Literal&&&string&&&Java Data Type Q&A1.&&&&&Why is java.lang.String the only class for which two ways of creation exist:
1) With normal way with "new" keyword.
String s = new String("abc");
With String literal (which is only available ...2.&&&&&And by string literals I mean those containing \123-like characters too.
I've written something but I don't know if it's perfect:
&STRING& {
...3.&&&&&This is sort of the Java analogue of .
Suppose I have a String object which I want to represent in code and I want to produce a ...4.&&&&&I have
if (localName.equals("TaxName")) {
but PMD says
Position literals first in String comparisons
5.&&&&&what is difference between
String str = new String("abc");
String str = "abc";
6.&&&&&i want to know where to use string object(in which scenario in my java code).
ok i understood the diff btwn string literal and string object, but i want to know ...7.&&&&&I'm processing some Java source code using Java. I'm extracting the string literals and feed them to a function taking a String. The problem is that I need to pass the ...8.&&&&&String date = new java.text.SimpleDateFormat("MM-dd-yyyy").format(new java.util.date());
upload.uploadfile("192.168.0.210", "muruganp", "vm4snk", "/home/media/Desktop/FTP Upload/+date+"_RB.zip"", "/fileserver/filesbackup/Emac/+date+"_RB.zip"");
uploadfile is a function which uploads the file 10-20-2010_RB.zip to the server location.
But since I do have the string "date" in ...9.&&&&&According to this article
I was coding some examples like :
public static void main(String[] args) {
String literalString1 = "someString";
String literalString2 = "someString";
...10.&&&&&I've come across a class that includes multiple uses of a string literal, "foo".
What I'd like to know, is what are the benefits and impact (in terms of object creation, memory ...11.&&&&&What are the mechanics behind Java automatically interpreting string literals as String objects, since there are no overloaded operators and there is no default support for low-level string buffers (not including ...12.&&&&&What is the difference between an addition of String Literal and String Object?
jdk = 1.6.22
For example
String s1 ="hello";
String s2 ="hello1";
...13.&&&&&I'm trying to use a constant instead of a string literal in this piece of code:
new InputStreamReader(new FileInputStream(file), "UTF-8")
"UTF-8" appears in the code rather often, and would be much better to ...14.&&&&&
Possible Duplicate:
Is there a library function that takes a string and converts is into ...15.&&&&&What is the type of "I like Comp Sci!"? I'm pretty sure its either a string or a literal, can anyone point out the difference between the two and help me ...16.&&&&&I have known that JVM maintains a string literal pool to increase performance and maintain JVM memory and learned that string literal is maintained in the string pool. But I want ...17.&&&&&How many String object are created
I am studying for the SCJP I cant seem to get my head round this String problem. I seem to see several possible answers depending on ...18.&&&&&So basically the user picks a category and your program picks a word from that category. Suppose those categories are numbered 0, 1, 2, ... n-1 where you have 'n' categories. ...19.&&&&&20.&&&&&1. If I print out the following unicode in a servlet-- String s = "\u1f26\u"; out.println(s); The unicode Greek characters are printed perfectly in HTML. 2. But when I get the unicode String (u1f26\u) from elsewhere, that is, I DO NOT initialize String s with the literal string as above, out.println statement produces, on the HTML page, the literal string -- ...21.&&&&&Hi, public class Test{ public static void main(String args){ String str ="abc"; String str1 = "ab"; str1 = str1.concat("c"); if(str.equals(str1){ System.out.println("In Equals method"); } if(str == str1){ System.out.println("In == method"); } } } I couldn't understand why it's printing only "In Equals method". "As per java API if compiler encounters a String literal , it checks the pool to see if ...22.&&&&&I have few basic doubts in String & its memory pool. With String s=new String("abc"); Two objects with value abc ,are created.One in prgram space referenced by s.& One goes in literal pool. 1. Is this String in literal pool remains lost or unreferenced. 2. And if program furthue uses s ,value abc comes from program space? 3. Is this literal ...23.&&&&&Thanks David for your suggestion but i feel some difficulty to implement your suggestion. In the meanwhile, I made the above program using the compareTo method using String type Arrays. But the same thing i want to do with String literal without arrays means a para of text which sort out the text in ascending order. I think this would required ...24.&&&&&25.&&&&&26.&&&&&Ernest, thanks for pointing me to the resources. But this brings another question in my mind. If for example I have a class A that invokes some methods in class B. class A{ public boolean insertDataInTable(String s){ B b = new B(); boolean result = b.insertData(s); } class B{ public boolean insertData(String s){ String sql = "insert in table"; ...27.&&&&&28.&&&&&hi.. can any body tell me what is String an object or literal ? i mean what is the difference when we declare string as String a = new String("abc text"); String b= "xyz text"; ??? and are call to functions in java are "call by refernce" or "call by value" i mean if i pass say someobject.someMethod("abc_text"); someobject.someMethod(a); // a ...29.&&&&&class Q15 { public static void main(String[] args) { System.out.print("\nab");// due to '\n' character it goes to new line and on new line prints 'ab' System.out.print("\bsi");// due to '\b' character, character 'b' will be deleted and 'si' will be appened so output become 'asi' System.out.print("\rha");// due to '\r' character, it starts printing from first character and overwrites previous output so 'ha' ...30.&&&&&31.&&&&&The soul is dyed the color of its thoughts. Think only on those things that are in line with your principles and can bear the light of day. The content of your character is your choice. Day by day, what you do is who you become. Your integrity is your destiny - it is the light that guides your way. - ...32.&&&&&Hi Can any body give me the difference between the String Literals and String Objects and how the JVM implements both. I came across the following piece of code. public class Stringcomparison { public static void main(String args[]) { String ss1=new String("Rajarsi"); String ss2=new String("Rajarsi"); String s1="Rajarsi"; String s2="Rajarsi"; System.out.println(" == comparison for StringObjects: "+(ss1==ss2)); System.out.println(" == comparison for StringLiterals: "+(s1==s2)); ...33.&&&&&Java or any compiler scans each character in the source file. When it encounters a double quote character it flags the the start of a string and when it is another double quote character it encounters the end of the string. Carefully read and run the following code for an example: public class ParseStrings { public static void main(String args[]) { ...34.&&&&&35.&&&&&A singleton is a single instance of a class per JVM per class loader. Please google for Java singleton design pattern for example. Classes are introduced into the Java environment when they are referenced by name in a class that is already running. There is a bit of magic that goes on to get the first class running (which is why ...36.&&&&&Ernest, I don't think Strings are that simple of a subject depending on how deep you want to get. When does a String come from the constant pool? Does the JVM use a StringBuilder when concatanating two Strings? Does calling the new method create a new String or not if it's already in the pool? etc... there are a lot of ...37.&&&&&Position literals first in String comparisons - that way if the String is null you won't get a NullPointerException, it'll just return false. As it is a very good approach- Position literals first in String comparisons - that way if the String is null you won't get a NullPointerException, it'll just return false. public class NullCheckEqual { public static void main(String[] ...38.&&&&&Hi, Can i know the reason for the String literal assignment . String str1 = new String("abc"); String str="abc"; System.out.println(str==str1); System.out.println("Str memory-->"+str.hashCode()); System.out.println("Str1 memeory-->"+str1.hashCode()); output: false Str memory-->96354 Str1 memeory-->96354 Even though a new String object is created both of them have same address still str1=str gives false. This is confusing . 39.&&&&&Hi, I happened to read the journal by Corey written in 2004 - /journal/200409/Journal200409.jsp#a1 Was happy to know few things... though... Why is it required for String objects to have references from two locations, when they are created as string literals? For eg... String a = "Ram"; "Ram" object is referred from the local variable table and from the string literal ...40.&&&&&The big problem with intern is once you intern a String, you are stuck with it in RAM until the program ends. It is no longer eligible for garbage collection, even if there are no more references to it. If you want a temporary interned String, you might consider interning manually. However, in the most recent JVMs, the interned string cache ...41.&&&&&42.&&&&&Hi All, I understand string concepts to some extent. But, now I have a different question. When we create a String like String a = "Sample"; The String object "Sample" is available in the heap and a reference is created in the string pool as well apart from the variable reference. Now when a code like this follows the above snippet... ...43.&&&&&1. When you need to create string as a ThreadLocal. 2. When a string is doesnt have to be living throughout the life of JVM. If you have a code that creates too many strings (in a loop for eg) and this is code is run highly infrequently in application life time, it is perhaps preferable to create strings on heap ...44.&&&&&Hello all, I'm trying to write a program that will allow users to type in a file path and return just the name of the file that they specified. I have the program written out, but every time I try to run it in Eclipse, it gives me the message, "string literal is not properly closed by a double quote." I ...45.&&&&&Both have their own trade-offs: If you create a string using String literals , the string stays in the string-pool forever. It is never garbage collected. The down side of this is that the string-pool has a fixed size and it will get full at sometime. With 'new' operator new strings objects are created out of the string-pool, and they are ...46.&&&&&47.&&&&&48.&&&&&
Sudhanshu Mishra wrote:Thanks for the suggestion,but i would appreciate if you could be more direct. Do we get a new object or not? Thanks... While the direct answer is quite simple to give, here at javaranch, every1 believes in pointing to the resource than reading it out as much as you can. It serves two purposes 1) It saves ...49.&&&&&Hello: I'm facing the following situation: 1. I've a method which, from an user input, receives many String parameters which need to be parsed to numbers. 2. Inside of it, the method which I use to parse them throws a specific kind of exception if it fails. 3. If such exception is thrown, I need to catch it and set it ...50.&&&&&The JVM maintains a pool of Strings, so that frequently used values can be reused instead of being instantiated all over again. In the following code, the two string variables refer to the same String instance: String a = "music"; String b = "music"; Strings a and b are both literals. Their values are assigned explicitly rather than through the use ...51.&&&&&java-forums.orgI'm writing a class to make a random password, but for some reason I can only print string literals in the main method. Can anyone tell me why? Java Code: public class RandomPassword { /* main method, only used for testing methods */ public static void main(String[] args) { System.out.print(RandomPassword.fixLast("this_")); } /* returns random char alpha (caps or no caps) or ...52.&&&&&53.&&&&&Hello all, Please forgive if this is dumb but I have a deadline to meet. I have a function func_name(int num, String ip_s1, String ip_s2) It works just fine if I do somaclass.func_name(123,"string1","string2"); but not for someclass.func_name(123,string_var,string_var2); even though the function "receives" the input correctly. I have no clue as to what to do. Please help ! 54.&&&&&Thanks for your answer. By knowing this, I have other questions regarding this issue. Would you be kind to answer it, thanks again! ProjectMoon wrote: When javac reads string concatenation like this, it replaces that code with calls that create a StringBuilder and use it to do the actual string concatenation. so... you meant just like what I assumed in the ...55.&&&&&I've got a little program that reads the input of a user, the input should look like "draw tree" for example. Now if the user gives "draw" as a command (the complete input is splitted on whitespace, the first wordt being the first element of an array), I'd like the program to execute a certain part of the code. However, the ...56.&&&&&Thank you for your reply. 1. String str = "Java"; 2. str = str.concat(" Beans "); 3. str = str.trim(); 4. String str1 = "abc"; 5. String str2 = new String("abc").intern(); But one of my colleague suggested that 5 java String objected will be created in the String Literal Pool "Java" // line1 " Beans " // line2 "Java Beans " ...57.&&&&&58.&&&&&59.&&&&&I understand how the string objects are created using new operator and using literal.I also understand how they are stored in memory.All I want to know is that when and why should I create a String object using new and when I should create an object using literal.Is there any specific reason for creating an object using new operator when objects ...60.&&&&&61.&&&&&I've read that if two literal Strings contain the same value, they reference the same object to save memory. Is this guaranteed? And is the same thing true of primitive wrappers with literal values like Integer(42)? The reason I ask is because I'm thinking of using a Map as a sparse array indexed by Integers... or, if not that, then Strings ...62.&&&&&Hi, I have a string which contains the name of the customer, followed by phone number. There is no specific length for the name part and the tel number( I cannot use the substring method directly). The tel number is sometimes followed by address( cannot take the last 10 chars as the tel num either). I need to parse the string ...63.&&&&&import java.util.regex.P import java.util.regex.M public class regexTest { public static void main (String []args) { Pattern p = pile("[a-zA-Z0-9- /]+"); Matcher m = p.matcher("//"); boolean b = m.matches(); String s1 = m.toString(); System.out.println(m.matches()); System.out.print(m.toString()); } } The program works fine and returns a Boolean true to the console indicating the match was successful for the following
&|&Email:info &|&& Demo Source and Support. All rights reserved.From chenct1976
1 02:26:10 2012
From: chenct1976
(chenct1976 )
Date: Thu, 01 Mar :10 +0800
Subject: [FFmpeg-user] A question: ffplay can not play from the given time
Message-ID:
Hello everybody,
I played the file Wildlife.wmv(Win7 OS Sample Videos) by using ffplay comand, but it can not play from the given seconds.
OS: Win7 32Bit + Wingw
ffplay version: V0.10
Command line:
./ffplay_g -i Wildlife.wmv -ss 2
Expected result: play the video file from the 2nd second.
Test result: play from 0 second.
Is it a ffplay's bug? or the tested file has problem?
Thanks in advance!
attach the test log:
-09:09: ffplay version 0.10
-09:09: Copyright (c)
the FFmpeg developers
-09:09: built on Jan 29 :46 with gcc 4.5.2
-09:09: configuration: --prefix=/ffmpeg/release10 --enable-debug --disable-static --enable-shared --enable-gpl --enable-version3 --enable-avfilter --enable-memalign-hack --enable-avisynth --enable-libgsm --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libtheora --enable-libvorbis --enable-libx264 --enable-libxvid --enable-libfaac --enable-nonfree --enable-libspeex --enable-libopenjpeg --enable-libxavs --enable-libvpx --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libschroedinger --enable-libcelt --enable-frei0r --enable-libdirac --disable-decoder=libdirac --extra-cflags=-I/ffmpeg/olibs9/include --extra-ldflags=-L/ffmpeg/olibs9/lib --extra-libs=/mingw/lib/libdl.a
-09:09: libavutil 51. 34.101 / 51. 34.101
-09:09: libavcodec 53. 60.100 / 53. 60.100
-09:09: libavformat 53. 31.100 / 53. 31.100
-09:09: libavdevice 53. 4.100 / 53. 4.100
-09:09: libavfilter 2. 60.100 / 2. 60.100
-09:09: libswscale 2. 1.100 / 2. 1.100
-09:09: libswresample 0. 6.100 / 0. 6.100
-09:09: libpostproc 52. 0.100 / 52. 0.100
-09:09: Input #0, asf, from 'Wildlife.wmv':
-09:09: Metadata:
-09:09: SfOriginalFPS : 299
-09:09: WMFSDKVersion : 11.0.
-09:09: WMFSDKNeeded : 0.0.0.0000
-09:09: IsVBR : 0
-09:09: title : Wildlife in HD
-09:09: copyright : ? 2008 Microsoft Corporation
-09:09: comment : Footage: Small World Productions, I Tourism New Zealand | Producer: Gary F. Spradling | Music: Steve Ball
-09:09: Duration:
-09:09: 00:00:30.09
-09:09: start:
-09:09: 0.000000
-09:09: bitrate:
-09:09: 6977 kb/s
-09:09: Stream #0:0
-09:09: (eng)
-09:09: Audio: wmav2 (a[1][0][0] / 0x0161), 44100 Hz, 2 channels, s16, 192 kb/s
-09:09: Stream #0:1
-09:09: (eng)
-09:09: Video: vc1 (Advanced) (WVC1 / 0x), yuv420p, 42 kb/s
-09:09: 29.97 tbr
-09:09: 1k tbn
-09:09: 1k tbc
From adam at
1 02:41:52 2012
From: adam at
(Adam N. Rosenberg)
Date: Wed, 29 Feb :52 -0700 (MST)
Subject: [FFmpeg-user] Vimeo uploading - may be off topic
Message-ID:
Hi Vimeo fans,
This may be off topic, but maybe somebody can take pity on me.
I have 284 aviation videos that I would like to upload to Vimeo.
(I used FFMPEG to add music soundtracks with help from this group.)
I could upload the AVI files one at a time and type my video titles
and descriptions, one at a time, or I could write a script on Linux.
I know how to write C code and stuff like that, I can cobble
together HTML and blunder my way through really-basic PHP code,
but the kind of interactive reactions that run Vimeo applications
are beyond my current knowledge.
Their documentation presumes one
already has a working knowledge of how to use Vimeo applications.
I haven't a clue, so I really need baby-steps instructions.
I downloaded their PHP file for uploading and I ran Vimeo's web
page that gave me a Consumer Key and a Consumer Secret, but I don't
know what I'm supposed to do next.
If I can get one PHP script
to upload one video with title and description, then I can write
284 scripts with a C program from my file list and submit them.
Any help would be appreciated.
Adam N. Rosenberg
mailto:adam at
http://www.
From dheianevans
1 08:05:08 2012
From: dheianevans
(Ian Evans)
Date: Thu, 1 Mar :08 -0500
Subject: [FFmpeg-user] converting 1080 to 720 mpeg2
In-Reply-To:
References:
Message-ID:
On Wed, Feb 29, 2012 at 5:18 PM, Andy Sheen
>> This is old, please try current git head.
> Teehee... I've been waiting for that.
> And change all the syntax... There should be no issues with map x:y -
> it's been there for yonks and always worked when I've needed it -
> except when I needed to demux 0:24 and it only mapped 0:20. The only
> problem was the new ffmpeg introduced a 100ms delay in the audio but
> you can w).
> Try a -map 0:0 infront of the video decode (after the -f filetype...) first.
Okay...updated to the latest from git tonight. Ran this command line:
ffmpeg -i 6204000.mpg -t 00:5:00.000 -f mpegts -map 0:0
-vcodec mpeg2video -qscale 2
-vf 'yadif=1,scale='
-map 0:1 -acodec copy -map 0:2 -acodec copy
video-01.mpeg.ts
Here's the info of the test file it generated:
ffmpeg version N-38403-gd2101bf Copyright (c)
the FFmpeg developers
built on Feb 29 :15 with gcc 4.4.5
configuration: --enable-gpl --enable-libfaac --enable-libmp3lame
--enable-libopencore-amrnb --enable-libopencore-amrwb
--enable-libtheora --enable-libvorbis --enable-libx264
--enable-nonfree --enable-version3 --enable-x11grab
51. 41.100 / 51. 41.100
libavcodec
4.100 / 54.
libavformat
2.100 / 54.
libavdevice
4.100 / 53.
libavfilter
2. 62.101 /
libswscale
libswresample
libpostproc
0.100 / 52.
Input #0, mpegts, from 'video-01.mpeg.ts':
Duration: 00:04:59.98, start: 1.400000, bitrate: 24036 kb/s
service_name
: Service01
service_provider: FFmpeg
Stream #0:0[0x100]: Video: mpeg2video (Main) ([2][0][0][0] / 0x0002),
[SAR 1:1 DAR 16:9], 104857 kb/s, 59.96 fps, 59.94
tbr, 90k tbn, 119.88 tbc
Stream #0:1[0x101]: Audio: ac3 ([129][0][0][0] / 0x0081), 48000 Hz,
5.1(side), s16, 384 kb/s
Stream #0:2[0x102]: Audio: ac3 ([129][0][0][0] / 0x0081), 48000 Hz,
stereo, s16, 128 kb/s
So am I ready to cat it to the 720 file? What step occurs after that?
Thanks to everyone for the help.
From dave.bevan at bbc.co.uk
1 09:10:13 2012
From: dave.bevan at bbc.co.uk (Dave Bevan)
Date: Thu, 1 Mar :13 -0000
Subject: [FFmpeg-user] Vimeo uploading - may be off topic
In-Reply-To:
References:
Message-ID:
> Subject: [FFmpeg-user] Vimeo uploading - may be off topic
Any help would be appreciated.
Adam N. Rosenberg
I reckon /api/docs/upload has everything you need to
http://www.bbc.co.uk/
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From chenct1976
1 09:33:04 2012
From: chenct1976
(chenct1976 )
Date: Thu, 01 Mar :04 +0800
Subject: [FFmpeg-user] Will you pls give me some advice? // A question:
ffplay can not play from the given time
Message-ID:
Hello everybody,
I played the file Wildlife.wmv(Win7 OS Sample Videos) by using ffplay comand, but it can not play from the given seconds.
OS: Win7 32Bit + Wingw
ffplay version: V0.10
Command line: ./ffplay_g -i Wildlife.wmv -ss 2
Expected result: play the video file from the 2nd second.
Test result: play from 0 second.
Is it a ffplay's bug? or the tested file has problem?
Thanks in advance!
attach the test log:
-09:09: ffplay version 0.10
-09:09: Copyright (c)
the FFmpeg developers
-09:09: built on Jan 29 :46 with gcc 4.5.2
-09:09: configuration: --prefix=/ffmpeg/release10 --enable-debug --disable-static --enable-shared --enable-gpl --enable-version3 --enable-avfilter --enable-memalign-hack --enable-avisynth --enable-libgsm --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libtheora --enable-libvorbis --enable-libx264 --enable-libxvid --enable-libfaac --enable-nonfree --enable-libspeex --enable-libopenjpeg --enable-libxavs --enable-libvpx --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libschroedinger --enable-libcelt --enable-frei0r --enable-libdirac --disable-decoder=libdirac --extra-cflags=-I/ffmpeg/olibs9/include --extra-ldflags=-L/ffmpeg/olibs9/lib --extra-libs=/mingw/lib/libdl.a
-09:09: libavutil 51. 34.101 / 51. 34.101
-09:09: libavcodec 53. 60.100 / 53. 60.100
-09:09: libavformat 53. 31.100 / 53. 31.100
-09:09: libavdevice 53. 4.100 / 53. 4.100
-09:09: libavfilter 2. 60.100 / 2. 60.100
-09:09: libswscale 2. 1.100 / 2. 1.100
-09:09: libswresample 0. 6.100 / 0. 6.100
-09:09: libpostproc 52. 0.100 / 52. 0.100
-09:09: Input #0, asf, from 'Wildlife.wmv':
-09:09: Metadata:
-09:09: SfOriginalFPS : 299
-09:09: WMFSDKVersion : 11.0.
-09:09: WMFSDKNeeded : 0.0.0.0000
-09:09: IsVBR : 0
-09:09: title : Wildlife in HD
-09:09: copyright : ? 2008 Microsoft Corporation
-09:09: comment : Footage: Small World Productions, I Tourism New Zealand | Producer: Gary F. Spradling | Music: Steve Ball
-09:09: Duration:
-09:09: 00:00:30.09
-09:09: start:
-09:09: 0.000000
-09:09: bitrate:
-09:09: 6977 kb/s
-09:09: Stream #0:0
-09:09: (eng)
-09:09: Audio: wmav2 (a[1][0][0] / 0x0161), 44100 Hz, 2 channels, s16, 192 kb/s
-09:09: Stream #0:1
-09:09: (eng)
-09:09: Video: vc1 (Advanced) (WVC1 / 0x), yuv420p, 42 kb/s
-09:09: 29.97 tbr
-09:09: 1k tbn
-09:09: 1k tbc
From pavel at sokolov.me
1 10:12:51 2012
From: pavel at sokolov.me (Pavel Sokolov)
Date: Thu, 01 Mar :51 +0400
Subject: [FFmpeg-user] AVI/AC3 -> VOB/LPCM
In-Reply-To:
References:
Message-ID:
29.02., Andrey Utkin ?????:
Pavel Sokolov:
>> Hi all!
>> How can I do remux of the AVI with AC3 stereo to the VOB/LPCM
>> When I tried the next cmd I hear bad audio (sh-sh-sh-sh):
>> ffmpeg -i video-mpeg4_720x544-audio_ac3_48000_stereo.avi -vcodec copy
>> -acodec pcm_s16be -sample_fmt s16 -f vob test1.vob
>> What I'm doing wrong?
> Full uncut console output is missing...
> Also publish your source file, can help.
> Sound is bad on PC also, or just on your specific hardware?
file: http://sokolov.me/tmp/video-mpeg4_720x544-audio_ac3_48000_stereo.avi
ffmpeg log:
[vob @ 0xdf5be0] buffer underflow i=1 bufi=4026 size=6144
Last message repeated 34 times
[vob @ 0xdf5be0] packet too large, ignoring buffer limits to mux it
[vob @ 0xdf5be0] buffer underflow i=1 bufi=4026 size=6144
[vob @ 0xdf5be0] buffer underflow i=1 bufi=6043 size=6144
[vob @ 0xdf5be0] packet too large, ignoring buffer limits to mux it
[vob @ 0xdf5be0] buffer underflow i=1 bufi=6043 size=6144
[vob @ 0xdf5be0] buffer underflow i=1 bufi=3928 size=6144
Last message repeated 3 times
[vob @ 0xdf5be0] packet too large, ignoring buffer limits to mux it
[vob @ 0xdf5be0] buffer underflow i=1 bufi=3928 size=6144
[vob @ 0xdf5be0] buffer underflow i=1 bufi=5945 size=6144
[vob @ 0xdf5be0] packet too large, ignoring buffer limits to mux it
[vob @ 0xdf5be0] buffer underflow i=1 bufi=5945 size=6144
[vob @ 0xdf5be0] buffer underflow i=1 bufi=3830 size=6144
[vob @ 0xdf5be0] packet too large, ignoring buffer limits to mux it
[vob @ 0xdf5be0] buffer underflow i=1 bufi=3830 size=6144
[vob @ 0xdf5be0] buffer underflow i=1 bufi=5847 size=6144
[vob @ 0xdf5be0] packet too large, ignoring buffer limits to mux it
[vob @ 0xdf5be0] buffer underflow i=1 bufi=5847 size=6144
[vob @ 0xdf5be0] buffer underflow i=1 bufi=3732 size=6144
[vob @ 0xdf5be0] packet too large, ignoring buffer limits to mux it
[vob @ 0xdf5be0] buffer underflow i=1 bufi=3732 size=6144
[vob @ 0xdf5be0] buffer underflow i=1 bufi=5749 size=6144
[vob @ 0xdf5be0] packet too large, ignoring buffer limits to mux it
From andrey.krieger.utkin
1 10:54:58 2012
From: andrey.krieger.utkin
(Andrey Utkin)
Date: Thu, 1 Mar :58 +0200
Subject: [FFmpeg-user] Will you pls give me some advice? // A question:
ffplay can not play from the given time
In-Reply-To:
References:
Message-ID:
> Hello everybody,
> I played the file Wildlife.wmv(Win7 OS Sample Videos) by using ffplay comand, but it can not play from the given seconds.
> OS: Win7 32Bit + Wingw
> ffplay version: V0.10
> Command line: ./ffplay_g -i Wildlife.wmv -ss 2
> Expected result: play the video file from the 2nd second.
> Test result: play from 0 second.
As far as i know, you can seek (jump in time) only to keyframe
location. Keyframes can be located with interval of several seconds.
Thus you request to start from 2 seconds offset, but ffmpeg plays from
closest keyframe that is at 0 seconds offset.
I think for now you can pass your file through ffmpeg, to cut off
unneeded beginning of file.
Or write application using ffmpeg API that will skip video not by
seeking, but by decoding every frame and not displaying unneeded ones.
Andrey Utkin
From ssmanian
1 03:26:46 2012
From: ssmanian
Date: Wed, 29 Feb :46 -0800 (PST)
Subject: [FFmpeg-user] Unable to play VOB file with audio.
Message-ID:
I have a vob file that plays just fine but no audio as my DLNA TV does not
support the DTS format
Input #0, mpeg, from 'test.VOB':
Duration: 00:18:50.66, start: , bitrate: 7595 kb/s
Stream #0:0[0x1e0]: Video: mpeg2video (Main), yuv420p, 720x576 [SAR
64:45 DAR 16:9], 9000 kb/s, 27.21 fps, 25 tbr, 90k tbn, 50 tbc
Stream #0:1[0x89]: Audio: dts (DTS), 48000 Hz, 5.1(side), s16, 768 kb/s
Stream #0:2[0x80]: Audio: ac3, 48000 Hz, 5.1(side), s16, 448 kb/s
At least one output file must be specified
As you can see, the first stream is DTS which is not supported but the
second stream is supported. So what I want to do is create another VOB file
and use the video as is and the second stream from the audio. So I did,
/volume1/@appstore/Serviio/bin/ffmpeg -i test.VOB -i test.VOB -map 0:0 -map
0:2 -vcode
c copy -acodec copy -y test-copyav.VOB
The resulting file now does not play video but the audio is fine. (Reverse
of the original issue). The output of the new file is:
Seems stream 1 codec frame rate differs from container frame rate: 50.00
(50/1) -> 50.00 (50/1)
Input #0, mpeg, from 'test-copyav.VOB':
Duration: 00:18:51.14, start: 1.042667, bitrate: 6780 kb/s
Stream #0:0[0x80]: Audio: ac3, 48000 Hz, 5.1(side), s16, 448 kb/s
Stream #0:1[0x1e0]: Video: mpeg2video (Main), yuv420p, 720x576 [SAR
64:45 DAR 16:9], 9000 kb/s, 27.14 fps, *50 tbr,* 90k tbn, 50 tbc
At least one output file must be specified
The only difference i see is the TBR is 50 in the new file and 25 on the
original. Is that the reason why it is not playing video? If so, is there an
option to force the tbr value. Am I doing something wrong in the above step?
Thanks for any help.
View this message in context: http://ffmpeg-users.933282./Unable-to-play-VOB-file-with-audio-tp3823.html
Sent from the FFmpeg-users mailing list archive .
From wodfer
1 12:51:58 2012
From: wodfer
(Andy Wodfer)
Date: Thu, 1 Mar :58 +0100
Subject: [FFmpeg-user] Problems encoding to DV25 .avi file
Message-ID:
I need some input on a problem I have.
I'm using the latest 0.10 version of ffmpeg compiled from source on a
FreeBSD 8.2 machine.
The source file is a Prores 422HQ file ( @ 50fps 720P).
I want to make this into a DV25 .avi file that is playable in Windows Media
Player and that will work in Ie. Adobe Premiere Pro 2.0 on a WinXP machine.
Ie: A Windows compatible DV25 file.
I've tried several options, but I can't make it work.
"-target pal-dv" gives me a file that plays fine in VLC player, but not in
Windows Media Player. Adobe Premiere reports unsupported format and won't
import it.
"-vcodec dvvideo
-s 720x576 -pix_fmt yuv420p -r 25 -acodec pcm_s16le -ac 2
-ar 48000" works if I choose .mov as a container. It opens in Quicktime and
VLC and import to Adobe Premiere works fine. Aspect ratio is also fine.
If I change the container to .avi it won't play back in Windows Media
Any ideas of what I can try next?
Best regards,
From de.techno
1 13:50:03 2012
From: de.techno
Date: Thu, 01 Mar :03 +0530
Subject: [FFmpeg-user] Need help with ffmpeg
In-Reply-To:
References:
Message-ID:
On 02/29/12 21:01, Carl Eugen Hoyos wrote:
> Aashish vivekanand<aashishvanand
>> link: /?3ri05yd2wej5wrm
> Which program plays this file?
> Carl Eugen
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
File command says that it's data....
From sheen.andy
1 14:40:58 2012
From: sheen.andy
(Andy Sheen)
Date: Thu, 1 Mar :58 +0000
Subject: [FFmpeg-user] converting 1080 to 720 mpeg2
In-Reply-To:
References:
Message-ID:
On 1 March , Ian Evans
> Okay...updated to the latest from git tonight. Ran this command line:
> So am I ready to cat it to the 720 file? What step occurs after that?
Have you tried it? If it works - you should just be able to play it. I'm
wondering about how the transition will be handled. The TS file format will
use PIDs and these will change when you change streams.
It may be better to remux (after concatenation) to something like a .mkv
From bostjan.strojan
1 15:26:58 2012
From: bostjan.strojan
(=?UTF-8?Q?Bo=C5=A1tjan_Strojan?=)
Date: Thu, 1 Mar :58 +0100
Subject: [FFmpeg-user] Scene detection
In-Reply-To:
References:
Message-ID:
On Wed, Feb 29, 2012 at 4:56 PM, atigian
> When I don't have an EDL, my approach is to do the scene detection during
> the encoding using keyint, min-keyint and scenecut, thus inserting a key
> frame in every scene change. Once I've encoded the file I just extract all
> key frames:
> ffmpeg -vf select="eq(pict_type\,PICT_TYPE_I)" -i myvideo.mp4 -vsync 2 -s
> 73x41 -f image2 thumbnails-%02d.jpeg
> I also create a text file with the timecodes of the key frames by adding:
> -loglevel debug 2>&1 | grep "pict_type:I -> select:1" | cut -d " " -f 6 - >
> keyframe-timecodes.txt
> And I use this file to sync the thumbnails of the scene changes with a video
> /blog/scene-change-detection-during-encoding-key-frame-extraction-code
> /blog/scene-change-detection-during-encoding-key-frame-extraction-code
Thanks, much appreciated.
From adam at
1 15:41:04 2012
From: adam at
(Adam N. Rosenberg)
Date: Thu, 1 Mar :04 -0700 (MST)
Subject: [FFmpeg-user] Vimeo uploading - may be off topic
In-Reply-To:
References:
Message-ID:
On Thu, 1 Mar 2012, Dave Bevan wrote:
>> Subject: [FFmpeg-user] Vimeo uploading - may be off topic
Any help would be appreciated.
Adam N. Rosenberg
> I reckon /api/docs/upload has everything you need to
I reckon it does have everything, except the basic stuff of how
to setup and to run the programs.
They figure you already know that.
I'm looking for real baby steps for a first-time Vimeo programmer.
Go the /setup-auth
Type your username, password, and description of the application
Press the GET KEY button.
Write down code and secret code.
Download PHP file from /get-upload-code
Add the following lines to the bottom
Once I can do that for one video, then, yes, I can figure out how
to do the other 283 of them.
Adam N. Rosenberg
mailto:adam at
http://www.
From de.techno
1 16:12:51 2012
From: de.techno
Date: Thu, 01 Mar :51 +0530
Subject: [FFmpeg-user] Unable to play VOB file with audio.
In-Reply-To:
References:
Message-ID:
On 03/01/12 07:56, Senthil wrote:
Stream #0:0[0x80]: Audio: ac3, 48000 Hz, 5.1(side), s16, 448 kb/s
Stream #0:1[0x1e0]: Video: mpeg2video (Main), yuv420p, 720x576 [SAR
Usually the first stream (0.0) is Video. Try moving it to the first
stream and make audio the second.
From tevans.uk
1 16:25:51 2012
From: tevans.uk
(Tom Evans)
Date: Thu, 1 Mar :51 +0000
Subject: [FFmpeg-user] Vimeo uploading - may be off topic
In-Reply-To:
References:
Message-ID:
On Thu, Mar 1, 2012 at 2:41 PM, Adam N. Rosenberg
> Hi Dave,
> ? ?I reckon it does have everything, except the basic stuff of how
> to setup and to run the programs. ?They figure you already know that.
> I'm looking for real baby steps for a first-time Vimeo programmer.
Might I humbly suggest if you are looking for vimeo support, the
ffmpeg users mailing list is not the best place to look. If you were
looking for baby steps for a first time ffmpeg programmer, you would
have been exactly right...
From longxd
1 16:59:36 2012
From: longxd
Date: Thu, 1 Mar :36 -0500
Subject: [FFmpeg-user] Where to find the version ffmpeg version
Message-ID:
I run into a problem of frame alignment between different version of ffmpeg.
I have a video decoded by someone else by the above version with some
information generated after processing the image. Now I also decoded
the video and processed the image and another type of information
generated. I want to marry the two types of information but first I
need to find the correct frame correspondence, which I find a
difficult task.
I am using the newest version of ffmpeg on windows, and the old
version someone else used is attached below, where can I download the
exact version of the software for: ffmpeg version N-3e9ca?
-------------------------------------
ffmpeg version N-3e9ca, Copyright (c)
the FFmpeg developers
built on Aug
6 :11 with gcc 4.6.1
configuration: --enable-gpl --enable-version3 --enable-memalign-hack
--enable-runtime-cpudetect --enable-avisynth --enable-bzlib
--enable-frei0r --enable-libopencore-amrnb --enable-libopencore-amrwb
--enable-libfreetype --enable-libgsm --enable-libmp3lame
--enable-libopenjpeg --enable-librtmp --enable-libschroedinger
--enable-libspeex --enable-libtheora --enable-libvorbis
--enable-libvpx --enable-libx264 --enable-libxavs --enable-libxvid
--enable-zlib
51. 11. 1 / 51. 11. 1
libavcodec
9. 1 / 53.
libavformat
6. 0 / 53.
libavdevice
2. 0 / 53.
libavfilter
2. 28. 0 /
libswscale
libpostproc
2. 0 / 51.
[wmv3 @ 01DDAB00] Extra data: 8 bits left, value: 0
Input #0, asf, from ?09068.wmv':
WMFSDKVersion
WMFSDKNeeded
: 0.0.0.0000
Buffer Average
Duration: 00:25:57.94, start: 0.000000, bitrate: 38455 kb/s
Stream #0.0(eng): Video: wmv3 (Main), yuv420p, , 49480
kb/s, 1k tbr, 1k tbn, 1k tbc
At least one output file must be specified
From nicolas.george at normalesup.org
1 17:13:07 2012
From: nicolas.george at normalesup.org (Nicolas George)
Date: Thu, 1 Mar :07 +0100
Subject: [FFmpeg-user] Where to find the version ffmpeg version
In-Reply-To:
References:
Message-ID:
Le duodi 12 vent?se, an CCXX, X. Long a ?crit?:
> I am using the newest version of ffmpeg on windows, and the old
> version someone else used is attached below, where can I download the
> exact version of the software for: ffmpeg version N-3e9ca?
The strange string at the end is the Git short hash, prefixed by "g", it
identifies the version exactly. If you have a complete git clone of the
source tree, the following command will put the tree at that version:
git checkout 6c4e9ca
(for references, it dates back to Sun Aug 7 00:32:11 )
Alternatively, you can browse the gitweb repository on . I do not know how to look for a particular
commit, though, but putting "h=6c4e9ca" in the URL seems to do the trick.
Then you can ask for a snapshot.
Nicolas George
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From ffmpeg-user
1 20:07:09 2012
From: ffmpeg-user
(Brian Johnson)
Date: Thu, 1 Mar :09 -0800
Subject: [FFmpeg-user] Problem "live" transcoding via piped input / Ceton
InfiniTV 4
Message-ID:
The command I am using: ffmpeg -i /dev/ctn91xx_mpeg0_0 -async 2
-acodec libmp3lame -ac 2 -vcodec libx264 -s
-aspect 16:9 -f
mpegts file.ts
The file /dev/ctn91xx_mpeg0_0 is actually the device (Ceton InfiniTV
4) itself where it returns a mpeg transport stream, basically a named
pipe containing the raw audio and video.
When I run the command below, I get tons of errors which repeats forever:
ES packet size mismatch.0 size=???? 797kB time=00:00:04.23
bitrate=1541.1kbits/s dup=25 drop=0
[ac3 @ 0x8393340] frame CRC mismatch
[mpeg2video @ 0x834e680] ac-tex damaged at 0 28
[mpeg2video @ 0x834e680] ac-tex damaged at 2 1
[mpeg2video @ 0x834e680] ac-tex damaged at 0 27
[mpeg2video @ 0x834e680] ac-tex damaged at 0 29
[mpeg2video @ 0x834e680] Warning MVs not available
[mpeg2video @ 0x834e680] concealing 1276 DC, 1276 AC, 1276 MV errors
[mpegts @ 0x8287aa0] PES packet size mismatch
frame sync error 34 q=29.0 size=??? 1202kB time=00:00:05.76
bitrate=1709.0kbits/s dup=145 drop=0
Error while decoding stream #0:27
ac-tex damaged at 4 2929.0 size=??? 1383kB time=00:00:09.17
bitrate=1235.1kbits/s dup=145 drop=0
[mpeg2video @ 0x834e680] mb incr damaged
[mpeg2video @ 0x834e680] ac-tex damaged at 35 29
[mpeg2video @ 0x834e680] concealing 880 DC, 880 AC, 880 MV errors
[mpegts @ 0x8287aa0] PES packet size mismatch
[ac3 @ 0x8393340] frame CRC mismatch
[mpegts @ 0x8287aa0] PES packet size mismatch
??? Last message repeated 1 times
[ac3 @ 0x8393340] frame sync error
Error while decoding stream #0:27
[mpeg2video @ 0x834e680] invalid cbp at 0 12
[mpeg2video @ 0x834e680] mb incr damaged
[mpeg2video @ 0x834e680] invalid mb type in B Frame at 3 24
[mpeg2video @ 0x834e680] 00 motion_type at 1 25
[mpeg2video @ 0x834e680] 00 motion_type at 0 26
[mpeg2video @ 0x834e680] invalid mb type in B Frame at 16 27
What is very interesting is that lower quality video transcodes (i.e.
320x240) has the same errors as above but only lasts about 1-2
seconds, and then output looks normal
frame= 1235 fps= 35 q=29.0 size=
1985kB time=00:00:39.53 bitrate=
411.2kbits/s dup=131 drop=0
I believe this to be related to
and in my own research, I found that this may have been a regression
from an older version, but again that is pure speculation on my part.
Any ideas? :)
From longxd
1 20:51:27 2012
From: longxd
Date: Thu, 1 Mar :27 -0500
Subject: [FFmpeg-user] Where to find the version ffmpeg version
In-Reply-To:
References:
Message-ID:
Thanks for the reply. It is very helpful. I am able to find the source
code.. But I want to run it on windows, is there a built based on this
version of the source code, or is it a easy way to build this so that
the software can run on windows.
Nicolas George :
> Le duodi 12 vent?se, an CCXX, X. Long a ?crit?:
>> I am using the newest version of ffmpeg on windows, and the old
>> version someone else used is attached below, where can I download the
>> exact version of the software for: ffmpeg version N-3e9ca?
> The strange string at the end is the Git short hash, prefixed by "g", it
> identifies the version exactly. If you have a complete git clone of the
> source tree, the following command will put the tree at that version:
> git checkout 6c4e9ca
> (for references, it dates back to Sun Aug 7 00:32:11 )
> Alternatively, you can browse the gitweb repository on
http://source.ffmpeg.org/ >. I do not know how to look for a particular
> commit, though, but putting "h=6c4e9ca" in the URL seems to do the trick.
> Then you can ask for a snapshot.
> Regards,
> ?Nicolas George
> -----BEGIN PGP SIGNATURE-----
> Version: GnuPG v1.4.11 (GNU/Linux)
> iEYEARECAAYFAk9PoBMACgkQsGPZlzblTJOVoQCdEvFQGnTAYVoBPOpA9iuSR9BP
> ZvUAni2ISZ/HKeP2Ioc5TH93NyqRT40a
> -----END PGP SIGNATURE-----
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
From wodfer
1 23:36:54 2012
From: wodfer
(Andy Wodfer)
Date: Thu, 1 Mar :54 +0100
Subject: [FFmpeg-user] Problems encoding to DV25 .avi file
In-Reply-To:
References:
Message-ID:
On Thu, Mar 1, 2012 at 12:51 PM, Andy Wodfer
> I need some input on a problem I have.
> I'm using the latest 0.10 version of ffmpeg compiled from source on a
> FreeBSD 8.2 machine.
> The source file is a Prores 422HQ file ( @ 50fps 720P).
> I want to make this into a DV25 .avi file that is playable in Windows
> Media Player and that will work in Ie. Adobe Premiere Pro 2.0 on a WinXP
> machine. Ie: A Windows compatible DV25 file.
> I've tried several options, but I can't make it work.
> "-target pal-dv" gives me a file that plays fine in VLC player, but not in
> Windows Media Player. Adobe Premiere reports unsupported format and won't
> import it.
> "-vcodec dvvideo
-s 720x576 -pix_fmt yuv420p -r 25 -acodec pcm_s16le -ac
> 2 -ar 48000" works if I choose .mov as a container. It opens in Quicktime
> and VLC and import to Adobe Premiere works fine. Aspect ratio is also fine.
> If I change the container to .avi it won't play back in Windows Media
> Any ideas of what I can try next?
> Thanks!!
> Best regards,
I seem to be making some progress with these parameters:
-vcodec dvvideo -s 720x576 -vtag dvsd -pix_fmt yuv420p -r 25 -f avi -acodec
pcm_s16le -ac 2 -ar 4
However, when playing back in WMP (which finally works) the 16x9 image is
squeezed into a 4x3 frame.
I need to make my parameters flexible so they can take either 4x3 or 16x9
and keep the aspect ratio on the output files, even if it's 16x9 HD
downscaled to PAL DV.
Any ideas?
From batguano999
2 00:01:38 2012
From: batguano999
(bat guano)
Date: Thu, 1 Mar :38 +0000
Subject: [FFmpeg-user] Where to find the version ffmpeg version
In-Reply-To:
References: ,
Message-ID:
----------------------------------------
.. But I want to run it on windows, is there a built based on this
> version of the source code
Windows build ffmpeg-git-6c4e9ca-win32-static.7z? 06-Aug-2011
It's available from here ---> /builds/win32/static/?
From renlifeng
2 06:23:56 2012
From: renlifeng
(Lifeng Ren)
Date: Fri, 2 Mar :56 +0800
Subject: [FFmpeg-user] [ffplay] after changing to another audio stream,
audio will become out of sync with video
Message-ID:
I want to report that after changing to another audio stream, audio and
video becomes out of sync.
steps to reproduce:
$ ffplay video-with-2-audio-stream.mpg
a (press a to toggle audio stream)
I also find this can be avoided by patching ffplay.c as such.
diff --git a/ffplay.c b/ffplay.c
index 93097e1..ac31ce6 100644
--- a/ffplay.c
+++ b/ffplay.c
@@ -25,7 @@ static void stream_cycle_channel(VideoState *is, int
codec_type)
AVFormatContext *ic = is->
int start_index, stream_
AVStream *
if (codec_type == AVMEDIA_TYPE_VIDEO)
start_index = is->video_
@@ -66,10 @@ static void stream_cycle_channel(VideoState *is, int
codec_type)
pos = get_master_clock(is);
stream_component_close(is, start_index);
stream_component_open(is, stream_index);
stream_seek(is, (int64_t)(pos * AV_TIME_BASE), 0, 0);
From nichot20
2 09:11:05 2012
From: nichot20
(Tim Nicholson)
Date: Fri, 02 Mar :05 +0000
Subject: [FFmpeg-user] Problems encoding to DV25 .avi file
In-Reply-To:
References:
Message-ID:
On 01/03/12 22:36, Andy Wodfer wrote:
> On Thu, Mar 1, 2012 at 12:51 PM, Andy Wodfer
>> I need some input on a problem I have.
>> I'm using the latest 0.10 version of ffmpeg compiled from source on a
>> FreeBSD 8.2 machine.
0.10 is not the latest, use git HEAD if you want that.
>> The source file is a Prores 422HQ file ( @ 50fps 720P).
>> I want to make this into a DV25 .avi file that is playable in Windows
>> Media Player and that will work in Ie. Adobe Premiere Pro 2.0 on a WinXP
>> machine. Ie: A Windows compatible DV25 file.
>> I've tried several options, but I can't make it work.
>> "-target pal-dv" gives me a file that plays fine in VLC player, but not in
>> Windows Media Player. Adobe Premiere reports unsupported format and won't
>> import it.
>> "-vcodec dvvideo
-s 720x576 -pix_fmt yuv420p -r 25 -acodec pcm_s16le -ac
>> 2 -ar 48000" works if I choose .mov as a container. It opens in Quicktime
>> and VLC and import to Adobe Premiere works fine. Aspect ratio is also fine.
>> If I change the container to .avi it won't play back in Windows Media
>> Player.
> I seem to be making some progress with these parameters:
> -vcodec dvvideo -s 720x576 -vtag dvsd -pix_fmt yuv420p -r 25 -f avi -acodec
> pcm_s16le -ac 2 -ar 4
> However, when playing back in WMP (which finally works) the 16x9 image is
> squeezed into a 4x3 frame.
> I need to make my parameters flexible so they can take either 4x3 or 16x9
> and keep the aspect ratio on the output files, even if it's 16x9 HD
> downscaled to PAL DV.
Two thoughts.
You may need to use the setdar filter to force the display aspect ratio.
I'm not sure if WMP recognises this setting in AVI's. Premi?re Pro seems to be
OK with it.
From de.techno
2 10:19:59 2012
From: de.techno
Date: Fri, 02 Mar :59 +0530
Subject: [FFmpeg-user] Problem "live" transcoding via piped input /
Ceton InfiniTV 4
In-Reply-To:
References:
Message-ID:
On 03/02/12 00:37, Brian Johnson wrote:
> The command I am using: ffmpeg -i /dev/ctn91xx_mpeg0_0 -async 2
> -acodec libmp3lame -ac 2 -vcodec libx264 -s
-aspect 16:9 -f
> mpegts file.ts
> The file /dev/ctn91xx_mpeg0_0 is actually the device (Ceton InfiniTV
> 4) itself where it returns a mpeg transport stream, basically a named
> pipe containing the raw audio and video.
> When I run the command below, I get tons of errors which repeats forever:
> ES packet size mismatch.0 size=
797kB time=00:00:04.23
> bitrate=1541.1kbits/s dup=25 drop=0
> [ac3 @ 0x8393340] frame CRC mismatch
> [mpeg2video @ 0x834e680] ac-tex damaged at 0 28
> [mpeg2video @ 0x834e680] ac-tex damaged at 2 1
> [mpeg2video @ 0x834e680] ac-tex damaged at 0 27
> [mpeg2video @ 0x834e680] ac-tex damaged at 0 29
> [mpeg2video @ 0x834e680] Warning MVs not available
> [mpeg2video @ 0x834e680] concealing 1276 DC, 1276 AC, 1276 MV errors
> [mpegts @ 0x8287aa0] PES packet size mismatch
> frame sync error 34 q=29.0 size=
1202kB time=00:00:05.76
> bitrate=1709.0kbits/s dup=145 drop=0
> Error while decoding stream #0:27
> ac-tex damaged at 4 2929.0 size=
1383kB time=00:00:09.17
> bitrate=1235.1kbits/s dup=145 drop=0
> [mpeg2video @ 0x834e680] mb incr damaged
> [mpeg2video @ 0x834e680] ac-tex damaged at 35 29
> [mpeg2video @ 0x834e680] concealing 880 DC, 880 AC, 880 MV errors
> [mpegts @ 0x8287aa0] PES packet size mismatch
> [ac3 @ 0x8393340] frame CRC mismatch
> [mpegts @ 0x8287aa0] PES packet size mismatch
Last message repeated 1 times
> [ac3 @ 0x8393340] frame sync error
> Error while decoding stream #0:27
> [mpeg2video @ 0x834e680] invalid cbp at 0 12
> [mpeg2video @ 0x834e680] mb incr damaged
> [mpeg2video @ 0x834e680] invalid mb type in B Frame at 3 24
> [mpeg2video @ 0x834e680] 00 motion_type at 1 25
> [mpeg2video @ 0x834e680] 00 motion_type at 0 26
> [mpeg2video @ 0x834e680] invalid mb type in B Frame at 16 27
> What is very interesting is that lower quality video transcodes (i.e.
> 320x240) has the same errors as above but only lasts about 1-2
> seconds, and then output looks normal
> frame= 1235 fps= 35 q=29.0 size=
1985kB time=00:00:39.53 bitrate=
> 411.2kbits/s dup=131 drop=0
> I believe this to be related to
> and in my own research, I found that this may have been a regression
> from an older version, but again that is pure speculation on my part.
> Any ideas? :)
> _______________________________________________
> ffmpeg-user mailing list
> ffmpeg-user at ffmpeg.org
> http://ffmpeg.org/mailman/listinfo/ffmpeg-user
Are you using the latest? This may be cause your PC is not as fast. Try
using -threads and -preset ultrafast.
From pavel at sokolov.me
2 12:11:53 2012
From: pavel at sokolov.me (Pavel Sokolov)
Date: Fri, 02 Mar :53 +0400
Subject: [FFmpeg-user] AVI/AC3 -> VOB/LPCM
In-Reply-To:
References:
Message-ID:
01.03., Pavel Sokolov ?????:
> 29.02., Andrey Utkin ?????:
Pavel Sokolov:
>>> Hi all!
>>> How can I do remux of the AVI with AC3 stereo to the VOB/LPCM
>>> When I tried the next cmd I hear bad audio (sh-sh-sh-sh):
>>> ffmpeg -i video-mpeg4_720x544-audio_ac3_48000_stereo.avi -vcodec copy
>>> -acodec pcm_s16be -sample_fmt s16 -f vob test1.vob
>>> What I'm doing wrong?
>> Full uncut console output is missing...
>> Also publish your source file, can help.
>> Sound is bad on PC also, or just on your specific hardware?
> http://sokolov.me/tmp/video-mpeg4_720x544-audio_ac3_48000_stereo.avi
> ffmpeg log:
> [vob @ 0xdf5be0] buffer underflow i=1 bufi=4026 size=6144
> Last message repeated 34 times
> [vob @ 0xdf5be0] packet too large, ignoring buffer limits to mux it
> [vob @ 0xdf5be0] buffer underflow i=1 bufi=4026 size=6144
> [vob @ 0xdf5be0] buffer underflow i=1 bufi=6043 size=6144
Is somebody know how to solve this issue?
With best regards, Pavel A. Sokolov
mobile: +7(921)419-1819
skype: pavel_a_sokolov
From wodfer
2 12:14:39 2012
From: wodfer
(Andy Wodfer)
Date: Fri, 2 Mar :39 +0100
Subject: [FFmpeg-user] Problems encoding to DV25 .avi file
In-Reply-To:
References:
Message-ID:
On Fri, Mar 2, 2012 at 9:11 AM, Tim Nicholson
> On 01/03/12 22:36, Andy Wodfer wrote:
> > On Thu, Mar 1, 2012 at 12:51 PM, Andy Wodfer
> >> I need some input on a problem I have.
> >> I'm using the latest 0.10 version of ffmpeg compiled from source on a
> >> FreeBSD 8.2 machine.
> 0.10 is not the latest, use git HEAD if you want that.
> >> The source file is a Prores 422HQ file ( @ 50fps 720P).
> >> I want to make this into a DV25 .avi file that is playable in Windows
> >> Media Player and that will work in Ie. Adobe Premiere Pro 2.0 on a WinXP
> >> machine. Ie: A Windows compatible DV25 file.
> >> I've tried several options, but I can't make it work.
> >> "-target pal-dv" gives me a file that plays fine in VLC player, but not
> >> Windows Media Player. Adobe Premiere reports unsupported format and
> >> import it.
> >> "-vcodec dvvideo
-s 720x576 -pix_fmt yuv420p -r 25 -acodec pcm_s16le
> >> 2 -ar 48000" works if I choose .mov as a container. It opens in
> Quicktime
> >> and VLC and import to Adobe Premiere works fine. Aspect ratio is also
> >> If I change the container to .avi it won't play back in Windows Media
> >> Player.
> > I seem to be making some progress with these parameters:
> > -vcodec dvvideo -s 720x576 -vtag dvsd -pix_fmt yuv420p -r 25 -f avi
> > pcm_s16le -ac 2 -ar 4
> > However, when playing back in WMP (which finally works) the 16x9 image is
> > squeezed into a 4x3 frame.
> > I need to make my parameters flexible so they can take either 4x3 or 16x9
> > and keep the aspect ratio on the output files, even if it's 16x9 HD
> > downscaled to PAL DV.
> Two thoughts.
> You may need to use the setdar filter to force the display aspect ratio.
> I'm not sure if WMP recognises this setting in AVI's. Premi?re Pro seems
> OK with it.
Thanks for your input Tim!
I tested some more with the file I generated with the -vtag and it seems to
work perfectly in Premiere. Correct aspect and no error messages. I seem to
recollect that WMP has had problems playing back 16x9 DV material from a
DV25 .avi file before.
From yogesh.bit2006
2 14:42:48 2012
From: yogesh.bit2006
(Yogesh Tyagi)
Date: Fri, 2 Mar :48 +0530
Subject: [FFmpeg-user] how to enable hardware acceleration in ffmpeg
Message-ID:
Is there any command line option in ffmpeg to enable hardware acceleration
for h264 decoding.I have configured and compiled ffmpeg with"
--enable-vaapi --enable-hwaccel=h264_vaapi" options but ffmpeg is not
decode_slice function of vaapi_h264.c.
Thanks and Regards,
From tevans.uk
2 15:29:13 2012
From: tevans.uk
(Tom Evans)
Date: Fri, 2 Mar :13 +0000
Subject: [FFmpeg-user] how to enable hardware acceleration in ffmpeg
In-Reply-To:
References:
Message-ID:
On Fri, Mar 2, 2012 at 1:42 PM, Yogesh Tyagi
> Is there any command line option in ffmpeg to enable hardware acceleration
> for h264 decoding.I have configured and compiled ffmpeg with"
> --enable-vaapi --enable-hwaccel=h264_vaapi" options but ffmpeg is not
> invoking ?decode_slice function of vaapi_h264.c.
As I understand it, vdpau and vaapi are useful for decoding when you
want to display that decoded content. It's not a usable API, eg for
decoding video as a step for subsequently re-encoding it (if that
makes sense).
IE, you can use vdpau/vaapi to display media using hwaccel, but you
can't use it to accelerate decoding whilst encoding.
From exampte01 at hotmail.fr
2 18:15:34 2012
From: exampte01 at hotmail.fr (aple ex)
Date: Fri, 2 Mar :34 +0100
Subject: [FFmpeg-user] Problem transcoding from MP4 (H264 - AAC) to OGG
(DIRAC - FLAC)
In-Reply-To:
References:
Message-ID:
> Date: Thu, 23 Feb :32 -0300
> From: agprus
> To: ffmpeg-user at ffmpeg.org
> Subject: [FFmpeg-user] Problem transcoding from MP4 (H264 - AAC) to OGG (DIRAC - FLAC)
> I am trying to do a lossy-to-lossless transcoding from an MP4 file (with
> one H264 video stream and one AAC audio stream) to an OGG file (with a
> DIRAC video stream and a FLAC audio stream). The command line I am using
> is this:
> ffmpeg -i input.mp4 -vcodec libschroedinger -q:v 0 -acodec flac output.ogg
> and I am getting the following errors (complete output is at the end):
> [ogg @ F29570] Unsupported codec id in stream 0
> Could not write header for output file #0 (incorrect codec parameters ?)
> Does anybody know an appropriate way to do this kind of transcoding?
> Thanks in advance,
> ffmpeg version N-38148-gb6ff81d Copyright (c)
the FFmpeg
> developers
built on Feb 23 :44 with gcc 4.6.2
configuration: --enable-gpl --enable-version3 --disable-w32threads
> --enable-runtime-cpudetect --enable-avisynth --enable-bzlib
> --enable-frei0r --enable-libopencore-amrnb --enable-libopencore-amrwb
> --enable-libfreetype --enable-libgsm --enable-libmp3lame
> --enable-libopenjpeg --enable-librtmp --enable-libschroedinger
> --enable-libspeex --enable-libtheora --enable-libvo-aacenc
> --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx
> --enable-libx264 --enable-libxavs --enable-libxvid --enable-zlib
51. 40.100 / 51. 40.100
libavcodec
4.100 / 54.
libavformat
1.100 / 54.
libavdevice
4.100 / 53.
libavfilter
2. 62.101 /
libswscale
libswresample
libpostproc
0.100 / 52.
> Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'h264aac1.mp4':
major_brand
minor_version
compatible_brands: isomiso2avc1mp41
creation_time
: Lavf52.36.0
Duration: 00:03:40.03, start: 0.000000, bitrate: 2571 kb/s
Stream #0:0(und): Video: h264 (Constrained Baseline) (avc1 /
> 0x), yuv420p, 640x480 [SAR 1:1 DAR 4:3], 2537 kb/s, 30 fps, 30
> tbr, 30 tbn, 60 tbc
creation_time
handler_name
: VideoHandler
Stream #0:1(und): Audio: aac (mp4a / 0x6134706D), 11025 Hz, stereo,
> s16, 31 kb/s
creation_time
handler_name
> [buffer @ F29F30] w:640 h:480 pixfmt:yuv420p tb:1/1000000
> sar:1/1 sws_param:
> [ogg @ F29570] Unsupported codec id in stream 0
> Output #0, ogg, to 'h264aac.ogg':
major_brand
minor_version
compatible_brands: isomiso2avc1mp41
creation_time
: Lavf54.1.100
Stream #0:0(und): Video: dirac, yuv420p, 640x480 [SAR 1:1 DAR 4:3],
> q=2-31, 200 kb/s, 30 tbn, 30 tbc
creation_time
handler_name
: VideoHandler
Stream #0:1(und): Audio: flac, 11025 Hz, stereo, s16, 128 kb/s
creation_time
handler_name
> Stream mapping:
Stream #0:0 -> #0:0 (h264 -> libschroedinger)
Stream #0:1 -> #0:1 (aac -> flac)
> Could not write header for output file #0 (incorrect codec parameters ?)
Hi according to their website dirac codec is supposed to be contained by ogg, see here :
http://diracvideo.org/schrodinger-faq/
(specially Question 6)
I have tested myself and I obtain the same result. I can provide my console output if necessary but dirac can be encoded in mpegts container.
From pobrejuanito
2 18:38:15 2012
From: pobrejuanito
Date: Fri, 2 Mar :15 -0800 (PST)
Subject: [FFmpeg-user] Generate 64kbps audio-only MPEGTS from MP4
Message-ID:
One of the Apple App Store's requirement for video apps is having a baseline
64kbps audio-only stream.
I am having trouble figuring out how to convert
my mp4 videos into mpegts to include audio-only.
Any hints as to how to generate 64kbps audio-only mpegts from mp4?
be nice to have a black screen and only audio playing which totals 64kbps.
View this message in context: http://ffmpeg-users.933282./Generate-64kbps-audio-only-MPEGTS-from-MP4-tp9211.html
Sent from the FFmpeg-users mailing list archive .
From tomnmillie
2 18:49:40 2012
From: tomnmillie
(tomnmillie)
Date: Fri, 2 Mar :40 -0800 (PST)
Subject: [FFmpeg-user] Invalid data found when processing input
In-Reply-To:
References:
Message-ID:
Thanks so much Carl!
View this message in context: http://ffmpeg-users.933282./Invalid-data-found-when-processing-input-tp9243.html
Sent from the FFmpeg-users mailing list archive .
From andrey.krieger.utkin
2 19:00:34 2012
From: andrey.krieger.utkin
(Andrey Utkin)
Date: Fri, 2 Mar :34 +0200
Subject: [FFmpeg-user] Generate 64kbps audio-only MPEGTS from MP4
In-Reply-To:
References:
Message-ID:
> One of the Apple App Store's requirement for video apps is having a baseline
> 64kbps audio-only stream. ?I am having trouble figuring out how to convert
> my mp4 videos into mpegts to include audio-only.
> Any hints as to how to generate 64kbps audio-only mpegts from mp4? ?It would
> be nice to have a black screen and only audio playing which totals 64kbps.
ffmpeg -i yourfile.mp4 -vn -acodec libfaac -b:a 64k out.ts
Andrey Utkin
From rosko
1 21:22:35 2012
From: rosko
(Steve Roskowski)
Date: Thu, 1 Mar :35 -0800
Subject: [FFmpeg-user] live transcoding flv files
Message-ID:
I have an http server which generates FLV format video "on the fly", which
allows flash to play them back in realtime - live streaming.
I would like to use ffmpeg to transcode them, specifically to lower bit
rates for mobile clients.
I cannot figure out how to get ffmpeg to do the transcoding in realtime.
FFMPEG seems to download the entire source file before beginning to
generate output.
VLC shows the exact same behavior - video does not start
playing till the entire flv file is downloaded... which on a live stream is
is this something I am missing in the options/configuration, or is this
something specific to the flv demuxer?
typical ffmpeg command
ffmpeg -i http://myserver/live_stream1/flv -b 64k -vcodec libx264 foo.flv
works fine, but only starts rendering the output file when the live stream
is closed.
Steve Roskowski
From funkyirish
1 21:45:34 2012
From: funkyirish
(booglyboo)
Date: Thu, 1 Mar :34 -0800 (PST)
Subject: [FFmpeg-user] bgra to yuv
Message-ID:
I've been trying to convert a
BGRA 30 fps video (no sound) which is
in the avi container into a yuv file for the purpose of using dirac for
compression. I've tried to use the information at
http://ffmpeg-users.933282./YUV-with-alpha-channel-td4351120.html,
but the framerate messes up and the file size changes so dramatically that I
don't believe the conversion was lossless.
To clarify:
I tried the exact
command and got a .nut file which seemed too small, then I changed the
-pix_fmt yuva420p to -pix_fmt yuv420p and tried to output a .avi file. This
resulted in something that was garbage.
I have tried many different things to convert this file including some
software that claimed it would do it but nothing preserves the video.
could use some help if anyone has the time.
View this message in context: http://ffmpeg-users.933282./bgra-to-yuv-tp6338.html
Sent from the FFmpeg-users mailing list archive .
From pobrejuanito
2 19:21:09 2012
From: pobrejuanito
Date: Fri, 2 Mar :09 -0800 (PST)
Subject: [FFmpeg-user] Generate 64kbps audio-only MPEGTS from MP4
In-Reply-To:
References:
Message-ID:
Gosh! Thank you! I added -f mpegts for the segmenter to break it up!
ffmpeg -i myvideo.mp4 -f mpegts -vn -acodec libfaac -b:a 64k
And it works!
Is there a way I can add an image to the audio playing?
I'll be submitting it to Apple Store to see if I meet their requirement.
View this message in context: http://ffmpeg-users.933282./Generate-64kbps-audio-only-MPEGTS-from-MP4-tp9344.html
Sent from the FFmpeg-users mailing list archive .
From andrey.krieger.utkin
2 19:27:25 2012
From: andrey.krieger.utkin
(Andrey Utkin)
Date: Fri, 2 Mar :25 +0200
Subject: [FFmpeg-user] Generate 64kbps audio-only MPEGTS from MP4
In-Reply-To:
References:
Message-ID:
> Gosh! Thank you! I added -f mpegts for the segmenter to break it up!
> ffmpeg -i myvideo.mp4 -f mpegts -vn -acodec libfaac -b:a 64k
> And it works!
> Is there a way I can add an image to the audio playing?
> I'll be submitting it to Apple Store to see if I meet their requirement.
Don't exactly what you got (without image) is what is required (audio only?)
I'm not an expert in apple-compliant streams, but... "Image" you add
will convert into video stream, which will take its part of bitrate,
and result will be certainly not just 64kbit audio-only stream.
Andrey Utkin
From maurice
2 19:24:54 2012
From: maurice
(Maurice Randall)
Date: Fri, 02 Mar :54 -0500
Subject: [FFmpeg-user] live transcoding flv files
In-Reply-To:
References:
Message-ID:
15:22, Steve Roskowski wrote:
> I cannot figure out how to get ffmpeg to do the transcoding in
> realtime.
FFMPEG seems to download the entire source file before beginning to
> generate output.
VLC shows the exact same behavior - video does not
> playing till the entire flv file is downloaded... which on a live
> stream is
> forever.
> ffmpeg -i http://myserver/live_stream1/flv -b 64k -vcodec libx264
> works fine, but only starts rendering the output file when the live
> is closed.
Is it possible to fetch the flv's from the same server using the rtmp
If so, that would do what you are after.
If only http can be used, maybe someone else can come up with an answer
From squeaky at sdf.org
2 20:31:41 2012
From: squeaky at sdf.org (Gary Taylor)
Date: Fri, 2 Mar :41 -0800
Subject: [FFmpeg-user] Getting a point in time as known by ffmpeg?
Message-ID:
When I watch a video (in xine or gxine for example)I see a
value displayed which is how long the video has been
If I find a scene where I want to process the
video starting at 00:00:40.000 as displayed in xine, in
ffmpeg I have to experiment to find that same spot. It's not
at 00:00:40.000, but instead 00:00:63.000.
It's like the notion of time in ffmpeg is different than
what is displayed by the players.
Is this user error,
should I be viewing them in something else, or is there a
way to use ffplay to see display the time?
fmpeg version N-37798-gcd1c12b Copyright (c)
FFmpeg developers
built on Feb 11 :52 with gcc 4.4.3
SDF Public Access UNIX System - http://sdf.lonestar.org
2 20:40:44 2012
Date: Fri, 2 Mar :44 -0900
Subject: [FFmpeg-user] Getting a point in time as known by ffmpeg?
In-Reply-To:
References:
Message-ID:
On Fri, 2 Mar :41 -0800
Gary Taylor
> When I watch a video (in xine or gxine for example)I see a
> value displayed which is how long the video has been
> playing.
If I find a scene where I want to process the
> video starting at 00:00:40.000 as displayed in xine, in
> ffmpeg I have to experiment to find that same spot. It's not
> at 00:00:40.000, but instead 00:00:63.000.
> It's like the notion of time in ffmpeg is different than
> what is displayed by the players.
Is this user error,
> should I be viewing them in something else, or is there a
> way to use ffplay to see display the time?
> fmpeg version N-37798-gcd1c12b Copyright (c)
> FFmpeg developers
built on Feb 11 :52 with gcc 4.4.3
ffplay shows the current time in the last line:
$ ffplay input.mkv
320KB vq= 5357KB sq=
In this ex