Re: [Libav-user] ERROR: libplacebo >= 4.192.0 not found using pkg-config

On Sun, Jun 23, 2024 at 2:01 PM Gianluca Cannata wrote: > Hi everyone, > > I hope this is the right mailing list. > > I am trying to enable the libplacebo av_filter because I am writing a > simple C program using libavfilter C API but no matter how I install > libplacebo ( with libplacebo-dev fro

Re: [Libav-user] encoder frame_size for AV_CODEC_ID_PCM_S16BE

On 1/16/13, Pavel Sokolov wrote: > Hi! > > I need to encode audio with AV_CODEC_ID_PCM_S16BE codec. > Which "frame_size" is right? > number of samples per channel * channels * 2 > ___ > Libav-user mailing list > Libav-user@ffmpeg.org > http://ffmpeg.o

Re: [Libav-user] How to specify output format

On 1/21/13, jim morgenstern wrote: >>> where can I find the list of formats and their extensions? >> In the source code > > Yes but the source code is hundreds of files; can you tell me which one > has > the list and the mapping? > > >> > . What is your recommended format for uncompressed H

Re: [Libav-user] How to specify output format

[mailto:libav-user-boun...@ffmpeg.org] >> On Behalf Of Paul B Mahol >> Sent: Monday, January 21, 2013 9:50 AM >> To: This list is about using libavcodec, libavformat, libavutil, > libavdevice and >> libavfilter. >> Subject: Re: [Libav-user] How to specify output fo

Re: [Libav-user] How to specify output format

ing, you just set output format to nut and it is all what is needed. I doubt you want nut spec if you gonna use libavformat to read nut files. > > > >> -Original Message- >> From: libav-user-boun...@ffmpeg.org >> [mailto:libav-user-boun...@ffmpeg.org] >>

Re: [Libav-user] How to specify output format

mand line utilities. >> >> I mean video codec should be set to rawvideo. And pixel format for that > codec >> should be same as one detected in input. >> >> > >> > So what all do I need to know about .nut? Googled but cannot find any >> > defi

Re: [Libav-user] How to specify output format

u specify pix_fmt: example: avctx->pix_fmt = PIX_FMT_YUV420P; There is example code: http://git.videolan.org/?p=ffmpeg.git;a=blob;f=doc/examples/muxing.c > Sure wish someone would write a book > > Thanks > > > >> -Original Message- >> From: libav-user-boun

Re: [Libav-user] How to specify output format

ble) pix fmt PIX_FMT_YUV422P > > jm > >> -Original Message- >> From: libav-user-boun...@ffmpeg.org >> [mailto:libav-user-boun...@ffmpeg.org] >> On Behalf Of Paul B Mahol >> Sent: Monday, January 21, 2013 3:54 PM >> To: This list is about us

Re: [Libav-user] yuv420p mjp2 track misidentified as gray8 ?

On 2/9/13, "Rene J.V. Bertin" wrote: > Hello, > > I'm playing with Motion-JPEG2000 support in my FFusion QuickTime component, > mostly for fun, > > I have a test video produced with a recent ffmpeg version, from an mp4v > video with `-vcodec libopenjpeg kk2.mov` > > The matching ffprobe identifies

Re: [Libav-user] Using swcale to crop picture

On 2/13/13, Eric Beuque wrote: > Hi, > > I'm using avcodec and swcale to to display a MPJEG stream in a Qt > application. > > avcodec decode image as YUV image and swscale rescale it in a RGB32 image > to fit in the windows. > > I would like to provide a zoom feature, i so i am wondering if any wa

Re: [Libav-user] Processing packed YUV422 10 bit data (v210) with libav*/swscale

On 2/19/13, Robert Krueger wrote: > Hi, > > I have to convert buffers that have been decoded using apple libraries > that contain packed YUV422 10 bit data (v210) using libswscale. As I > see it, there are no suitable packed pixel formats for that purpose or > am I misreading pixfmt.h? Is there a

Re: [Libav-user] Converting audio sample buffer format

On 2/25/13, Brad O'Hearne wrote: > > On Feb 18, 2013, at 3:50 PM, Carl Eugen Hoyos wrote: > >> While I have _no_ idea what the "flv audio codec" could >> be, please use either the aconvert filter or libswresample >> directly to convert from one audio format to another. > > This has turned out to

Re: [Libav-user] Converting audio sample buffer format

On 2/25/13, Rene J.V. Bertin wrote: > > > Carl Eugen Hoyos wrote: > >>Brad O'Hearne writes: >> >>> On Feb 18, 2013, at 3:50 PM, Carl Eugen Hoyos wrote: >>> >>> > While I have _no_ idea what the "flv audio codec" could >>> > be, please use either the aconvert filter or libswresample >>> > direct

Re: [Libav-user] Converting audio sample buffer format

On 2/25/13, "Rene J.V. Bertin" wrote: > On Feb 25, 2013, at 12:53, Carl Eugen Hoyos wrote: > >> Rene J.V. Bertin writes: >> >>> I for one assume that the gcc devs would not have provided >>> default-on auto-vectorisation if the feature triggered >>> (too many) bugs they know of. >> >> You probabl

Re: [Libav-user] Converting audio sample buffer format

On 2/25/13, "Rene J.V. Bertin" wrote: > On Feb 25, 2013, at 17:26, Brad O'Hearne wrote: > >> - It would appear from the replies of more than one person that the source >> code I posted was either missed or not received. I attached a text file >> with source code to my original message. I did that

Re: [Libav-user] Why (c->frame_size = 0) ?

On 2/25/13, Joe Flowers wrote: > In the decoding_encoding.c file, I have changed > > > codec = avcodec_find_encoder(AV_CODEC_ID_MP2); > to > codec = avcodec_find_encoder(AV_CODEC_ID_PCM_S16LE); > > > but now I get > > > "Could not allocate -22 bytes for samples buffer" > > back from > > buffer_siz

Re: [Libav-user] Why (c->frame_size = 0) ?

rking on decoding_encoding.c example > > Regards > > Haridas Sagar N > > From: libav-user-boun...@ffmpeg.org [libav-user-boun...@ffmpeg.org] on > behalf of Paul B Mahol [one...@gmail.com] > Sent: Tuesday, February 26, 2013 12:23 AM &g

Re: [Libav-user] Why (c->frame_size = 0) ?

On 2/26/13, Joe Flowers wrote: >> Without exact source code? Unlikely. > > It's the decoding_encoding.c file inside of ffmpeg-1.1.2.tar.gz. So you need help what needs changed for free? Unfortunately I'm out of time to help you. > ___ > Libav-user mai

Re: [Libav-user] Converting audio sample buffer format

On 2/26/13, Brad O'Hearne wrote: > On Feb 26, 2013, at 4:39 AM, Rene J.V. Bertin wrote: > >> On Feb 26, 2013, at 02:54, Brad O'Hearne wrote: >>> >>> - Linear PCM, 24 bit little-endian signed integer, 2 channels, 44100 Hz >> >> You realise that your earlier message mentioned 32 bit float capture >

Re: [Libav-user] libswresample vs libavfilter for target format conversion

On 3/18/13, Jorge Israel Pena wrote: > Hey, this is my first time decoding audio and I'm currently trying to > figure out the best way to convert audio to the target format. > > My understanding is that when the audio is decoded, in order to play it > through the sound system (with say pulseaudio)

Re: [Libav-user] libswresample vs libavfilter for target format conversion

On 3/19/13, "Rene J.V. Bertin" wrote: > On Mar 19, 2013, at 10:37, Carl Eugen Hoyos wrote: >> >> I believe the main benefit is that if you want to (also) >> use another filter, you only need to open one filtergraph. >> If you are not using another audio filter, there probably > > So, out of curios

Re: [Libav-user] Sample format questions

On 3/26/13, Brad O'Hearne wrote: > I have four questions: > > 1. Is there a function out there to return the proper sample format based > upon whether a sample is linear / planar, signed/unsigned, and byte size? In > samplefmt.h, it seems there are functions to retrieve the sample format by > name

Re: [Libav-user] Sample format questions

On 3/26/13, Brad O'Hearne wrote: > On Mar 26, 2013, at 10:29 AM, Paul B Mahol wrote: > > Thx for the reply! > >> sample format is set by decoder to what sample format decoder outputs its >> data, >> similar apply for encoder input. > > Yes, I'm act

Re: [Libav-user] QTKit -> Libav: has it ever been done?

On 3/27/13, Brad O'Hearne wrote: > In lieu of having no luck over several weeks getting video + audio samples > captured from QTKit resampled and encoded with Libav to FLV (with video), > I've kind of hit a bit of a brick wall. The runnable Mac app and source > demonstrating this use case hasn't a

Re: [Libav-user] creating a new filter...

On 3/26/13, Richard Schilling wrote: > Greetings. > > This is my first post. I looked in the listserv archives but didn't find > anyone talking about this, so here it goes. > > I need to implement a new audio (audio only) filter. I see the example code > in filtering_audio.c that uses a buffer s

Re: [Libav-user] QTKit -> Libav: has it ever been done?

On 3/27/13, Brad O'Hearne wrote: > On Mar 27, 2013, at 4:44 AM, Rene J.V. Bertin wrote: >> So if the QTSampleBuffers contain non-native endian data the >> FFmpeg-encoded output will inevitably be the "wrong way around" unless >> it is converted before being encoded. Not? > > Thanks for the replie

Re: [Libav-user] QTKit -> Libav: has it ever been done?

On 3/27/13, Brad O'Hearne wrote: > On Mar 27, 2013, at 1:08 PM, Paul B Mahol wrote: >> Than use AV_SAMPLE_FMT_FLTP, you do not need to manually interleave >> samples. >> Each channel samples are put into separate frame->data[X] where X is >> channel >> n

Re: [Libav-user] AAC encoding error

On 4/7/13, Justin wrote: > Hi, > > I am trying to encode audio using the AAC audio encoder with the program > below, but when I call the avcodec_open2 funtion, the function always return > -733130664. I don't know where is wrong in it. I'll very appreciate someone > who can point out the wrong

Re: [Libav-user] Problems converting raw camera frames

On 4/10/13, Bogdan Popa wrote: > Hi Carl, > > Sorry for posting the code to pastebin, I hadn't realised that that was > frowned upon. > I just tried all four pixel formats that you mentioned and the results are > as follows: > > AV_PIX_FMT_ARGB: http://i.imgur.com/zg7p9AT.png > AV_PIX_FMT_ABGR: ht

Re: [Libav-user] AAC with FLV

On 4/20/13, Brad O'Hearne wrote: > On Apr 19, 2013, at 3:54 PM, Carl Eugen Hoyos wrote: > >> One takes s16 (which you probably expect), the other takes >> flt-p (which is *not* flt, at least not exactly). >> You can test if mono works to find out if planar is your >> issue. > > Yeah, that's just

Re: [Libav-user] No accelerated colorspace conversion - What does it mean?

On 4/20/13, Gary Overall wrote: > I receive the following message: > > > > > > > > [swscaler @ 0x8b0de00] No accelerated colorspace conversion found from > yuv420p to rgb24. > when issuing: > > > > > > > > sws_ctx = > sws_getContext > ( > pCodecCtx->width, >

Re: [Libav-user] swscale YUV444P10 -> RGB48LE

On 4/22/13, Al Crate wrote: > Hi all, > > I've been trying to explode a ProRes 444 movie to a sequence of 16bit > images. It would appear that this doesn't work as libswscale always > outputs non full chroma when converting YUV->RGB. > > Can someone confirm my suspicion ? What "outputs non full c

Re: [Libav-user] swscale YUV444P10 -> RGB48LE

On 4/22/13, Al Crate wrote: > On 22/04/13 10:22, Paul B Mahol wrote: >> On 4/22/13, Al Crate wrote: >>> Hi all, >>> >>> I've been trying to explode a ProRes 444 movie to a sequence of 16bit >>> images. It would appear that this doesn't wor

Re: [Libav-user] Audio quality loss while encoding

On 4/24/13, Pradeep Karosiya wrote: > Hi > I'm trying to encode decoded audio sample to an avi file. The audio samples > are decoded from different file. So I've both input and output file. The > decoded audio samples are in AV_SAMPLE_FMT_FLTP (float planar) in one large > buffer. The first half c

Re: [Libav-user] AAC with FLV

On 4/24/13, Brad O'Hearne wrote: > On Apr 24, 2013, at 5:05 AM, Pradeep Karosiya wrote: > >> Hi Brad, >> >> Have you found the solution to your issueof audio distortion. I'm also >> facing a similar issue while encoding with AAC and this is happening for >> same audio parameters sample rate: 4410

Re: [Libav-user] Audio quality loss while encoding

On 4/24/13, Brad O'Hearne wrote: > On Apr 24, 2013, at 8:30 AM, Claudio Freire wrote: > >> On Wed, Apr 24, 2013 at 8:21 AM, Pradeep Karosiya >> wrote: >>> The codec id used is AV_CODEC_ID_AAC. The encoding goes fine but I'm >>> getting >>> some noise is the final audio output. The quality is get

Re: [Libav-user] AAC with FLV

On 4/24/13, Brad O'Hearne wrote: > On Apr 24, 2013, at 10:38 AM, Paul B Mahol wrote: >> Stop this nonsense trollfest. There is no bug in ffmpeg. > > This is simple problem solving, no trolling. As another poster advised, > rechecking assumptions. All along I have treate

Re: [Libav-user] Audio quality loss while encoding

On 4/24/13, Brad O'Hearne wrote: > On Apr 24, 2013, at 10:57 AM, Paul B Mahol wrote: > >> For planar sample format, each channel is in separate array: >> >> channel left = data[0] >> channel right = data[1] >> >> If you use

Re: [Libav-user] AAC with FLV

On 4/24/13, Brad O'Hearne wrote: > On Apr 24, 2013, at 11:13 AM, Paul B Mahol wrote: > >> You are still trolling. > > I've had a working app plus source posted on Github for the better part of a > month trying to solve a problem for a client: > > https://g

Re: [Libav-user] Audio quality loss while encoding

On 4/24/13, Brad O'Hearne wrote: > On Apr 24, 2013, at 11:04 AM, Alex Cohn wrote: > >> And to set >> >> data[1]=data[0] + plane_size; >> >> is most likely not enough because both data[0] and data[1] must be >> aligned. > > Hey Alex, thx for the reply. You hit one of the magic words ("aligned") th

Re: [Libav-user] Resampling to an identical audio format

On 4/20/13, Brad O'Hearne wrote: > Suppose one were to create a resampler context for audio where the source > and destination sample formats, channel layout, and sample rates were the > same. Should the source data bytes be modified in any way when assigning to > the destination data? No it sho

Re: [Libav-user] Audio quality loss while encoding

On 4/25/13, Brad O'Hearne wrote: > On Apr 24, 2013, at 4:54 PM, Paul B Mahol wrote: > >> You call av_samples_alloc_array_and_samples() with align set to 0, which >> means >> channel data planes are aligned along 32 byte boundary. Is this really the >> case?

Re: [Libav-user] writing .avi with rawvideo and rgb data

On 4/25/13, Steffen wrote: > But write_header also gives me return value AVERROR_INVALIDDATA with > pix_fmt > BGRA and BGRA24. > I think this is because of validate_codec_tag call in mux.c. No, you most likely need to set bits per coded sample and tag. > > > > -- > View this message in context: >

Re: [Libav-user] Audio quality loss while encoding

On 4/25/13, Brad O'Hearne wrote: > On Apr 24, 2013, at 5:32 PM, Paul B Mahol wrote: > >> On 4/25/13, Brad O'Hearne wrote: >>> Here is the information on the sample buffer received from QTKit which >>> is >>> being used to fill the source data arr

Re: [Libav-user] Audio quality loss while encoding

On 4/25/13, Claudio Freire wrote: > On Thu, Apr 25, 2013 at 6:29 AM, Paul B Mahol wrote: >> I listened the sample from mentioned github repo. And its evident that >> there are >> either holes (end of every? channel data is cut off) or extra noise >> after each chan

Re: [Libav-user] av_samples_fill_arrays and memory

On 4/26/13, Brad O'Hearne wrote: > Last night I delved into the source code looking for the exact handling of > pointers and population of audio sample arrays. I had a question from the > av_samples_fill_arrays in samplefmt.c, specifically lines 167-169: > > audio_data[0] = (uint8_t *)buf; >

Re: [Libav-user] non-causal or two-pass filters

On 4/26/13, Claudio Freire wrote: > On Fri, Apr 26, 2013 at 6:44 PM, Stefano Sabatini > wrote: >> In data Friday 2013-04-26 21:21:04 +0300, Yaron Segalov ha scritto: >>> Is it possible to create a non-causal filter, or even a two-pass filter? >> >> Non-causal?? > > One that uses "future samples"

Re: [Libav-user] Audio quality loss while encoding

On 4/29/13, Claudio Freire wrote: > On Thu, Apr 25, 2013 at 2:16 PM, Paul B Mahol wrote: >> On 4/25/13, Claudio Freire wrote: >>> On Thu, Apr 25, 2013 at 6:29 AM, Paul B Mahol wrote: >>>> I listened the sample from mentioned github repo. And its evident that &g

Re: [Libav-user] Creating animated GIF using libavformat

On 5/2/13, Carl Eugen Hoyos wrote: > Gustav Gonzalez writes: > >> ERROR: gif only handles the rgb24 pixel format. > > Originally, there was a gif muxer that only accepted > rgb24 rawvideo as input. > In current git head, you will find a gif encoder (an > encoder instead of a muxer) that can be us

Re: [Libav-user] Flushing queued frames

On 5/6/13, Brad O'Hearne wrote: > I am seeing the following output in the console upon trying to close a > codec. > > [libmp3lame @ 0x10101fe00] 9 frames left in the queue on closing > > What is the proper convention for flushing the codec prior to closing it? Isn't it explained in documentation.

Re: [Libav-user] avcodec_decode_audio4() help

On 5/6/13, Gonzalo Garramuno wrote: > I placed a print statement with: >> >> const char *av_get_sample_fmt_name(enum AVSampleFormat sample_fmt); >> >> and the return value for 1.0.6 is s16, while for 1.1.4 is fltp, which >> I assume is float planar. >> >> Does this mean I need to use the swresampl

Re: [Libav-user] issues with a wavepack file

On 5/7/13, cyril apan wrote: > I tried to feed this hybrid wavepack file to ffmpeg: > http://towerofbabel.free.fr/test/onemoretime-onemorechance.wv > This is a lossy audio file but I also got the accompanying .wvc file to get > back the full lossless version of the song album. > The thing is that

Re: [Libav-user] Force ffplay to play video for a given fps

On 5/10/13, Mert Gedik wrote: > Hello team, > > I am working on a security application to show 16 cams at the same time. To > be able to do that I think, reducing the fps is a good idea. (maybe it > isn't, I am not sure...) > > I can do it this modifying my stream player to refresh it's video vie

Re: [Libav-user] Flushing Audio Frame Crash

On 5/10/13, mohM wrote: > I saw another thread on here similar to this, but it didn't really solve my > problem. Basically, I'm still working on getting my audio encoding to work, > and I'm now having trouble getting a basic audio encode to work...What I'm > trying to do is get a single frame of a

Re: [Libav-user] Flushing Audio Frame Crash

On 5/10/13, Brad O'Hearne wrote: > On May 10, 2013, at 3:44 AM, mohM wrote: > >> I saw another thread on here similar to this, but it didn't really solve >> my >> problem. Basically, I'm still working on getting my audio encoding to >> work, >> and I'm now having trouble getting a basic audio enc

Re: [Libav-user] Does anyone know how to generate quicktime compatiable H264 video by ffmpeg?

On 5/16/13, Xian Yan Yang wrote: > I've generated H264 video stream from my camera, but the video can't play > in quicktime, only works in VLC. > > Does anyone face same problem as me? Add -pix_fmt yuv420p, because quicktime supports only baseline h264. > > Thanks > Br.Luffy > __

Re: [Libav-user] Does anyone know how to generate quicktime compatiable H264 video by ffmpeg?

Thanks > BR.Luffy > > > 2013/5/16 Paul B Mahol > >> On 5/16/13, Xian Yan Yang wrote: >> > I've generated H264 video stream from my camera, but the video can't >> > play >> > in quicktime, only works in VLC. >> > >> > Does anyone

Re: [Libav-user] # of audio samples, calculated vs. codec context

On 5/18/13, Brad O'Hearne wrote: > I have the following audio use-case: > > audio capture -> resample captured audio to destination format for encoding > -> encode audio -> stream audio > > I have developed an app which has worked decently well for fairly common > sample rates (44100, 48000). Howe

Re: [Libav-user] # of audio samples, calculated vs. codec context

On 5/21/13, Brad O'Hearne wrote: > On May 20, 2013, at 8:39 AM, Brad O'Hearne > wrote: > >> I take it by sound of crickets (no response) to my question above that >> either I've done a bad job communicating the issue, or it is indeed a real >> stumper. In the event that it is the former, I'm goin

Re: [Libav-user] # of audio samples, calculated vs. codec context

On 5/21/13, Brad O'Hearne wrote: > On May 21, 2013, at 7:05 AM, Paul B Mahol wrote: > >> On 5/21/13, Brad O'Hearne wrote: >> >> I'm safe to conclude you are troller. > > In the interest of civility and productivity, let's dispense with this kin

Re: [Libav-user] Encoding with variable frame rate

On 5/21/13, Brad O'Hearne wrote: > On May 21, 2013, at 9:34 AM, Kalileo wrote: > >>> 1. Does FFmpeg support variable frame rate, or not? >> >> ffmpeg gives you the tools to handle it. Don't mix statements about ffmpeg >> command line tool with what you can do with the ffmpeg libraries. > > Does F

Re: [Libav-user] # of audio samples, calculated vs. codec context

On 5/21/13, Brad O'Hearne wrote: > On May 21, 2013, at 9:28 AM, Paul B Mahol wrote: > >> I don't see any problem, ffmpeg can resample and encode at same time. > > I didn't see any problem in principle either. In practice, it all works fine > until encountering

Re: [Libav-user] # of audio samples, calculated vs. codec context

On 5/21/13, Brad O'Hearne wrote: > On May 21, 2013, at 11:10 AM, Pradeep Karosiya wrote: > >> I had similar issue sometime back. I matched the number of sample to that >> of >> codec context. I have used buffering scheme and which still working for >> me, >> so I didn't explore any other option.

Re: [Libav-user] Encoding with variable frame rate

On 5/22/13, Brad O'Hearne wrote: > On May 22, 2013, at 9:02 AM, Robert Krueger wrote: > >> After all, most of the work is >> done by people in their spare time and I haven't found many developers >> who enjoy writing documentation (no matter how important docs are, I >> think we all agree on that

Re: [Libav-user] AVCODEC_MAX_AUDIO_FRAME_SIZE

On 5/22/13, prathap wrote: > Hi, > > I am new to ffmpeg , when i am doing work on sound functions i got errors > like this > > > In function `audio_callback': > > error: `AVCODEC_MAX_AUDIO_FRAME_SIZE' undeclared (first use in this > function) > > note: each undeclared identifier is reported only o

Re: [Libav-user] Written mjpeg files have no duration

On 5/23/13, Steffen wrote: > Written mjpeg files have no duration. Is there any way to retrieve them > from > anywhere? No, as raw mjpeg does not store durations. > > > > > -- > View this message in context: > http://libav-users.943685.n4.nabble.com/Written-mjpeg-files-have-no-duration-tp4657728

Re: [Libav-user] Encoding with variable frame rate

On 5/22/13, Robert Krueger wrote: > On Wed, May 22, 2013 at 6:53 PM, Brad O'Hearne > wrote: >> On May 22, 2013, at 9:02 AM, Robert Krueger wrote: >> >>> After all, most of the work is >>> done by people in their spare time and I haven't found many developers >>> who enjoy writing documentation (

Re: [Libav-user] Encoding with variable frame rate

On 5/23/13, Brad O'Hearne wrote: > On May 22, 2013, at 11:58 PM, Kalileo wrote: > >> Some more examples: I just re-read last night other threads on the search >> for some Info about planar aac which I remembered having seen somewhere >> here, threads such as "AAC with FLV", "Audio quality loss wh

Re: [Libav-user] qscale for vorbis encoder

On 5/20/13, YIRAN LI wrote: > Hi Guys, > > I'm trying encoding ogg video with native experimental vorbis encoder. > While documentation here says the max qscale is 10 for vorbis audio, and > default is 6, http://ffmpeg.org/trac/ffmpeg/wiki/TheoraVorbisEncodingGuide > > But I found that even if I p

Re: [Libav-user] Encoding with variable frame rate

On 5/25/13, Brad O'Hearne wrote: > On May 25, 2013, at 1:03 AM, Kalileo wrote: > >> As Robert said, please, please, please: "Please, take a step back and >> reality-check your statements. You're so off-track here." > > I am going to conclude my discourse here by passing on one more nugget of > in

Re: [Libav-user] Handling of 24 bit audio in libav* and libswresample

On 6/4/13, Hendrik Schreiber wrote: > Hello everybody: > > I'm working on a little (java) library for decoding audio using > FFmpeg/libav* and have some questions regarding the handling of 24 bit > audio. > > 1. (SHIFTING) When decoding, 24bit audio is apparently shifted, i.e. 24bit > become 32bit

Re: [Libav-user] Handling of 24 bit audio in libav* and libswresample

On 6/7/13, Hendrik Schreiber wrote: > On Jun 4, 2013, at 1:34 PM, Paul B Mahol wrote: > >> On 6/4/13, Hendrik Schreiber wrote: >>> >>> 1. (SHIFTING) When decoding, 24bit audio is apparently shifted, i.e. >>> 24bit >>> become 32bit, as there

Re: [Libav-user] Handling of 24 bit audio in libav* and libswresample

On 6/11/13, Hendrik Schreiber wrote: >> >> Note that dithering should be done when doing 32bit to 24bit case >> and source audio have >24bits used. > > Yes - definitely. > >> >>> >>> Dithering is only necessary, when converting the data somewhere in >>> between >>> (e.g. changing the sample rate w

Re: [Libav-user] Way to convert .wav file waveform into video .mjpeg file

On 6/11/13, Olivier Daubry wrote: > Hello, > > I'm searching for a way to convert an array of values (44100 samples coded > on 16bits each every second) into a series of plot (graph) images (24 per > seconds), and then make a video out of it so that I could have a converter > from .wav to .mjpeg s

Re: [Libav-user] Way to convert .wav file waveform into video .mjpeg file

ollowing filters: showspectrum showwaves avectorscope And others visualizations can be added to. > > > On Tue, Jun 11, 2013 at 4:28 PM, Paul B Mahol wrote: > >> On 6/11/13, Olivier Daubry wrote: >> > Hello, >> > >> > I'm searching for a way to convert an

Re: [Libav-user] Possible fix for rare core-dumps in H264

On 6/21/13, Carl Eugen Hoyos wrote: > Mark Stevans writes: > >> Frankly, I don't understand how patches could be >> ignored on TRAC, yet observed in "ffmpeg-devel". > > I was just describing what experience tells me. > >> But I will send my patch there > > It appears that I was unclear again, sor

Re: [Libav-user] Possible fix for rare core-dumps in H264

On 6/21/13, Carl Eugen Hoyos wrote: > Paul B Mahol writes: > >> Are you enjoying yourself? > > You mean compared to you when you (intentionally!) > commit other people's patches? What other people's patches? ___ Lib

Re: [Libav-user] mp3 live stream to stdout

On 6/23/13, pablo platt wrote: > Hi, > > I'm trying to transcode a live RTMP stream to mp3. > I can't make avconv write the output to stdout. > > I was trying: > avconv -i rtmp://127.0.0.1/audio/test -f mp3 - > and > avconv -i rtmp://127.0.0.1/audio/test -f mp3 pipe:1 > > But all I'm getting is a

Re: [Libav-user] mp3 live stream to stdout

gt; I don't see anything in the shell Really? Paste full uncut console output. > > > On Sun, Jun 23, 2013 at 3:57 AM, pablo platt wrote: > >> Sorry but the mailing list is called libav. >> How can I install ffmpeg on ubuntu 12.04 and above? >> >> >>

Re: [Libav-user] Conversion from mp3 to aac/mp4 container problem

On 6/24/13, Taha Ansari wrote: > Hi Carl, > > On your suggestion, I downloaded latest available build from Zeranoe site: > ffmpeg-20130623-git-c329713-win32. I had some trouble getting my code to > run with latest build, but finally was able to do so. > > I had to make following additions: > > ins

Re: [Libav-user] How to retrieve number of frames

On 6/24/13, Steffen wrote: > Is there any way to retirve the number of frames of a stream if > (AVStream *)stream->nb_frames == 0? Yes, by demuxing/decoding every single one. ___ Libav-user mailing list Libav-user@ffmpeg.org http://ffmpeg.org/mailman/li

Re: [Libav-user] Conversion from mp3 to aac/mp4 container problem

On 6/24/13, Taha Ansari wrote: > You can't change input sample format like that, input sample format is > >> what decoder outputs, nothing else. >> >> Same apply to sample format that encoder accepts. >> >> So you first need to make sure that you do not change sample format >> of decoder and that

Re: [Libav-user] Conversion from mp3 to aac/mp4 container problem

On 6/24/13, Taha Ansari wrote: >>>Do you have any link to any example that uses planar data correctly? I > have been trying >>since your email to get it to work, but no success so > far. > >> > Since encoding from mp4 to mp4 seems to be working fine with my modified mp4 to mp4 is not encoding but

Re: [Libav-user] Conversion from mp3 to aac/mp4 container problem

On 6/24/13, Taha Ansari wrote: >>>mp4 to mp4 is not encoding but remuxing which can also do >>> transcoding. > > Understood... only my point is: since this decode -> resample -> encode > process is happening from my code, it should be applicable to other formats > as well, shouldn't it? Yes.

Re: [Libav-user] Audio conversion from planar to non-planar formats

On 6/26/13, Taha Ansari wrote: > Hi! > > I am working on audio conversion. Looking at different example online, I > have been successfully able to convert FLTP format to S16. To my > understanding, FLTP is planar, with values in floating range: -1.0 to +1.0. > This converts to S16 just fine. Now,

Re: [Libav-user] avcodec_encode_video2() crashes after encoding a few pictures

On 6/27/13, Arvind Raman wrote: > I am developing an application using the FFmpeg libraries and am facing a > crash issue that I am finding extremely hard to debug. My printf logs > suggest that crash happens within the avcodec_encode_video2() function but I > am not sure what is causing the crash

Re: [Libav-user] LIBFAAC

On 7/2/13, Tushar Paithankar wrote: > Hi, > > In libfaac library, in the Encode call of AAC (frame.c) there is > > if (hEncoder->frameNum <= 3) /* Still filling up the buffers */ > return 0; > > What is the purpose of this buffer...? > > Can i work for single input frame and single out

Re: [Libav-user] Raw video encoding using AV_CODEC_ID_RAWVIDEO

On 7/9/13, Pradeep Karosiya wrote: > Hi, > > I'm trying to encoding images into raw video using codec id as > av_codec_id_rawvideo. > The encoding works fine till the video size is less than 2GB. However when > video size exceed 2GB > I'm getting weird video sometime just blank or sometime stitche

Re: [Libav-user] Raw video encoding using AV_CODEC_ID_RAWVIDEO

On 7/9/13, Pradeep Karosiya wrote: > Hi Paul, > > I'm using FFmpeg 1.2.1 and api which I used avcodec2_encode_video(). I'm > following muxing.c example. > The container format of my file is .avi. I'm using it with Visual Studio > 2012 on Windows 7 32bits. > I think there was a limit on avi file si

Re: [Libav-user] ffmpeg 2.0 aacdec.c depends on ape?

On 7/12/13, John Orr wrote: > The ffmpeg 2.0 version of libavformat\aacdec.c now contains some > reference to APE. In includes apetag.h and calls ff_ape_parse_tag(). > It did not do that in ffmpeg 1.2. > > When I tried to upgrade from 1.2 to 2.0, I found I had a link failure. > When I compile ffm

Re: [Libav-user] Allocate and free audio frame

On 7/20/13, Massimo Battistel wrote: > hello, > to allocate video frame buffers I use "avpicture_alloc" and to free it I > use "avpicture_free". > > to allocate audio frame buffers I first have to allocate a buffer using > "av_malloc" and then use "avcodec_fill_audio_frame". Is there a specific >

Re: [Libav-user] can't decode RTMP stream from FMS

On 7/26/13, mikeversteeg wrote: > Carl Eugen Hoyos wrote >> mikeversteeg writes: >> Then I really, really honestly cannot help you. > > OK, well thanks for trying. You did help a little bit in demonstrating > there > is apparently a bug in ffmpeg, where your specific build works and "fu

Re: [Libav-user] can't decode RTMP stream from FMS

On 7/26/13, Carl Eugen Hoyos wrote: > Paul B Mahol writes: > >> What about disabling native FFmpeg rtmp code? >> Assuming librtmp one works fine. > > It's the other way 'round... Really? AFAIK currently if you have librtmp protocols enabled at same time as na

Re: [Libav-user] Problem reading motion vectors

On 7/30/13, Luis Brocan Broki wrote: > Hi everyone! > > I'm currently developing a program that reads the motion_val table in each > frame for doing some computations. The problem if since i've upgraded to > ffmpeg 2.0 (i come from avcodec-53 version), the second level of motion_val > table is alw

Re: [Libav-user] HLS options problem

On 8/13/13, Andrey Mochenov wrote: > Hi gentlemen, > > We are using FFmpeg libraries git-ee94362 libavformat v55.2.100. > We are trying to write a simple HLS code example based on muxing.c standard > one. > Let be two input streams, video and audio (they can be synthetic, doesn't > matter). > Our

Re: [Libav-user] using ffmpeg libraries to split a video

On 8/11/13, Yousef Alhashemi wrote: > Hi, > > I asked a question on the ffmpeg support forum (ffmpeg.gusari.org) and > someone pointed me to this list, so I'm repeating my question here: > > I would like to use ffmpeg libraries to split a video into equal-sized > chunks of smaller files. For examp

Re: [Libav-user] Dinamically change volume

On 8/14/13, Massimo Battistel wrote: > hello, > I see libavfilter has the command "avfilter_graph_send_command" to change > filters parameters on runtime. > I also see not all the filters implement the "process_command" method, > volume filter including. > > Is it possible to add "process_command"

Re: [Libav-user] Watermarks implementation

On 8/15/13, Andrey Mochenov wrote: > Hi, > > We are using FFmpeg libraries git-ee94362 libavformat v55.2.100. > Our purpose is to add watermark (semi-transparent PNG image) on the video. > The corresponding option in MMpeg application is "-vf". > > Our question: How to implement the feature using

Re: [Libav-user] Encoding and Decoding Single Frames with libAV

On 9/5/13, James Board wrote: >>>On 9/5/13, James Board wrote: >>> Is there a way to use libav to encode and decode a single frame of >>> image data with a lossless codec such as ffvhuff? I know this can >>> be done with video. But if I have a single frame of image data stored >>> in a ppm or p

Re: [Libav-user] Encoding and Decoding Single Frames with libAV

On 9/5/13, James Board wrote: > Is there a way to use libav to encode and decode a single frame of > image data with a lossless codec such as ffvhuff? I know this can > be done with video. But if I have a single frame of image data stored > in a ppm or pnm file, can I read it into my libAV progr

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