Enables the use of a more traditional style of prediction where the spectral coefficients transmitted are replaced by the difference of the current coefficients minus the previous "predicted" coefficients.
In theory and sometimes in practice this can improve quality for low to mid bitrate audio. The default, AAC "Low-complexity" profile. Is the most compatible and produces decent quality. Introduced in MPEG4. Introduced in MPEG2. This does not mean that one is always faster, just that one or the other may be better suited to a particular system.
The AC-3 metadata options are used to set parameters that describe the audio, but in most cases do not affect the audio encoding itself. Some of the options do directly affect or influence the decoding and playback of the resulting bitstream, while others are just for informational purposes.
A few of the options will add bits to the output stream that could otherwise be used for audio data, and will thus affect the quality of the output. Those will be indicated accordingly with a note in the option list below. Allow Per-Frame Metadata. Specifies if the encoder should check for changing metadata for each frame.
Center Mix Level. The amount of gain the decoder should apply to the center channel when downmixing to stereo. This field will only be written to the bitstream if a center channel is present. The value is specified as a scale factor.
There are 3 valid values:. Surround Mix Level. The amount of gain the decoder should apply to the surround channel s when downmixing to stereo. This field will only be written to the bitstream if one or more surround channels are present. Audio Production Information is optional information describing the mixing environment. Either none or both of the fields are written to the bitstream. Mixing Level.
Specifies peak sound pressure level SPL in the production environment when the mix was mastered. Valid values are 80 to , or -1 for unknown or not indicated. The default value is -1, but that value cannot be used if the Audio Production Information is written to the bitstream.
Room Type. Describes the equalization used during the final mixing session at the studio or on the dubbing stage. A large room is a dubbing stage with the industry standard X-curve equalization; a small room has flat equalization.
Dialogue Normalization. This parameter determines a level shift during audio reproduction that sets the average volume of the dialogue to a preset level. The goal is to match volume level between program sources. A value of dB will result in no volume level change, relative to the source volume, during audio reproduction.
Valid values are whole numbers in the range to -1, with being the default. Dolby Surround Mode. Specifies whether the stereo signal uses Dolby Surround Pro Logic. This field will only be written to the bitstream if the audio stream is stereo. Original Bit Stream Indicator. Specifies whether this audio is from the original source and not a copy.
It is grouped into 2 parts. If any one parameter in a group is specified, all values in that group will be written to the bitstream. Default values are used for those that are written but have not been specified. Preferred Stereo Downmix Mode. Dolby Surround EX Mode. Indicates whether the stream uses Dolby Surround EX 7. Dolby Headphone Mode. Indicates whether the stream uses Dolby Headphone encoding multi-channel matrixed to 2. Stereo Rematrixing.
This option is enabled by default, and it is highly recommended that it be left as enabled except for testing purposes. Set lowpass cutoff frequency. If unspecified, the encoder selects a default determined by various other encoding parameters. These options are only valid for the floating-point encoder and do not exist for the fixed-point encoder due to the corresponding features not being implemented in fixed-point.
The per-channel high frequency information is sent with less accuracy in both the frequency and time domains. This allows more bits to be used for lower frequencies while preserving enough information to reconstruct the high frequencies. This option is enabled by default for the floating-point encoder and should generally be left as enabled except for testing purposes or to increase encoding speed.
Coupling Start Band. Sets the channel coupling start band, from 1 to If a value higher than the bandwidth is used, it will be reduced to 1 less than the coupling end band. If auto is used, the start band will be determined by the encoder based on the bit rate, sample rate, and channel layout. This option has no effect if channel coupling is disabled. Sets the compression level, which chooses defaults for many other options if they are not set explicitly.
Valid values are from 0 to 12, 5 is the default. Chooses if rice parameters are calculated exactly or approximately. Multi Dimensional Quantization. If set to 1 then a 2nd stage LPC algorithm is applied after the first stage to finetune the coefficients. This is quite slow and slightly improves compression. This is a native FFmpeg encoder for the Opus format. Currently its in development and only implements the CELT part of the codec. Its quality is usually worse and at best is equal to the libopus encoder.
If unspecified it uses the number of channels and the layout to make a good guess. Sets the maximum delay in milliseconds. Lower delays than 20ms will very quickly decrease quality. Requires the presence of the libfdk-aac headers and library during configuration.
You need to explicitly configure the build with --enable-libfdk-aac. The library is also incompatible with GPL, so if you allow the use of GPL, you should configure with --enable-gpl --enable-nonfree --enable-libfdk-aac. If the bitrate is not explicitly specified, it is automatically set to a suitable value depending on the selected profile. Note that VBR is implicitly enabled when the vbr value is positive. If not specified or explicitly set to 0 it will use a value automatically computed by the library.
Enable afterburner feature if set to 1, disabled if set to 0. This improves the quality but also the required processing power. Set VBR mode, from 1 to 5. Requires the presence of the libmp3lame headers and library during configuration. You need to explicitly configure the build with --enable-libmp3lame. See libshine for a fixed-point MP3 encoder, although with a lower quality.
The following options are supported by the libmp3lame wrapper. The lame -equivalent of the options are listed in parentheses. Set constant quality setting for VBR. Set algorithm quality. Valid arguments are integers in the range, with 0 meaning highest quality but slowest, and 9 meaning fastest while producing the worst quality. Enable use of bit reservoir when set to 1.
LAME has this enabled by default, but can be overridden by use --nores option. Enable the encoder to use ABR when set to 1. The lame --abr sets the target bitrate, while this options only tells FFmpeg to use ABR still relies on b to set bitrate.
Requires the presence of the libopencore-amrnb headers and library during configuration. You need to explicitly configure the build with --enable-libopencore-amrnb --enable-version3. This is a mono-only encoder. Set bitrate in bits per second. Only the following bitrates are supported, otherwise libavcodec will round to the nearest valid bitrate. Allow discontinuous transmission generate comfort noise when set to 1. The default value is 0 disabled. Most libopus options are modelled after the opusenc utility from opus-tools.
The following is an option mapping chart describing options supported by the libopus wrapper, and their opusenc -equivalent in parentheses. Set VBR mode. The FFmpeg vbr option has the following valid arguments, with the opusenc equivalent options in parentheses:. Set encoding algorithm complexity. Valid options are integers in the range. The default is Set maximum frame size, or duration of a frame in milliseconds. The argument must be exactly the following: 2.
Smaller frame sizes achieve lower latency but less quality at a given bitrate. Sizes greater than 20ms are only interesting at fairly low bitrates. The default is 20ms. Enable inband forward error correction. Default is disabled. Set cutoff bandwidth in Hz. The argument must be exactly one of the following: , , , , or , corresponding to narrowband, mediumband, wideband, super wideband, and fullband respectively.
The default is 0 cutoff disabled. Set channel mapping family to be used by the encoder. The default value of -1 uses mapping family 0 for mono and stereo inputs, and mapping family 1 otherwise. The default also disables the surround masking and LFE bandwidth optimzations in libopus, and requires that the input contains 8 channels or fewer.
Other values include 0 for mono and stereo, 1 for surround sound with masking and LFE bandwidth optimizations, and for independent streams with an unspecified channel layout.
If set to 0, disables the use of phase inversion for intensity stereo, improving the quality of mono downmixes, but slightly reducing normal stereo quality. The default is 1 phase inversion enabled. Shine is a fixed-point MP3 encoder. It has a far better performance on platforms without an FPU, e.
However, as it is more targeted on performance than quality, it is not on par with LAME and other production-grade encoders quality-wise.
Requires the presence of the libshine headers and library during configuration. You need to explicitly configure the build with --enable-libshine. The following options are supported by the libshine wrapper. The shineenc -equivalent of the options are listed in parentheses. Requires the presence of the libtwolame headers and library during configuration.
You need to explicitly configure the build with --enable-libtwolame. The following options are supported by the libtwolame wrapper. The twolame -equivalent options follow the FFmpeg ones and are in parentheses. Default value is k. Set quality for experimental VBR support. Maximum value range is from to 50, useful range is from to The higher the value, the better the quality.
Set psychoacoustic model to use in encoding. The argument must be an integer between -1 and 4, inclusive. The default value is 3. Requires the presence of the libvo-amrwbenc headers and library during configuration. You need to explicitly configure the build with --enable-libvo-amrwbenc --enable-version3.
Requires the presence of the libvorbisenc headers and library during configuration. You need to explicitly configure the build with --enable-libvorbis.
The following options are supported by the libvorbis wrapper. The oggenc -equivalent of the options are listed in parentheses. The value should be a float number in the range of Set cutoff bandwidth in Hz, a value of 0 disables cutoff. This only has effect on ABR mode. Set noise floor bias for impulse blocks. The value is a float number from A negative bias instructs the encoder to pay special attention to the crispness of transients in the encoded audio.
The tradeoff for better transient response is a higher bitrate. The following shared options are effective for this encoder. Only special notes about this particular encoder will be documented here. For the general meaning of the options, see the Codec Options chapter. For this encoder, the range for this option is between and Default is automatically decided based on sample rate and number of channel.
Set whether to enable optimization for mono. This option is only effective for non-mono streams. Available values:. If disabled, every frame will always have a palette written, even if there is a global palette supplied. Specifies the number of chunks to split frames into, between 1 and This permits multithreaded decoding of large frames, potentially at the cost of data-rate.
The encoder may modify this value to divide frames evenly. Specifies the second-stage compressor to use. If set to none , chunks will be limited to 1, as chunked uncompressed frames offer no benefit.
The native jpeg encoder is lossy by default, the -q:v option can be used to set the encoding quality. Lossless encoding can be selected with -pred 1. By default, when this option is not used, compression is done using the quality metric. This option allows for compression using compression ratio. The compression ratio for each level could be specified. The compression ratio of a layer l species the what ratio of total file size is contained in the first l layers.
This would compress the image to contain 3 layers, where the data contained in the first layer would be compressed by times, compressed by in the first two layers, and shall contain all data while using all 3 layers. Requires the presence of the rav1e headers and library during configuration. You need to explicitly configure the build with --enable-librav1e. See Transcode batches with multiple computers using Compressor.
Note: Job segmenting is not available when outputting an MPEG-4 audio file or when the Multi-pass checkbox is selected in the Video inspector. Default location: Choose an item from the pop-up menu to set the default save location for transcoded files. Format: Use this pop-up menu to specify whether the output includes video and audio, video only, or audio only.
For podcasting and digital music playback, choose audio only. If format is set to Audio, audio-only files are created with the. If format is set to Audio and Video or Video, files are created with the. Optimize for network use: Select this checkbox to create a file that will start playing after only a small portion of the file has been downloaded from the network.
Enhanced podcast: Select this checkbox to have Compressor embed podcasting information annotations, markers, and artwork into the output media file. See Work with metadata annotations in Compressor and Add markers using Compressor.
Sets the processing algorithm used to adjust the frame rate during transcoding. Select one of the following options:. Enter a value in the percentage field or choose a preset value from the adjacent pop-up menu with a down arrow. Set to: Sets the duration of the clip.
Enter a timecode duration in the field or click the arrows to increase or decrease the time. So source frames play at [frame rate] fps: Nondestructively changes the playback speed of the clip without discarding frames or creating new frames. For more information, see Retime video and audio using Compressor.
See Intro to supporting captions in Compressor. Specifies how metadata is embedded in the transcode. See Work with metadata annotations in Compressor. This is the default setting. Pass through source file metadata: Passes the existing metadata from the source file to the transcode. Ignores Job Annotations listed in the Job Inspector. Include metadata from the source file that cannot be displayed as a job annotation: Available when Use Job Annotations is selected.
Includes the metadata from the Job Annotations listed in the Job Inspector and passes the existing metadata from the source file to the transcode. Some video properties are enabled only for codecs that support them. Frame size: Use this pop-up menu to set the frame size resolution for the output file. There are four categories to choose from:. Pixel aspect ratio: Use this pop-up menu to set the pixel aspect ratio the ratio between the encoded width and the display width.
Frame rate: Use this pop-up menu to set the playback rate the number of images displayed per second for the output file. See Retiming options in Compressor. Field order: Use the pop-up menu to set the output scanning method either the field dominance or a conversion to progressive scanning. There are four options:. Automatic: Selects the most appropriate field order, based on the field order of the source and the capabilities of the selected codec.
Progressive: The video is displayed in complete frames with all lines sampled at the same instant in time. Top First: The video is interlaced and displayed as two separate interleaved fields. The field containing the top line even lines is sampled at an earlier instant in time than the field containing the bottom line odd lines.
This field order is commonly used for high-definition video and standard-definition PAL video. Bottom First: The video is interlaced and displayed as two separate interleaved fields. The field containing the bottom line odd lines is sampled at an earlier instant in time than the field containing the top line even lines.
This field order is commonly used for standard-definition NTSC video. Color space: Use this pop-up menu to convert the source media to a new color space , including wide color gamut. Choose Automatic to allow Compressor to choose the best color space based on the selected preset. You can also choose a manual setting to override the default. Choose Automatic to allow Compressor to choose the conversion method. Select a custom LUT to transform your video from one color space to another.
Encoder type: Use this pop-up menu to set the type of encoder. As mentioned above that H. And MP4 usually refers to a video format or format container. YouTube officially claims that the best video formats is MP4 with H.
So you have to deal with the video bitrate carefully to reduce quality loss as much as possible. And it provides integrated support for both video transmission and storage. You might or might noticed that it has been adopted in all most all multi-media fields:. MEPG A method of defining compression of both video and audio data.
It consists of several standards parts. MPEG-4 Part is one of them and used to define video compression. Will H. This is a question we'll ask every time new codecs come up but never settled down.
Some thought it might be replaced by HEVC. But as for the year , AVC still has its place. And Plex still only transcodes to H. In retrospect, plenty of video codecs are knocked out, such as H. At least now H. And it is reasonable to predict that when codecs wtih a higher compression ratio went viral on every application and device, the battlefield would turn to the patent pool.
But, as some talents have developed open source encoders x. Just wait and see whether it can survive in the next video codec revolution. Cecilia Hwung is the marketing manager of Digiarty Software and the editor-in-chief of VideoProc team. She pursues common progress with her team and expects to share more creative content and useful information to readers.
She has strong interest in copywriting and rich experience in video editing tips. Create cinematic videos and beyond. Learn More. VideoProc Converter One-stop video processing software. Convert, transcode, compress, download and record. VideoProc Converter Convert, transcode, compress, download and record.
Everything You Should Know about H. Part 1. What Is H. Part 2. The Evolution of H. The Advantages of H. It supports high definition videos including p and 4K 60fps. Part 4. How H. Inter and Intra Prediction The eternal purpose of video coding is to improve efficiency and save up bit rate as much as possible. I-frame contains the complete information of the image, it is coded independently of other non-I-frame pictures. P-frame contains differences relative to preceding frames.
B-frame contains differences relative to both preceding and following frames. The more frames between I-frames, the longer the GOP. Motion estimation: Frames are departed into macroblocks. The process of motion estimation is to search and find out the best matching of pixel blocks in the inter frame.
Motion compensation: It is a process of subtracting an inter prediction from the current macroblock after motion estimation. Transformation and Quantization Although inter and intro prediction removes large amounts of redundant data, the residual data can be further modified. Deblocking filter Since H. Entropy Coding Entropy coding is a lossless data compression scheme to convert image data into bitstreams that can be stored and transmitted. Part 5. How Does H. Part 7.
Frequently Asked Questions about H.
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