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• Libx264 without cygwin1.dll ?

  11 juin 2012, par Khuất Văn Phiến

  I have build libx264-125.dll by MinGW and Msys. However, libx254-125.dll need call cygwin1.dll and cyggcc_s-1.dll for run. Could anyone tell me how to build static libx264-125.dll Thanks for your help!

• Tune FFmpeg H.264 Decoder

  21 avril 2012, par Ilya Shcherbak

  I'm using the FFmpeg avcodec to decode live video - the avcodec_decode_video2 function from libx264 to be exact. How can I decrease the decoding time for each frame? At the moment it takes 20 ms for each frame (frame size about 1.5 KB).

• Using ffmpeg with Flash Media Server and HDS

  20 avril 2012, par Jonathan

  I want to use ffmpeg to encode and publish a live stream to Flash Media Server. In order to support iOS devices, I need to implement HTTP Live Streaming as well. The video needs to be in H.264 format and the audio should be AAC. I don't have much experience working with ffmpeg, and I'm having a hard time getting this to work. This is the command that I've tried (and some variations as well):

  ffmpeg.exe -threads 15 -f dshow -i video="USB2.0 UVC WebCam":audio="Microphone (Realtek High Defini" \
        -map_channel 0.1.1 -r 24 -acodec libvo_aacenc -ar 22050 -ab 128k -vcodec libx264 \
        -s vga -vb 100k -f flv "rtmp:///livepkgr/livestream1?adbe-live-event=liveevent" \
        -r 24 -acodec libvo_aacenc -ar 22050 -ab 128k -vcodec libx264 -s qvga -vb 200k \
        -f flv "rtmp:///livepkgr/livestream2?adbe-live-event=liveevent" \
        -r 24 -acodec libvo_aacenc -ar 22050 -ab 128k -vcodec libx264 -s vga -vb 350k 
        -f flv "rtmp:///livepkgr/livestream3?adbe-live-event=liveevent"
  

  When I run this, it appears to connect to FMS, but then I get a lot of error messages about dropped frames - I'm not sure if ANY frames get encoded successfully. My CPU usage is very high as well. I get a 404 error from FMS when I enter the URL of the *.m3u8 file for one of the individual streams (the main livestream.m3u8 file is accessible though). I have also tried outputting to a file instead of FMS, with no success. All I get is some very garbled sound and no video.

  Any suggestions for what options/commands I should use to get this working? Is anyone using ffmpeg with FMS to do HTTP Dynamic Streaming / HLS with MP4 video? I've been struggling to get HDS/HLS working for some time now, and any help would be much appreciated! It shouldn't make a difference, but I'm using FMS on Amazon EC2 with their AMI image.

  Thanks!

• streaming H.264 over RTP with libavformat

  16 avril 2012, par Jacob Peddicord

  I've been trying over the past week to implement H.264 streaming over RTP, using x264 as an encoder and libavformat to pack and send the stream. Problem is, as far as I can tell it's not working correctly.

  Right now I'm just encoding random data (x264_picture_alloc) and extracting NAL frames from libx264. This is fairly simple:

  x264_picture_t pic_out;
  x264_nal_t* nals;
  int num_nals;
  int frame_size = x264_encoder_encode(this->encoder, &nals, &num_nals, this->pic_in, &pic_out);
  
  if (frame_size <= 0)
  {
      return frame_size;
  }
  
  // push NALs into the queue
  for (int i = 0; i < num_nals; i++)
  {
      // create a NAL storage unit
      NAL nal;
      nal.size = nals[i].i_payload;
      nal.payload = new uint8_t[nal.size];
      memcpy(nal.payload, nals[i].p_payload, nal.size);
  
      // push the storage into the NAL queue
      {
          // lock and push the NAL to the queue
          boost::mutex::scoped_lock lock(this->nal_lock);
          this->nal_queue.push(nal);
      }
  }
  

  nal_queue is used for safely passing frames over to a Streamer class which will then send the frames out. Right now it's not threaded, as I'm just testing to try to get this to work. Before encoding individual frames, I've made sure to initialize the encoder.

  But I don't believe x264 is the issue, as I can see frame data in the NALs it returns back. Streaming the data is accomplished with libavformat, which is first initialized in a Streamer class:

  Streamer::Streamer(Encoder* encoder, string rtp_address, int rtp_port, int width, int height, int fps, int bitrate)
  {
      this->encoder = encoder;
  
      // initalize the AV context
      this->ctx = avformat_alloc_context();
      if (!this->ctx)
      {
          throw runtime_error("Couldn't initalize AVFormat output context");
      }
  
      // get the output format
      this->fmt = av_guess_format("rtp", NULL, NULL);
      if (!this->fmt)
      {
          throw runtime_error("Unsuitable output format");
      }
      this->ctx->oformat = this->fmt;
  
      // try to open the RTP stream
      snprintf(this->ctx->filename, sizeof(this->ctx->filename), "rtp://%s:%d", rtp_address.c_str(), rtp_port);
      if (url_fopen(&(this->ctx->pb), this->ctx->filename, URL_WRONLY) < 0)
      {
          throw runtime_error("Couldn't open RTP output stream");
      }
  
      // add an H.264 stream
      this->stream = av_new_stream(this->ctx, 1);
      if (!this->stream)
      {
          throw runtime_error("Couldn't allocate H.264 stream");
      }
  
      // initalize codec
      AVCodecContext* c = this->stream->codec;
      c->codec_id = CODEC_ID_H264;
      c->codec_type = AVMEDIA_TYPE_VIDEO;
      c->bit_rate = bitrate;
      c->width = width;
      c->height = height;
      c->time_base.den = fps;
      c->time_base.num = 1;
  
      // write the header
      av_write_header(this->ctx);
  }
  

  This is where things seem to go wrong. av_write_header above seems to do absolutely nothing; I've used wireshark to verify this. For reference, I use Streamer streamer(&enc, "10.89.6.3", 49990, 800, 600, 30, 40000); to initialize the Streamer instance, with enc being a reference to an Encoder object used to handle x264 previously.

  Now when I want to stream out a NAL, I use this:

  // grab a NAL
  NAL nal = this->encoder->nal_pop();
  cout << "NAL popped with size " << nal.size << endl;
  
  // initalize a packet
  AVPacket p;
  av_init_packet(&p);
  p.data = nal.payload;
  p.size = nal.size;
  p.stream_index = this->stream->index;
  
  // send it out
  av_write_frame(this->ctx, &p);
  

  At this point, I can see RTP data appearing over the network, and it looks like the frames I've been sending, even including a little copyright blob from x264. But, no player I've used has been able to make any sense of the data. VLC quits wanting an SDP description, which apparently isn't required.

  I then tried to play it through gst-launch:

  gst-launch udpsrc port=49990 ! rtph264depay ! decodebin ! xvimagesink

  This will sit waiting for UDP data, but when it is received, I get:

  ERROR: element /GstPipeline:pipeline0/GstRtpH264Depay:rtph264depay0: No RTP format was negotiated. Additional debug info: gstbasertpdepayload.c(372): gst_base_rtp_depayload_chain (): /GstPipeline:pipeline0/GstRtpH264Depay:rtph264depay0: Input buffers need to have RTP caps set on them. This is usually achieved by setting the 'caps' property of the upstream source element (often udpsrc or appsrc), or by putting a capsfilter element before the depayloader and setting the 'caps' property on that. Also see http://cgit.freedesktop.org/gstreamer/gst-plugins-good/tree/gst/rtp/README

  As I'm not using GStreamer to stream itself, I'm not quite sure what it means with RTP caps. But, it makes me wonder if I'm not sending enough information over RTP to describe the stream. I'm pretty new to video and I feel like there's some key thing I'm missing here. Any hints?

• Given an x264 stream and an ogg vorbis stream, how do I make a muxed stream that mplayer/VLC can read ?

  14 avril 2012, par dascandy

  I'm confused and a bit stuck with this question. All I can find on Google is basic usage of transcoding software, which is not related to the question.

  I'm making a game and I'd like to include native capture ability to stream video. I would much like to stream this to a standard-ish client, such as VLC. It needs to be both in a format it recognizes and it needs to be multiplexed in order for this to work.

  My question therefore is, I know how to encode stuff from raw video frames to x264 (see also How to encode series of images into H264 using x264 API? (C/C++) ). I know how to encode raw audio samples into ogg/vorbis. Now, how do I put one and one together for VLC?