libavformat

JavaFX Media Player crashes when opening MP3s even though libav 53 libraries are present?

由 谁说我不能喝 提交于 2019-12-09 07:35:27
Building a java 8 (oracle) application that uses the JavaFX media player under linux. Needing MP3 support I identified that I needed to ensure that libavformat53 and libavcodec53 were installed on the system. Now even though they are installed on the system and all the relavent .so files are linked and in the right locations. JavaFX still crashes when opening MP3s. Any ideas why this is still occurring? Research Digged deeper into the issue found target system I was installing on already had libav 55 libraries on it. Using strace I saw that java was trying to load version 55 of the libraries

How to publish selfmade stream with ffmpeg and c++ to rtmp server?

由 不羁岁月 提交于 2019-12-08 14:22:32
Have a nice day to you, people! I am writing an application for Windows that will capture the screen and send the stream to Wowza server by rtmp (for broadcasting). My application use ffmpeg and Qt. I capture the screen with WinApi, convert a buffer to YUV444(because it's simplest) and encode frame as described at the file decoding_encoding.c (from FFmpeg examples): /////////////////////////// //Encoder initialization /////////////////////////// avcodec_register_all(); codec=avcodec_find_encoder(AV_CODEC_ID_H264); c = avcodec_alloc_context3(codec); c->width=scr_width; c->height=scr_height; c-

Read H264 SPS & PPS NAL bytes using libavformat APIs

由 拟墨画扇 提交于 2019-12-08 06:48:05
问题 How to read H264 SPS & PPS NAL bytes using libavformat APIs? I tried reading video data to 'AVPacket' structure using "av_read_frame(input_avFormatContext, &avPkt)" API, from a .mp4 video (codec is h264) file. I dumped avPkt->data to a file. But 1st frame read is an IDR frame. File generated using "ffmpeg -i video.mp4 video.h264" will contain SPS & PPS in the starting before start of IDR. I want to extract raw .h264 video from .mp4 file and dump it in SPS,PPS, IDR, P1, P2... order. I want to

JavaFX Media Player crashes when opening MP3s even though libav 53 libraries are present?

由 五迷三道 提交于 2019-12-08 06:32:27
问题 Building a java 8 (oracle) application that uses the JavaFX media player under linux. Needing MP3 support I identified that I needed to ensure that libavformat53 and libavcodec53 were installed on the system. Now even though they are installed on the system and all the relavent .so files are linked and in the right locations. JavaFX still crashes when opening MP3s. Any ideas why this is still occurring? 回答1: Research Digged deeper into the issue found target system I was installing on already

Unable to store pcm audio in .mp4 container file

由 女生的网名这么多〃 提交于 2019-12-05 01:36:16
Is it possible to store a pcm audio file into .mp4 file? I used ffmpeg command "ffmpeg -i tempA.wav -acodec copy temp.mp4" but unable to store it in mp4 container file. Input #0, wav, from 'tempA.wav': Duration: 00:01:36.51, bitrate: 128 kb/s Stream #0.0: Audio: pcm_s16le, 8000 Hz, 1 channels, s16, 128 kb/s [mp4 @ 0x7d70e0] Tag [1][0][0][0]/0x00000001 incompatible with output codec id '65536' Output #0, mp4, to 'temp.mp4': Stream #0.0: Audio: pcm_s16le, 8000 Hz, 1 channels, 128 kb/s Stream mapping: Stream #0.0 -> #0.0 Could not write header for output file #0 (incorrect codec parameters ?)

keyframe is not a keyframe? AV_PKT_FLAG_KEY does not decode to AV_PICTURE_TYPE_I

由 谁都会走 提交于 2019-12-04 14:14:23
问题 After decoding a packet containing AV_PKT_FLAG_KEY in the flags, I was expecting to get I-frames, but instead I got P-frames: After a call to: avcodec_decode_video2(codecCtx, frame, &frameFinished, &packet); // mpeg2 video I print out the following as a sanity check: printf("packet flags: %d picture type: %c\n", packet.flags, av_get_picture_type_char(frame->pict_type)); Returns the output: packet flags: 1 picture type: P When I was expecting: packet flags: 1 picture type: I Where '1' == AV

Different between fragmented mp4 files generated by ffmpeg and by code

由 ▼魔方 西西 提交于 2019-12-04 12:33:19
问题 Currently I have a problem when generating fragmented MP4 file from code using libavformat. My file can be played using VLC, but can't be streamed (via WebSocket) and played (via MediaSource) in (Chrome) browser. (I used this to test streaming fragmented MP4 file to browser via WebSocket). Note: The files below is encoded by Baseline profile, level 4. So you should change the MIME type (in index.html) to const mimeCodec = 'video/mp4; codecs="avc1.42C028"'; to be able to play them. I checked

Decode MP3, then increase the audio volume, and then encode the new audio

由 眉间皱痕 提交于 2019-12-04 11:11:48
问题 I want to first decode a MP3 audio file, and then increase the volume of the audio, and then encode it again into a new MP3 file. I want to use libavformat or libavcodec for this. Can you help me how I can do this? Any example? 回答1: You can use the "-filter" parameter with the "volume" option to set a multiplier for the audio. More info: http://ffmpeg.org/ffmpeg-filters.html#volume Since you are dealing only with MP3 files (that have only one audio track), you can use the "-af" parameter,

Trouble syncing libavformat/ffmpeg with x264 and RTP

由 [亡魂溺海] 提交于 2019-12-03 10:48:09
问题 I've been working on some streaming software that takes live feeds from various kinds of cameras and streams over the network using H.264. To accomplish this, I'm using the x264 encoder directly (with the "zerolatency" preset) and feeding NALs as they are available to libavformat to pack into RTP (ultimately RTSP). Ideally, this application should be as real-time as possible. For the most part, this has been working well. Unfortunately, however, there is some sort of synchronization issue:

Trouble syncing libavformat/ffmpeg with x264 and RTP

由 穿精又带淫゛_ 提交于 2019-12-03 00:21:10
I've been working on some streaming software that takes live feeds from various kinds of cameras and streams over the network using H.264. To accomplish this, I'm using the x264 encoder directly (with the "zerolatency" preset) and feeding NALs as they are available to libavformat to pack into RTP (ultimately RTSP). Ideally, this application should be as real-time as possible. For the most part, this has been working well. Unfortunately, however, there is some sort of synchronization issue: any video playback on clients seems to show a few smooth frames, followed by a short pause, then more
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