Ffmpeg which bitrate

When I check the file after they have been created with mediainfo I notice that the audio bitrate displayed is always exactly the same I as specified, but video bitrate is different. I was just wondering why this is?

Why is audio bitrate always the same as selected and not video? The full command is:. The issue is that you are using -crf with -b:v. If you want to have a specific bitrate, remove the crf command. CRF is "constant rate factor", which means a quality variable is specified then a variable bitrate is calculated based on the content.

It's my favorite for h. Setting a constant bitrate works as expected. Compression will be nearly exactly what you entered, but quality may suffer for it. I never use it for h. I can't think of a use case where I would. The bitrate you specify is a target bitrate, not a guaranteed bitrate.

There is no way to know exactly how large a frame will be until after its encoded. Encoders use something called rate control to achieve this.

If the frames end up being slightly larger than expected, it will more aggressively loose information on the next frame. Since most people specify bitrate to target specific use cases streaming over the internet for example a lot of stream control algorithms err on the side of to low, as too high could cause issues like buffering.

Sign up to join this community. Outdev disables output devices using sndio. If you're not using OpenBSD, you probably don't have to have these enabled. We already know it's gcc, but here it is configured. XML is used to configure the fonts. It offers mixers, filters and generators that you can control through parameters you send in queries to its plugin API. GnuTLS provides this support if you decided you didnt want to use openssl, libtls or mbedtls.

Sorry, no actual emus. So basically if you find yourself stuck in the 80s, this is the library for you. It's for predicting subjective video quality based on reference and distorted video sequence. You can use the metric to evaluate the quality of different video codecs, encoders, encoding settings and transmission variants in order to achieve the best performance for your video.

The library libvmaf gives you tools for scoring and measuring video quality according to this metric. It supports bitmap and scalable formats.

It encodes narrowband signals at variable bit rates. Each library handles a different flavor of adaptive, multi-rate speech codec. It allows you to change the tempo and pitch of an audio stream or recording dynamically.

This is necessary to prevent changes in speed and pitch of audio that can occur during a transfer of recorded material if a tool like this is not in use.

Speex is an audio codec designed specifically to reproduce human speech. It's free an can be used for VoIP applications and podcasts. It's based on the CELP coding algorithm. The advantage is use of UDP allows for delivery of video across unreliable or dirty networks. Vorbis is a free, open-source software project run by Xiph. It's usually used with the ogg container format.

Libopus is another of the Xiph. Same deal with libtheora , except it's Xiph's video codec. It's free and it uses lossy compression. It can be used to remove undesirable shakes and jitters.

The library is based on a codec implementation by VisualOn as part of the Stagefright framework from the Google Android project. It serves as the reference software implementation for the VP8 and VP9 video coding formats, and for AV1 a special fork named libaom that was stripped of backwards compatibility. It is also the basis for the libxslt library which processes XSLT It requires the presence of the libzvbi headers and library during configuration.

This filter forces the output display aspect ratio to be the same as the input, by changing the output sample aspect ratio. This section shows you details about your video live stream. Here's the first line:. This explains how video is being input. We can see here that the multimedia container matroska and the open video format webm is being used to capture video. Next, there's more information about the streams being input:. The audio stream is using the opus codec with a sample rate of Hz, which is standard for digital video files.

Mono means the audio is monophonic, meaning all sounds are mixed together in a single channel. And fltp is Planar Floating point format. It's a way of retrieving the sample format for your audio. Sample format information is used to determine how many bits will be in each audio sample.

The higher the number of bits per sample in a file, the more data that will be stored in the sample. You can learn more about sample format here: What is Sample Format? The video stream is using the h codec with a constrained baseline. The term constrained baseline describes the video stream profile. A profile determines the methods that an encoder can use to generate streams and also describe the methods a decoder is expected to perform in order to decode the stream later.

The term 'yuvp' describes the type of video signal you're using. YUV is a type of video signal where colors are split into three channels, one for luminance brightness which includes green, and two for chrominance color which include blue and red in separate channels. Normally, you'd notate this as It's telling you that the color channels have half the resolution of the luminance channel. If you want to learn a bit more about this way of representing color, see the api.

The term progressive means that the video is in a progressive format, which means that the TBR value coming up in our discussion is talking about the frame rate rather than the field rate.

There's confusion about both these terms. For the purpose of this walkthrough, let's keep things simple. A progressive video format will use frame rate to describe frames per second. An interlaced video format will use field rate. It takes two fields to show you one image. SAR stands for Sample Aspect Ratio, and it's showing you the ratio between the width and the height of each individual pixel - DAR stands for Display Aspect Ratio and that's the ratio between the width and height of a full image in your video.

In this case it's also In the final line, it closes out with "1k tbr, 1k tbn default. At the start of a video stream, TBR is low. In our stream, it's 1 at the very start. Sometimes if you have a video format that's interlaced, the TBR will be twice as high as expected, because there are two fields used to create an image. TBC is the time base for the codec for the stream.

Stream is our video stream. This means that the input stream is from the first file '0' and it is the second stream in that file '1. It's being mapped to h libx meaning that the output stream is going to be H. Stream is our audio stream. This means that the input stream is from the first file '0 and its the first stream in that file '0.

We're mapping to libx, so now we receive some details about how libx is configured for our stream. To understand CBR, we need to understand what is bitrate.

Bitrate is the amount of information or data stored per second in any media file such as audio or video. In technical terms, it is the rate of the data output of an encoder. For example, kbps as a bitrate means that kilobits of data is stored per second. However, it is not possible to have constant bitrate for all the frames because of the different frame types and encoding priorities. Hence, if the bitrate is constant over a pre-determined time-period, the video is called a CBR-encoded video.

In order to output a constant bitrate video, we would be using the following FFmpeg output options :. After this, we also need to specify the video encoder using the -c:v option. As mentioned earlier, we are going to use the libx encoder in all the three cases.