffmpeg-devel/2014-October/164574.html

 I think 4x can be done fast enough, but 3x will take time.

 [...]
   +typedef struct {
   +uint32_t rgbtoyuv[124];
 
  We should avoid this 64MiB. Also the table should be possibly
 static,
  so you don't have to fill it per each xBR instance.
 

 So, I requested to do it exactly the same as HQx because this part
 is
 common according to the specifications. This should be kept the
 same
 vf_hqx, and then factorized.

   
 Now about removing this allocation, I did benchmark this LUT vs
 computation (see attached patch for comp. version). And the
 problem is
 that it's slightly slower, probably due to the /1000.
   
why do you divide at all ?
cant you do the computations with full precission ?
  
   I wasn't able to... but I was probably doing it wrong.
  
   And anyway, so far I observed this:
 lut: 127fps
 nolut+div:   119fps
 nolut+nodiv: 123fps
  
   So even with fast computation, it's still slower than the LUT. It
   probably
   doesn't matter that much in practice, and dropping that huge table
 might be
   worth the performance impact (feel free to discuss).
  
   Note that looking at the original code (which was working on rgb565
 only),
   it was bitexact. The rgb 24-bit was added in the modern hqx with
 float
   point. So we can probably tolerate the inaccuracy. Still, if you find a
   way of keeping full accuracy with the modern implementation...
  
   Typically, I tried stuff like this:
  
 const uint32_t y = (uint32_t)((1225*r + 2404*g +  467*b + (111)) 
   12);
 const uint32_t u = (uint32_t)((-692*r - 1356*g + 2048*b + (111)) 
   12) + 128;
 const uint32_t v = (uint32_t)((2048*r - 1716*g -  332*b + (111)) 
   12) + 128;
  
   ...but I'm probably doing it very wrong somewhere (sign issue maybe?),
   haven't
   looked deeper. I went up to 15 bits, still didn't match, so I was
 probably
   doing something stupid.
  
also instead of doing 2 rgb2yuv and then taking their difference you
can do the difference in rgb space and convert the rgb difference to
a yuv difference
its just aM - bM = (a-b)M
  
   Ah, sounds like a good idea, I guess I'll try that.
  
   [...]
  
   --
   Clément B.
  
   Updated the patch. How should I finally go about converting rgb to yuv?
 
 
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  From 905fb0f15033e447fc344097dd649ca671fa1074 Mon Sep 17 00:00:00 2001
  From: Arwa Arif arwaarif1...@gmail.com
  Date: Thu, 30 Oct 2014 18:05:45 +0530
  Subject: [PATCH] [PATCH]lvafi: add xbr filter

 lavfi: add xbr filter

 
  ---

   doc/filters.texi |   81 ++--

 what happened with this? Did you edit the patch by hand? I can't apply
 the patch...

   libavfilter/Makefile |1 +
   libavfilter/allfilters.c |1 +
   libavfilter/vf_xbr.c |  319
 ++
   4 files changed, 335 insertions(+), 67 deletions(-)
   create mode 100644 libavfilter/vf_xbr.c
 
 [...]
 --
 FFmpeg = Fostering Fundamental Minimal Plastic Enhanced Geisha
 ___
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From b9b56b594f856fef8b113b283df2d2045e8357f7 Mon Sep 17 00:00:00 2001
From: Arwa Arif arwaarif1...@gmail.com
Date: Thu, 30 Oct 2014 22:06:20 +0530
Subject: [PATCH] [PATCH]lvafi: add xbr filter

---
 doc/filters.texi |5 +
 libavfilter/Makefile |1 +
 libavfilter/allfilters.c |1 +
 libavfilter/vf_xbr.c |  319 ++
 4 files changed, 326 insertions(+)
 create mode 100644 libavfilter/vf_xbr.c

diff --git a/doc/filters.texi b/doc/filters.texi
index 7be29de..2905e5d 100644
--- a/doc/filters.texi
+++ b/doc/filters.texi
@@ -9163,6 +9163,11 @@ Only deinterlace frames marked as interlaced.
 Default value is @samp{all}.
 @end table
 
+@section xbr
+Apply high-quality magnification filter which is designed for pixel art. It follows a set
+of edge-detection rules @url{http://www.libretro.com/forums/viewtopic.php?f=6t=134}.
+This filter was originally created by Hyllian.
+
 @anchor{yadif}
 @section yadif
 
diff --git a/libavfilter/Makefile

[FFmpeg-devel] [PATCH] libxcb-xshape in configure.

2014-10-30 Thread Horváth Balázs 
This is my attempt at fixing the broken build because xcbgrab
uses xcb_shape_rectangles, but doesn't link xcb-shape.

Signed-off-by: Horváth Balázs q...@qroa.ch
---
 configure | 11 +--
 1 file changed, 9 insertions(+), 2 deletions(-)

diff --git a/configure b/configure
index e6e3de3..49feee3 100755
--- a/configure
+++ b/configure
@@ -254,6 +254,7 @@ External library support:
   --enable-libxcb  enable X11 grabbing using XCB [auto]
   --enable-libxcb-shm  enable X11 grabbing shm communication [auto]
   --enable-libxcb-xfixes   enable X11 grabbing mouse rendering [auto]
+  --enable-libxcb-xshape   enable X11 grabbing shape rendering [auto]
   --enable-libxvid enable Xvid encoding via xvidcore,
native MPEG-4/Xvid encoder exists [no]
   --enable-libzmq  enable message passing via libzmq [no]
@@ -1387,6 +1388,7 @@ EXTERNAL_LIBRARY_LIST=
 libxcb
 libxcb_shm
 libxcb_xfixes
+libxcb_shape
 libxvid
 libzmq
 libzvbi
@@ -5040,8 +5042,13 @@ if enabled libxcb; then
 enabled libxcb_xfixes  die ERROR: libxcb_xfixes not found;
 }  enable libxcb_xfixes; }
 
-add_cflags $xcb_event_cflags $xcb_shm_cflags $xcb_xfixes_cflags
-add_extralibs $xcb_event_libs $xcb_shm_libs $xcb_xfixes_libs
+disabled libxcb_shape || {
+check_pkg_config xcb-shape xcb/shape.h xcb_shape_rectangles || {
+enabled libxcb_shape  die ERROR: libxcb_shape not found;
+}  enable libxcb_shape; }
+
+add_cflags $xcb_event_cflags $xcb_shm_cflags $xcb_xfixes_cflags 
$xcb_shape_cflags
+add_extralibs $xcb_event_libs $xcb_shm_libs $xcb_xfixes_libs 
$xcb_shape_cflags
 fi
 fi
 
-- 
2.1.3

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[FFmpeg-devel] [PATCH] libxcb-xhsape in configure

2014-10-30 Thread Horváth Balázs 
This is my attempt at fixing the broken build because xcbgrab
uses xcb_shape_rectangles, but doesn't link xcb-shape.

Horváth Balázs (1):
  libxcb-xshape in configure.

 configure | 11 +--
 1 file changed, 9 insertions(+), 2 deletions(-)

-- 
2.1.3

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Re: [FFmpeg-devel] [PATCH] libxcb-xshape in configure.

2014-10-30 Thread Michael Niedermayer 
On Thu, Oct 30, 2014 at 09:26:17PM +0100, Horváth Balázs wrote:
 This is my attempt at fixing the broken build because xcbgrab
 uses xcb_shape_rectangles, but doesn't link xcb-shape.
 
 Signed-off-by: Horváth Balázs q...@qroa.ch
 ---
  configure | 11 +--
  1 file changed, 9 insertions(+), 2 deletions(-)
 
 diff --git a/configure b/configure
 index e6e3de3..49feee3 100755
 --- a/configure
 +++ b/configure
 @@ -254,6 +254,7 @@ External library support:
--enable-libxcb  enable X11 grabbing using XCB [auto]
--enable-libxcb-shm  enable X11 grabbing shm communication [auto]
--enable-libxcb-xfixes   enable X11 grabbing mouse rendering [auto]
 +  --enable-libxcb-xshape   enable X11 grabbing shape rendering [auto]
--enable-libxvid enable Xvid encoding via xvidcore,
 native MPEG-4/Xvid encoder exists [no]
--enable-libzmq  enable message passing via libzmq [no]
 @@ -1387,6 +1388,7 @@ EXTERNAL_LIBRARY_LIST=
  libxcb
  libxcb_shm
  libxcb_xfixes
 +libxcb_shape
  libxvid
  libzmq
  libzvbi
 @@ -5040,8 +5042,13 @@ if enabled libxcb; then
  enabled libxcb_xfixes  die ERROR: libxcb_xfixes not found;
  }  enable libxcb_xfixes; }
  
 -add_cflags $xcb_event_cflags $xcb_shm_cflags $xcb_xfixes_cflags
 -add_extralibs $xcb_event_libs $xcb_shm_libs $xcb_xfixes_libs
 +disabled libxcb_shape || {
 +check_pkg_config xcb-shape xcb/shape.h xcb_shape_rectangles || {
 +enabled libxcb_shape  die ERROR: libxcb_shape not found;
 +}  enable libxcb_shape; }
 +
 +add_cflags $xcb_event_cflags $xcb_shm_cflags $xcb_xfixes_cflags 
 $xcb_shape_cflags
 +add_extralibs $xcb_event_libs $xcb_shm_libs $xcb_xfixes_libs 
 $xcb_shape_cflags

doesnt this still break build if someone uses
--disable-libxcb-xshape
?

[...]
-- 
Michael GnuPG fingerprint: 9FF2128B147EF6730BADF133611EC787040B0FAB

Frequently ignored answer#1 FFmpeg bugs should be sent to our bugtracker. User
questions about the command line tools should be sent to the ffmpeg-user ML.
And questions about how to use libav* should be sent to the libav-user ML.


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Re: [FFmpeg-devel] [PATCH 1/3] ffplay: implement separete audio decoder thread

2014-10-30 Thread Marton Balint 


On Thu, 30 Oct 2014, wm4 wrote:


On Thu, 30 Oct 2014 00:31:25 +0100
Marton Balint c...@passwd.hu wrote:


Signed-off-by: Marton Balint c...@passwd.hu
---
 ffplay.c | 265 ---
 1 file changed, 153 insertions(+), 112 deletions(-)

diff --git a/ffplay.c b/ffplay.c
index a979164..24bcae2 100644
--- a/ffplay.c
+++ b/ffplay.c
@@ -121,7 +121,8 @@ typedef struct PacketQueue {

 #define VIDEO_PICTURE_QUEUE_SIZE 3
 #define SUBPICTURE_QUEUE_SIZE 16
-#define FRAME_QUEUE_SIZE FFMAX(VIDEO_PICTURE_QUEUE_SIZE, SUBPICTURE_QUEUE_SIZE)
+#define SAMPLE_QUEUE_SIZE 9
+#define FRAME_QUEUE_SIZE FFMAX(SAMPLE_QUEUE_SIZE, 
FFMAX(VIDEO_PICTURE_QUEUE_SIZE, SUBPICTURE_QUEUE_SIZE))

 typedef struct AudioParams {
 int freq;
@@ -196,6 +197,7 @@ typedef struct Decoder {
 typedef struct VideoState {
 SDL_Thread *read_tid;
 SDL_Thread *video_tid;
+SDL_Thread *audio_tid;
 AVInputFormat *iformat;
 int no_background;
 int abort_request;
@@ -217,6 +219,7 @@ typedef struct VideoState {

 FrameQueue pictq;
 FrameQueue subpq;
+FrameQueue sampq;

 Decoder auddec;
 Decoder viddec;
@@ -242,8 +245,6 @@ typedef struct VideoState {
 unsigned int audio_buf1_size;
 int audio_buf_index; /* in bytes */
 int audio_write_buf_size;
-int audio_buf_frames_pending;
-int audio_last_serial;
 struct AudioParams audio_src;
 #if CONFIG_AVFILTER
 struct AudioParams audio_filter_src;
@@ -252,7 +253,6 @@ typedef struct VideoState {
 struct SwrContext *swr_ctx;
 int frame_drops_early;
 int frame_drops_late;
-AVFrame *frame;

 enum ShowMode {
 SHOW_MODE_NONE = -1, SHOW_MODE_VIDEO = 0, SHOW_MODE_WAVES, 
SHOW_MODE_RDFT, SHOW_MODE_NB
@@ -712,12 +712,29 @@ static Frame *frame_queue_peek_writable(FrameQueue *f)
 return f-queue[f-windex];
 }

+static Frame *frame_queue_peek_readable(FrameQueue *f)
+{
+/* wait until we have a readable a new frame */
+SDL_LockMutex(f-mutex);
+while (f-size - f-rindex_shown = 0 
+   !f-pktq-abort_request) {
+SDL_CondWait(f-cond, f-mutex);
+}
+SDL_UnlockMutex(f-mutex);
+
+if (f-pktq-abort_request)
+return NULL;
+
+return f-queue[(f-rindex + f-rindex_shown) % f-max_size];
+}
+
 static void frame_queue_push(FrameQueue *f)
 {
 if (++f-windex == f-max_size)
 f-windex = 0;
 SDL_LockMutex(f-mutex);
 f-size++;
+SDL_CondSignal(f-cond);
 SDL_UnlockMutex(f-mutex);
 }

@@ -1280,6 +1297,7 @@ static void stream_close(VideoState *is)

 /* free all pictures */
 frame_queue_destory(is-pictq);
+frame_queue_destory(is-sampq);
 frame_queue_destory(is-subpq);
 SDL_DestroyCond(is-continue_read_thread);
 #if !CONFIG_AVFILTER
@@ -2100,6 +2118,93 @@ end:
 }
 #endif  /* CONFIG_AVFILTER */

+static int audio_thread(void *arg)
+{
+VideoState *is = arg;
+AVFrame *frame = av_frame_alloc();
+Frame *af;
+#if CONFIG_AVFILTER
+int last_serial = -1;
+int64_t dec_channel_layout;
+int reconfigure;
+#endif
+int got_frame = 0;
+AVRational tb;
+int ret = 0;
+
+if (!frame)
+return AVERROR(ENOMEM);
+
+do {
+if ((got_frame = decoder_decode_frame(is-auddec, frame, NULL))  0)
+goto the_end;
+
+if (got_frame) {
+tb = (AVRational){1, frame-sample_rate};
+
+#if CONFIG_AVFILTER
+dec_channel_layout = 
get_valid_channel_layout(frame-channel_layout, av_frame_get_channels(frame));
+
+reconfigure =
+cmp_audio_fmts(is-audio_filter_src.fmt, 
is-audio_filter_src.channels,
+   frame-format, 
av_frame_get_channels(frame))||
+is-audio_filter_src.channel_layout != dec_channel_layout 
||
+is-audio_filter_src.freq   != frame-sample_rate 
||
+is-auddec.pkt_serial   != last_serial;
+
+if (reconfigure) {
+char buf1[1024], buf2[1024];
+av_get_channel_layout_string(buf1, sizeof(buf1), -1, 
is-audio_filter_src.channel_layout);
+av_get_channel_layout_string(buf2, sizeof(buf2), -1, 
dec_channel_layout);
+av_log(NULL, AV_LOG_DEBUG,
+   Audio frame changed from rate:%d ch:%d fmt:%s layout:%s 
serial:%d to rate:%d ch:%d fmt:%s layout:%s serial:%d\n,
+   is-audio_filter_src.freq, 
is-audio_filter_src.channels, av_get_sample_fmt_name(is-audio_filter_src.fmt), 
buf1, last_serial,
+   frame-sample_rate, av_frame_get_channels(frame), 
av_get_sample_fmt_name(frame-format), buf2, is-auddec.pkt_serial);
+
+is-audio_filter_src.fmt= frame-format;
+is-audio_filter_src.channels   = 
av_frame_get_channels(frame);
+is-audio_filter_src.channel_layout = dec_channel_layout;

Re: [FFmpeg-devel] FFserver bug?

2014-10-30 Thread Michael Niedermayer 
On Fri, Oct 31, 2014 at 02:14:29AM +0100, Sevan Gelici wrote:
 Hi,
 
 Two days ago i updated my linux system and also ffmpeg was updated. When i
 started to stream it was not working anymore. I am using ffserver. i
 determined the problem which causes the problem thats the bitrate it makes
 like 50mb of it. i reinstalled the system a couple times because of some
 tests and still i don't get it work. Can someone help me with this pls.

do you know which commit / revission has caused the regression ?

[...]

-- 
Michael GnuPG fingerprint: 9FF2128B147EF6730BADF133611EC787040B0FAB

DNS cache poisoning attacks, popular search engine, Google internet authority
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