Hi everyone, I'm trying to slow down a mp4 video. I've read that i should use setpts filter for video and atempo filter for audio. The setpts filter works pretty good, but there is no way to make atempo works. I'm using the transcoding ffmpeg example ( https://ffmpeg.org/doxygen/trunk/doc_2examples_2transcoding_8c-example.html), and changed the filter_spec from "null" and "anull" to "setpts=2*PTS" and "atempo=0.5" respectively. The problem is the audio encoder complains about the frame size:
[libfdk_aac @ 06de5960] more samples than frame size (avcodec_encode_audio2) Here is the complete log for the execution: Input #0, mov,mp4,m4a,3gp,3g2,mj2, from '../highlight.mp4': Metadata: major_brand : isom minor_version : 512 compatible_brands: isomiso2avc1mp41 encoder : Lavf56.38.100 Duration: 00:00:10.09, start: 0.000000, bitrate: 734 kb/s Stream #0:0(und): Video: h264 (High) (avc1 / 0x31637661), yuvj420p(pc, bt709), 432x240 [SAR 1:1 DAR 9:5], 597 kb/s, 29.97 fps, 29.97 tbr, 120k tbn, 119.88 tbc (default) Metadata: handler_name : VideoHandler Stream #0:1(und): Audio: aac (LC) (mp4a / 0x6134706D), 48000 Hz, stereo, fltp, 127 kb/s (default) Metadata: handler_name : SoundHandler [libx264 @ 01b115e0] using SAR=1/1 [libx264 @ 01b115e0] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2 AVX [libx264 @ 01b115e0] profile High, level 3.1 Output #0, mp4, to '../highlightLento.mp4': Stream #0:0: Video: h264 (libx264), yuv420p, 432x240 [SAR 1:1 DAR 9:5], q=10-51, 128 kb/s, 119.88 tbc Stream #0:1: Audio: aac (libfdk_aac), 48000 Hz, stereo, s16, 128 kb/s [mp4 @ 01b11080] Using AVStream.codec.time_base as a timebase hint to the muxer is deprecated. Set AVStream.time_base instead. [mp4 @ 01b11080] Using AVStream.codec.time_base as a timebase hint to the muxer is deprecated. Set AVStream.time_base instead. [swscaler @ 06e288e0] deprecated pixel format used, make sure you did set range correctly [libfdk_aac @ 06de5960] more samples than frame size (avcodec_encode_audio2) I searched for the exact line where the message is printed and it compares the nb_samples in the AVFrame with the frame_size in the AVCodecContext gave to the avcodec_encode_audio2 function. So the question is, what is the correct way to set up the output codec context to works with the atempo filter? I supposed the filter will do all the work but it seems i was wrong. Any help would be very appreciated. Thanks in advance. Kind regards, Sebastián Arriagada.
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