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B 2 Mp4

Adds arguments from a playlist request to URIs of fragments.This may be useful for performing client authorization at the moment ofrequesting a fragment, or when protecting an HLS stream with thengx_http_secure_link_modulemodule.

B 2 mp4

If an HLS stream is protected with thengx_http_secure_link_modulemodule, $uri should not be used in thesecure_link_md5expression because this will cause errors when requesting the fragments.Base URI should be usedinstead of $uri($hls_uri in the example):

During metadata processing, a larger buffer may become necessary.Its size cannot exceed the specified size,or else nginx will return the server error500 (Internal Server Error),and log the following message:

FFmpeg supports two AAC-LC encoders (aac and libfdk_aac) and one HE-AAC (v1/2) encoder (libfdk_aac). The license of libfdk_aac is not compatible with GPL, so the GPL does not permit distribution of binaries containing incompatible code when GPL-licensed code is also included. Therefore this encoder have been designated as "non-free", and you cannot download a pre-built ffmpeg that supports it. This can be resolved by compiling ffmpeg yourself.

The Fraunhofer FDK AAC codec library. This is currently the highest-quality AAC encoder available with ffmpeg. Requires ffmpeg to be configured with --enable-libfdk-aac (and additionally --enable-nonfree if you're also using --enable-gpl).

These settings target a specific bit rate, with less variation between samples. It gives you greater control over file size, and it is compatible with the HE-AAC profile. As a rule of thumb, for audible transparency, use 64 kBit/s for each channel (so 128 kBit/s for stereo, 384 kBit/s for 5.1 surround sound).

Convert the video with libx264, with a target of fitting a 90-minute movie on a 700 MB (=5734400 KB) CD-ROM, mixing the audio down to two channels (Windows users should use NUL rather than /dev/null and ^ instead of \):

This is a pair of AAC profiles tailored for low bit rates (version 1 and version 2). HE-AAC version 1 is suited for bit rates below 64kb/s (for stereo audio) down to about 48 kb/s, while HE-AAC version 2 is suited for bit rates as low as 32 kb/s (again, for stereo).

Unfortunately, many devices that can play AAC-LC (the default profile for libfdk_aac) simply cannot play either version of HE-AAC, so this is not recommended for surround sound audio, which normally needs to be compatible with such hardware players. If you are only going to play it on your computer, or you are sure that your hardware player supports HE-AAC, you can aim for a bit rate of 160kb/s for version 1, or 128kb/s for version 2. As always, experiment to see what works for your ears.

The native FFmpeg AAC encoder. This is currently the second highest-quality AAC encoder available in FFmpeg and does not require an external library like the other AAC encoders described here. This is the default AAC encoder.

By default the MP4 muxer writes the 'moov' atom after the audio stream ('mdat' atom) at the end of the file. This results in the user requiring to download the file completely before playback can occur. Relocating this moov atom to the beginning of the file can facilitate playback before the file is completely downloaded by the client.

To install FFmpeg with support for libvpx-vp9, look at the Compilation Guides and compile FFmpeg with the --enable-libvpx option. Note that for 10-bit encoding, you need to set the --enable-vp9-highbitdepth configuration option.

libvpx-vp9 offers a simple variable bitrate (VBR) mode by default. This is also sometimes called "Average Bitrate" or "Target Bitrate". In this mode, it will simply try to reach the specified bit rate on average, e.g. 2 MBit/s.

There are two different 2-pass encoding methods available in libvpx-vp9: a conventional mode for targeting an average bitrate, and a two-pass constant quality mode that uses the more contemporary CRF-style approach for the final pass to achieve a certain perceptual quality level while still gaining the aforementioned compression benefits by also doing a first pass.

For libvpx-vp9, the traditional wisdom of speeding up the first pass by using a faster encoding speed setting does not apply; -speed values from 0 to 4 result in the same speed for the first pass and yield the exact same results in the final encode, whereas any speed setting above 4 results in the first pass utilising only a single core, slowing things down significantly. Therefore the -speed switch can be entirely omitted from the first pass, since the default value of 1 will result in fast speed.

In addition to the "default" two-pass modes, there's a constant quality (CQ) mode (similar to CRF in the x264 encoder) that targets a certain perceptual quality level while only using a single pass. While using this single-pass mode will result in less efficient compression due to libvpx's preference for 2-pass encoding, this mode may still be useful if the extra time required for the first pass and the additional CPU cycles used for better compression in 2-pass mode aren't worth it for your use case.

To trigger this mode, you must use a combination of -crf and -b:v 0. Note that -b:v MUST be 0. Setting it to anything higher or omitting it entirely will instead invoke the Constrained Quality mode.

libvpx-vp9 also has a constrained quality (CQ) mode that will ensure that a constant (perceptual) quality is reached while keeping the bitrate below a specified upper bound or within a certain bound. While the caveats of single-pass encoding mentioned above stil apply, this method can still be useful for bulk encoding videos in a generally consistent fashion.

The quality is determined by the -crf, and the bitrate limit by the -b:v where the bitrate MUST be non-zero. Note: for videos that are "easy" to encode this mode behaves exactly like the Constant Quality mode and actual bitrate may be lower than specified bitrate while for videos that are "hard" to encode the quality will be bounded by the maximum bitrate and will behave like the Variable Bitrate mode.

Like most other encoders, libvpx offers a constant bitrate (CBR) encoding mode as well, which tries to encode the video in such a way that an average bitrate is reached. This doesn't mean that every frame is encoded with the same amount of bits (since it would harm quality), but the bitrate will be very constrained. You should use constant bitrate encoding if you need your video files to have a certain size, or if you're streaming the videos over a channel that only allows a certain bit rate. Generally though, using constrained quality is the recommended option in this case.

When the deadline/quality parameter is good or best, values for -cpu-used can be set between 0 and 5. The default is 0. Using 1 or 2 will increase encoding speed at the expense of having some impact on quality and rate control accuracy. 4 or 5 will turn off rate distortion optimization, having even more of an impact on quality.

With version 1.7.0, libvpx added support for row based multithreading which greatly enhances the number of threads the encoder can utilise. This improves encoding speed significantly on systems that are otherwise underutilised when encoding VP9. Since the amount of additional threads depends on the number of tiles, which itself depends on the video resolution, encoding higher resolution videos will see a larger performance improvement.

FFmpeg added support for row based multithreading in version 3.4, released on January 25th, 2018. As of libvpx version 1.7.0 this multithreading enhancement is not enabled by default and needs be manually activated with the -row-mt 1 switch.

MP4, aka MPEG-4, is a standard that defines the container format for compressed digital audio and visual data. In addition to video and audio, the MP4 format container also holds still images and subtitles.

The relationship between mdat and moov boxes might be hard to grasp. Stay with me, it will become clear as we dive into the moov box. Though let me start by addressing some confusion you might have on this topic.

At first I thought this meant there would be 3 (GPMF) and 2 (CAMM) distinct streams of data in these videos. This is wrong. All the raw data for the streams is together in binary found in the mdat box. ffprobe knows there are 3 and 2 streams (types of data) in each video because of the description (metadata) of the mdat box nested in the moov box.

In the case of video and audio traks the stsd box is where describes the encoding (e.g H265) using the boxes. I mention this as it might be helpful to think as CAMM or GPMF in this way (as a sort of codec).

These telemetry trak boxes are not unique to each codec (h265, camm, gpmf, etc), and the data each box contains is indistinguishable, regardless of the codec. I will attempt to explain that in the next post.

ffmpeg is a very fast video and audio converter that can also grab froma live audio/video source. It can also convert between arbitrary samplerates and resize video on the fly with a high quality polyphase filter.

ffmpeg reads from an arbitrary number of input "files" (which can be regularfiles, pipes, network streams, grabbing devices, etc.), specified by the-i option, and writes to an arbitrary number of output "files", which arespecified by a plain output url. Anything found on the command line whichcannot be interpreted as an option is considered to be an output url.

Each input or output url can, in principle, contain any number of streams ofdifferent types (video/audio/subtitle/attachment/data). The allowed number and/ortypes of streams may be limited by the container format. Selecting whichstreams from which inputs will go into which output is either done automaticallyor with the -map option (see the Stream selection chapter).

To refer to input files in options, you must use their indices (0-based). E.g.the first input file is 0, the second is 1, etc. Similarly, streamswithin a file are referred to by their indices. E.g. 2:3 refers to thefourth stream in the third input file. Also see the Stream specifiers chapter. 041b061a72

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