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AVCODEC_MAX_AUDIO_FRAME_SIZE macro undeclared · …
https://github.com/mpenkov/ffmpeg-tutorial/issues/13
norpol mentioned this issue on Oct 26, 2014. Redefines AVCODEC_MAX_AUDIO_FRAME_SIZE which is depricated performous/composer#20. Merged. johab mentioned this issue on Oct 27, 2014. mising symbol, dvbcut won't compile with ffmpeg 2.2.5 nextghost/dvbcut-qt4#10. Closed. This issue was closed .
'AVCODEC_MAX_AUDIO_FRAME_SIZE' undeclared · Issue …
https://github.com/3d0c/gmf/issues/1
1: error: 'AVCODEC_MAX_AUDIO_FRAME_SIZE' undeclared (first use in this function) 1: note: each undeclared identifier is reported only once for each function it appears in 1: error: 'avcodec_get_context_defaults2' undeclared (first use in this function)
Compile error: ‘AVCODEC_MAX_AUDIO_FRAME_SIZE’ …
https://github.com/ayyi/samplecat/issues/10
Hi, commit de9b4a6 defines AVCODEC_MAX_AUDIO_FRAME_SIZE if missing. This may hopefully fix things for users of newer versions of ffmpeg. I will create a new release in the next few days so that Arch users may benefit from this fix. Many thanks for reporting the issue.
FFmpeg: libavcodec/avcodec.h File Reference
https://www.ffmpeg.org/doxygen/0.6/avcodec_8h.html
AVCODEC_MAX_AUDIO_FRAME_SIZE 192000: #define : FF_INPUT_BUFFER_PADDING_SIZE 8 : Required number of additionally allocated bytes at the end of the input bitstream for decoding. #define : FF_MIN_BUFFER_SIZE 16384 : minimum encoding buffer size Used to avoid some checks during header writing. #define : FF_MAX_B_FRAMES 16: #define : …
ffmpeg - Difference in audio sample size between ...
https://stackoverflow.com/questions/66720676/difference-in-audio-sample-size-between-avcodeccontext-frame-size-and-avframe
I am playing with ffmpeg to understand audio data, but I see there is a difference between audio data, AVCodecContext->frame_size shows it to be 1152, but the value I get fromAVFrame->nb_samples shows it to be 47. Both data fields describes the same thing i.e no of samples in an audio frame per channel, then why is there a difference.
C++ (Cpp) avcodec_decode_audio4 Examples - HotExamples
https://cpp.hotexamples.com/examples/-/-/avcodec_decode_audio4/cpp-avcodec_decode_audio4-function-examples.html
if (mContext->audiocodecctx->channels>1) { size *= 2; } unsigned long long duration = (unsigned long long)( (double(mContext->audiocodecctx->time_base.num) / mContext->audiocodecctx->time_base.den) * 1000 ); mContext->audiostream->addSample( (char *)mContext->audiobuffer, size, mContext->audiotime, duration );
ffmpeg tutorial - dranger.com
http://dranger.com/ffmpeg/tutorial03.html
void audio_callback(void *userdata, Uint8 *stream, int len) { AVCodecContext *aCodecCtx = (AVCodecContext *)userdata; int len1, audio_size; static uint8_t audio_buf[(AVCODEC_MAX_AUDIO_FRAME_SIZE * 3) / 2]; static unsigned int audio_buf_size = 0; static unsigned int audio_buf_index = 0; while(len > 0) { if(audio_buf_index >= audio_buf_size) { …
FFmpeg-based console audio/video player (e.g. ffplay ...
https://gist.github.com/Lovesan/e29d815586ea20642d85875d47cefe03
if (is-> videoq. size >= MAX_QUEUE_SIZE || is-> audioq. size >= MAX_QUEUE_SIZE) {Sleep (10); continue;} rv = av_read_frame (is-> pFormatCtx, &pkt); if (rv < 0) break; Sleep (10); // need more locking, actually: if (pkt. stream_index == is-> audioStream) {PacketQueuePut (&is-> audioq, &pkt);} else if (pkt. stream_index == is-> videoStream) {PacketQueuePut (&is-> videoq, &pkt);}
Migration/10 - Libav
https://wiki.libav.org/Migration/10
AVCODEC_MAX_AUDIO_FRAME_SIZE was an arbitrarily chosen number that was considered to be large enough for all encoders, so the caller could safely use it as the buffer size. In the new audio encoding API, the output buffer should in most cases (unless the caller specifically requires otherwise, for some weird reason) be allocated by the encoder, which has a much better …
avcodec_decode_audio2_weixin_30741653的博客-程序员宝宝 - 程 …
https://cxybb.com/article/weixin_30741653/99313288
int out_size = AVCODEC_MAX_AUDIO_FRAME_SIZE; //#define AVCODEC_MAX_AUDIO_FRAME_SIZE 2<<20 uint8_t * inbuf = (uint8_t *)malloc(out_size); FILE* pFileWav; int fileSize; pFileWav= fopen("b.pcm","wb+"); while(av_read_frame(pFormatCtx, &packet)>=0) { if(packet.stream_index==audioStream) { pktdata = packet.data; pktsize = …
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