Recherche avancée

Sur d’autres sites (8642)

• Is there any way to switch the resolution on the fly when streaming with RTMP ?

  25 mai 2021, par wwd

  I built nginx with nginx-http-flv-module as the RTMP server. And I used ffmpeg-python to upload the stream. I have searched a lot about "how to switch the resolution on the fly". However, it seems like that nobody do something like that.

  


  So, I did a test by periodically rerunning a new ffmpeg uploading process while the client(opencv-python) keeps receiving the data.
I found that the resolution was switched successfully, but the frame near the switch was often broken. And opencv sometimes exits without any logs.

  


  Is there any way to achieve this ?

  


  Here is the test code of my streamer :

  


  import cv2
import ffmpeg


if __name__ == "__main__":
    cap1 = cv2.VideoCapture("videos/1944x960/video.mp4")
    cap2 = cv2.VideoCapture("videos/972x480/video.mp4")

    pushing_process = None
    count = 0
    hr = True

    while True:
        ret1, frame1 = cap1.read()
        ret2, frame2 = cap2.read()
        if not ret1 or not ret2:
            break

        if hr:
            if pushing_process is None:
                pushing_process = ffmpeg.input("pipe:", format="rawvideo", pix_fmt="bgr24",
                                               s=f"{1944}x{960}") \
                    .output("rtmp://127.0.0.1:1935/myapp/s", pix_fmt="yuv420p", f="flv", vcodec="h264",
                            loglevel="error") \
                    .global_args("-re").run_async(pipe_stdin=True)

            frame = frame1
        else:
            if pushing_process is None:
                pushing_process = ffmpeg.input("pipe:", format="rawvideo", pix_fmt="bgr24",
                                               s=f"{972}x{480}") \
                    .output("rtmp://127.0.0.1:1935/myapp/s", pix_fmt="yuv420p", f="flv", vcodec="h264",
                            loglevel="error") \
                    .global_args("-re").run_async(pipe_stdin=True)
            frame = frame2
        pushing_process.stdin.write(frame.tobytes())
        count += 1
        if count == 60:
            hr = not hr
            pushing_process.stdin.close()
            pushing_process.wait()
            pushing_process = None
            count = 0

    pushing_process.stdin.close()
    pushing_process.wait()



  


  And the client :

  


  import cv2

if __name__ == "__main__":
    cap = cv2.VideoCapture("http://127.0.0.1/live?app=myapp&stream=s")
    while True:
        ret, frame = cap.read()
        if not ret:
            break
        cv2.imshow("frame", cv2.resize(frame, (1944, 960), interpolation=cv2.INTER_CUBIC))
        cv2.waitKey(0)


  


• avcodec_decode_video2 fails to decode after frame resolution change

  10 avril 2021, par Krzysztof Kansy

  I'm using ffmpeg in Android project via JNI to decode real-time H264 video stream. On the Java side I'm only sending the the byte arrays into native module. Native code is running a loop and checking data buffers for new data to decode. Each data chunk is processed with :
  

  



  int bytesLeft = data->GetSize();
int paserLength = 0;
int decodeDataLength = 0;
int gotPicture = 0;
const uint8_t* buffer = data->GetData();
while (bytesLeft > 0) {
    AVPacket packet;
    av_init_packet(&packet);
    paserLength = av_parser_parse2(_codecPaser, _codecCtx, &packet.data, &packet.size, buffer, bytesLeft, AV_NOPTS_VALUE, AV_NOPTS_VALUE, AV_NOPTS_VALUE);
    bytesLeft -= paserLength;
    buffer += paserLength;

    if (packet.size > 0) {
        decodeDataLength = avcodec_decode_video2(_codecCtx, _frame, &gotPicture, &packet);
    }
    else {
        break;
    }
    av_free_packet(&packet);
}

if (gotPicture) {
// pass the frame to rendering
}


  



  The system works pretty well until incoming video's resolution changes. I need to handle transition between 4:3 and 16:9 aspect ratios. While having AVCodecContext configured as follows :

  



  _codecCtx->flags2|=CODEC_FLAG2_FAST;
_codecCtx->thread_count = 2;
_codecCtx->thread_type = FF_THREAD_FRAME;

if(_codec->capabilities&CODEC_FLAG_LOW_DELAY){
    _codecCtx->flags|=CODEC_FLAG_LOW_DELAY;
}


  



  I wasn't able to continue decoding new frames after video resolution change. The got_picture_ptr flag that avcodec_decode_video2 enables when whole frame is available was never true after that.
  
This ticket made me wonder if the issue isn't connected with multithreading. Only useful thing I've noticed is that when I change thread_type to FF_THREAD_SLICE the decoder is not always blocked after resolution change, about half of my attempts were successfull. Switching to single-threaded processing is not possible, I need more computing power. Setting up the context to one thread does not solve the problem and makes the decoder not keeping up with processing incoming data.
  
Everything work well after app restart.
  

  



  I can only think of one workoround (it doesn't really solve the problem) : unloading and loading the whole library after stream resolution change (e.g as mentioned in here). I don't think it's good tho, it will propably introduce other bugs and take a lot of time (from user's viewpoint).

  



  Is it possible to fix this issue ?

  



  EDIT :
  
I've dumped the stream data that is passed to decoding pipeline. I've changed the resolution few times while stream was being captured. Playing it with ffplay showed that in moment when resolution changed and preview in application froze, ffplay managed to continue, but preview is glitchy for a second or so. You can see full ffplay log here. In this case video preview stopped when I changed resolution to 960x720 for the second time. (Reinit context to 960x720, pix_fmt: yuv420p in log).

  


• FFmpeg - Convert video changing resolution and bitrates

  8 avril 2021, par parvares

  I’m trying to convert a video and change these parameters :

  


  resolution : from 1.280x720 to 640x360 pixels

  


  average bitrate : from 1.410 to 300 kb/s

  


  maximum bitrate : from 1.680 to 2500 kb/s

  


  Sampling rate : from 48.0 to 44.1 kHz

  


  Bit rate audio : 128 kb/s

  


  I used this string with ffmpeg vers. 4.3.2-2021-02-20 full build :

  


  ffmpeg -i input.mp4 -c:v -vf scale=640:360 libx264 -b:v 300k -maxrate 2500k -bufsize 2500k -crf 25 -c:a -ar 44100 libfdk_aac -b:a 128k output.mp4


  


  I get the error : “ffmpeg unable to find a suitable output format for scale ‘640:360’ scale 640:360 : invalid argument”. Could you help me please ?

  


  Below is the original file analyzed by MediaInfo, thanks !

  


  General

  


  Complete name : C :\Users\parvares\Desktop\input.mp4
Format : MPEG-4
Format profile : Base Media
Codec ID : isom (isom/iso2/avc1/mp41)
File size : 424 MiB
Duration : 38 min 21 s
Overall bit rate : 1 545 kb/s
Writing application : Lavf58.64.100

  


  Video

  


  ID : 1
Format : AVC
Format/Info : Advanced Video Codec
Format profile : High@L3.1
Format settings : CABAC / 4 Ref Frames
Format settings, CABAC : Yes
Format settings, Reference frames : 4 frames
Codec ID : avc1
Codec ID/Info : Advanced Video Coding
Duration : 38 min 21 s
Bit rate : 1 410 kb/s
Maximum bit rate : 1 680 kb/s
Width : 1 280 pixels
Height : 720 pixels
Display aspect ratio : 16:9
Frame rate mode : Constant
Frame rate : 25.000 FPS
Color space : YUV
Chroma subsampling : 4:2:0
Bit depth : 8 bits
Scan type : Progressive
Bits/(Pixel*Frame) : 0.061
Stream size : 387 MiB (91%)
Writing library : x264 core 161 r3027 4121277
Encoding settings : cabac=1 / ref=2 / deblock=1:0:0 / analyse=0x3:0x113 / me=hex / subme=6 / psy=1 / psy_rd=1.00:0.00 / mixed_ref=1 / me_range=16 / chroma_me=1 / trellis=1 / 8x8dct=1 / cqm=0 / deadzone=21,11 / fast_pskip=1 / chroma_qp_offset=-2 / threads=12 / lookahead_threads=2 / sliced_threads=0 / nr=0 / decimate=1 / interlaced=0 / bluray_compat=0 / constrained_intra=0 / bframes=3 / b_pyramid=2 / b_adapt=1 / b_bias=0 / direct=1 / weightb=1 / open_gop=0 / weightp=1 / keyint=250 / keyint_min=125 / scenecut=40 / intra_refresh=0 / rc_lookahead=125 / rc=crf / mbtree=1 / crf=20.0 / qcomp=0.60 / qpmin=0 / qpmax=69 / qpstep=4 / vbv_maxrate=1680 / vbv_bufsize=8400 / crf_max=0.0 / nal_hrd=none / filler=0 / ip_ratio=1.40 / aq=1:1.00
Color range : Limited
Color primaries : BT.709
Transfer characteristics : BT.709
Matrix coefficients : BT.709
Codec configuration box : avcC

  


  Audio

  


  ID : 2
Format : AAC LC
Format/Info : Advanced Audio Codec Low Complexity
Codec ID : mp4a-40-2
Duration : 38 min 21 s
Bit rate mode : Constant
Bit rate : 129 kb/s
Channel(s) : 2 channels
Channel layout : L R
Sampling rate : 48.0 kHz
Frame rate : 46.875 FPS (1024 SPF)
Compression mode : Lossy
Stream size : 35.3 MiB (8%)
Default : Yes
Alternate group : 1