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• Revision 32970 : On progresse doucement mais surement.

  15 novembre 2009, par vxl@… — Log

  On progresse doucement mais surement.

• FFmpeg cant recognize 3 channels with each 32 bit

  4 avril 2022, par Chryfi

  I am writing the linearized depth buffer of a game to openEXR using FFmpeg. Unfortunately, FFmpeg does not adhere to the openEXR file specification fully (like allowing unsigned integer for one channel) so I am writing one float channel to openEXR, which is put into the green channel with this command -f rawvideo -pix_fmt grayf32be -s %WIDTH%x%HEIGHT% -r %FPS% -i - -vf %DEFVF% -preset ultrafast -tune zerolatency -qp 6 -compression zip1 -pix_fmt gbrpf32le %NAME%_depth_%d.exr.

  


  The float range is from 0F to 1F and it is linear. I can confirm that the calculation and linearization is correct by testing 16 bit integer (per pixel component) PNG in Blender compositor. The 16 bit integer data is written like this short s = (short) (linearzieDepth(depth) * (Math.pow(2,16) - 1)) whereas for float the linearized value is directly written to OpenEXR without multiplying with a value.

  


  However, when viewing the openEXR file it doesn't have the same "gradient" as the 16 bit png... when viewing them side by side, it appears as if the values near 0 are not linear, and they are not as dark as they should be like in the 16 bit png.
(And yes, I set the image node to linear), and comparing it with 3d tracking data from the game I cant reproduce the depth and cant mask things using the depth buffer where as with the png I can.

  


  How is it possible for a linear float range to turn out so different to a linear integer range in an image ?

  


  UPDATE :

  


  I now write 3 channels to the ffmpeg with this code

  


  float f2 = this.linearizeDepth(depth);

buffer.putFloat(f2);
buffer.putFloat(0);
buffer.putFloat(0);


  


  the byte buffer is of the size width * height * 3 * 4 -> 3 channels with each 4 bytes. The command is now -f rawvideo -pix_fmt gbrpf32be -s %WIDTH%x%HEIGHT% -r %FPS% -i - -vf %DEFVF% -preset ultrafast -tune zerolatency -qp 6 -compression zip1 -pix_fmt gbrpf32le %NAME%_depth_%d.exr which should mean that the input (byte buffer) is expecting 32 bit floats with 3 channels.

  


  FFmpeg is somehow splitting up channels or whatever... could be a bug, could be my fault ?

  


• ffmpeg libavfilter acrossfade : I jget just first input in output

  29 juillet 2024, par Alex

  I'm trying to combine two raw audio chunks (s16, 48000, mono) through acrossfade.

  


  I create filter nodes (skip all error checking here) :

  


  avfilter_graph_create_filter(&mediaIn_1,
    avfilter_get_by_name("abuffer"),
    "MediaIn_1",
    "sample_rate=48000:sample_fmt=s16:channel_layout=mono",
    NULL, graph);
avfilter_graph_create_filter(&mediaIn_2,
    avfilter_get_by_name("abuffer"),
    "MediaIn_2",
    "sample_rate=48000:sample_fmt=s16:channel_layout=mono",
    NULL, graph);
avfilter_graph_create_filter(&mediaOut,
    avfilter_get_by_name("abuffersink"),
    "MediaOut",
    NULL, NULL, graph);
avfilter_graph_create_filter(&crossfade,
    avfilter_get_by_name("acrossfade"),
    "crossfade", "nb_samples=6000:c1=tri:c2=tri", NULL, graph);


  


  Then I link them in a graph :

  


  avfilter_link(mediaIn_1, 0, crossfade, 0);
avfilter_link(mediaIn_2, 0, crossfade, 1);
avfilter_link(crossfade, 0, mediaOut, 0);
avfilter_graph_config(graph, NULL);


  


  After that I create a frame with all the chunk data :

  


  frame1 = av_frame_alloc();
frame1->format = AV_SAMPLE_FMT_S16;
frame1->nb_samples = buf1sz / 2;
frame1->sample_rate = 48000;
frame1->ch_layout.order = AV_CHANNEL_ORDER_NATIVE;
frame1->ch_layout.nb_channels = 1;
frame1->ch_layout.u.mask = AV_CH_LAYOUT_MONO;
frame1->ch_layout.opaque = NULL;
frame1->pts = 0;
frame1->duration = buf1sz / 2;
frame1->time_base.num = 1;
frame1->time_base.den = 48000;

av_frame_get_buffer(frame1, 0);
memcpy(frame1->buf[0]->data, buf1, buf1sz);


  


  Same for second chunk.
And send frames into each input buffer :

  


  av_buffersrc_add_frame_flags(mediaIn_1, frame1, 0);
av_buffersrc_add_frame_flags(mediaIn_1, NULL, AV_BUFFERSRC_FLAG_PUSH);

...
av_buffersrc_add_frame_flags(mediaIn_2, frame2, 0);
av_buffersrc_add_frame_flags(mediaIn_2, NULL, AV_BUFFERSRC_FLAG_PUSH);


  


  Then I'm getting output frame :

  


  oframe = av_frame_alloc();
av_buffersink_get_frame_flags(mediaOut, oframe, 0);
...
av_frame_unref(oframe);


  


  The oframe contains (frame1->nb_samples - 6000) number of samples instead of (frame1->nb_samples + frame2->nb_samples - something_for_xfade_needs).
Next call to av_buffersink_get_frame_flags returns AVERROR_EOF.

  


  What is wrong with this algorithm ?

  


  I tried afade filter with "t=in" input and it works fine. I think the problem is with two inputs. I don't get it.