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How to convert sample rate from …
https://stackoverflow.com/questions/14989397/how-to-convert-sample-rate-from-av-sample-fmt-fltp-to-av-sample-fmt-s16
At some point in the last 2-3 years FFmpeg's AAC decoder's output format changed from AV_SAMPLE_FMT_S16 to AV_SAMPLE_FMT_FLTP. This means that each audio channel has it's own buffer, and each sample value is a 32-bit floating point value scaled from -1.0 to +1.0. Whereas with AV_SAMPLE_FMT_S16 the data is in a single buffer, with the samples …
ffmpeg avcodec_open2 changed audio format in ...
https://www.tutorialguruji.com/cpp/ffmpeg-avcodec_open2-changed-audio-format-in-avcodeccontext/
avcodec_decode_audio4(avAudioCodecCtx, frame, &frameComplete, avPacket); frame->format // contains AV_SAMPLE_FMT_FLTP Answer. Yes, this is expected. So, this may be slightly annoying, but avcodec_open2() simply sets sample_fmt to the “most likely” format given the information stored in the AVCodecContext when opening the codec. For some ...
FFmpeg - resampling from AV_SAMPLE_FMT_FLTP to …
https://stackoverflow.com/questions/22822515/ffmpeg-resampling-from-av-sample-fmt-fltp-to-av-sample-fmt-s16-got-very-bad-so
In new ffmpeg, the output samples are fltp format, so I have to convert it from AV_SAMPLE_FMT_FLTP to AV_SAMPLE_FMT_S16 . PS: in old ffmpeg as libavcodec 54.12.100, it is directly S16, so do not need resampling and without any sound quality problem. Then I've tried three ways to resampling, using swr_convert. using avresample_convert. convert ...
C++ (Cpp) avcodec_decode_audio4 Examples - HotExamples
https://cpp.hotexamples.com/examples/-/-/avcodec_decode_audio4/cpp-avcodec_decode_audio4-function-examples.html
C++ (Cpp) avcodec_decode_audio4 - 30 examples found. These are the top rated real world C++ (Cpp) examples of avcodec_decode_audio4 extracted from open source projects. You can rate examples to help us improve the quality of examples.
ffmpeg how to save decoded audio data to pcm - Tutorial Guruji
https://www.tutorialguruji.com/android/ffmpeg-how-to-save-decoded-audio-data-to-pcm/
I know that the decoded pcm is AV_SAMPLE_FMT_FLTP any help would be appreciated. Answer. FLTP is planar float, so in case of stereo, you have two buffers, data[0] and data[1], which are per-channel planes.
メソッドとFFmpegのlibavcodecのオーディオリサンプリン …
https://www.codetd.com/ja/article/6411389
libavcodecのリサンプリングプロセスは、複数のチャネルデータがマージされる必要と平面源のいくつかのタイプのために特に、左右のチャンネルは、同じではありません。. libswresampleは、これらのトラブルを省略すること。. 以下にした後、データ型AV_SAMPLE_FMT ...
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