FFmpeg Tutorial (1)

FFmpeg is a leading source for encode/decode video, audio and widely use
But,... the only problem is ffmpeg nearly don't have enough documents

I'm currently working with ffmpeg and live streaming, And I'll note here some experience I meet while working and investigate ffmpeg :)

OK, first, this is some useful link:

https://www.ffmpeg.org/

http://dranger.com/ffmpeg/ <- some code on this site is depricated, but this still useful to understand how ffmpeg work, how to display movie, and some issues

https://github.com/chelyaev/ffmpeg-tutorial/issues/11 <- this is same as dranger but updated sourcecode (still have one problem with stereo planar audio, you will hear noise instead of correct audio, using this to resample: https://github.com/illuusio/ffmpeg-tutorial)

And this is some term you should to know 
(google it for better result if you don't understand what I explain:D)

> streaming

+ split video file into frames and send to client buffer

> terms:

+ live streaming: one record and broadcast, one received and display in real time

+ rtmp: real time messaging protocol. it’s tcp-base protocol

+ rtmps: rtmp with tls/ssl connection

+ rtmpe: rtmp encrypted using adobe’s own security

+ rtmp send smooth because it’s send fragments, fragment size is negociate dynamically between server and client. default for audio is 64 and video is 128

> audio:

+ in real world (analog), audio have frequency and volumes. in digital world, we using:

+ sample rate (how frequency sound record) to represented frequency. the higher sample rate, the more accurate the sound describe.

=> how often or how much the sound is describe

+ bit rate (how good sound record) to represented volume. the higher bitrate, the more accuracy we describe how loud the volume is.

=> how well the sound  is describe

+ bit depth: mean how well we describe the volume, i.e: 16bit, mean we have 65536 volumes level to choose

==> when going in audio devices, the audio processor base on sample rate and bit rate to re-produce the analog signal (we can hear digital signal)

(So, how to record sound better ? is improve bitrate or samplerate ? as mention above, sound is both frequency and volume, so, keep the balance of them is the best. Usually using 44.1Khz/16bit, 44.1 is from  Nyquist–Shannon sampling theorem so 44.1/2 = 20.5Khz - human maximum freq)

> audio in ffmpeg:

+ ffmpeg supply samples, but it’ll not give partial sample (even stereo).

> video synced

+ DTS: Decoding Time Stamp => when we need to decode frame

+ PTS: Presentation Time Stamp => when we need to display frame

=> so the decoding order and display order of frames are not identical. then, we need to know our newly decoded frame PTS to display it right time

+ How video are stored: I-Intra, B-Bidirectional, P-Predicted frames

+ I frame contain full image.

+ P frame depend on I frame and previous P frames

+ B frame depend on both previous and next frames

=> FFmpeg reorder the packets so that the DTS of the packet being processed by avcodec_decode_video() will always be the same as the PTS of the frame it returns

I'm working with ffmpeg on MacOS and iOS. So, next post I'll note some build, config project, .v.v and how to using cocoa framework to display raw frame or playing raw audio samples :)