Aften: A/52 audio encoder Version 0.0.8 (c) 2006-2007 Justin Ruggles, Prakash Punnoor, et al. usage: aften [options]options: CONSOLE OUTPUT OPTIONS [-h] Print out list of commandline options. [-longhelp] Print out commandline option details. [-version] Print out the build version. [-v #] Verbosity This controls the level of console output to stderr. 0 - Quiet Mode. No output to stderr. 1 - Shows a running average of encoding statistics. This is the default setting. 2 - Shows the statistics for each frame. ENCODING OPTIONS [-threads #] Number of threads Aften can use multiple threads to speed up encoding. By default, Aften uses one thread for each logical CPU your system has, but you can override this value. A value of 0 is the default and indicates that Aften should try to detect the number of CPUs. [-nosimd X] Comma-separated list of SIMD instruction sets not to use Aften will auto-detect available SIMD instruction sets for your CPU, so you shouldn't need to disable sets explicitly - unless for speed or debugging reasons. Available sets are mmx, sse, sse2, sse3 and altivec. No spaces are allowed between the sets and the commas. Example: -nosimd sse2,sse3 [-b #] CBR bitrate in kbps CBR mode is selected by default. This option allows for setting the fixed bitrate. The default bitrate depends on the number of channels (not including LFE). 1 = 96 kbps 2 = 192 kbps 3 = 256 kbps 4 = 384 kbps 5 = 448 kbps [-q #] VBR quality A value 0 to 1023 which corresponds to SNR offset, where q=240 equates to an SNR offset of 0. 240 is the default value. This scale will most likely be replaced in the future with a better quality measurement. [-fba #] Fast bit allocation Fast bit allocation is a less-accurate search method for CBR bit allocation. It only narrows down the SNR value to within 16 of the optimal value. The result is lower overall quality, but faster encoding. This may not give the same results each time when using parallel encoding. [-fes #] Fast exponent strategy decision By default, the exponent strategy for each channel in a frame is decided by finding the best choice out of 5 pre-defined sets of strategies. When this fast option is turned on, the same set is always used for every channel in every frame, which leads to generally lower quality but gives a significant speed increase. [-pad #] Start-of-stream padding The AC-3 format uses an overlap/add cycle for encoding each block. By default, Aften pads the delay buffer with a block of silence to avoid inaccurate encoding of the first frame of audio. If this behavior is not wanted, it can be disabled. The pad value can be a 1 (default) to use padding or 0 to not use padding. [-w #] Bandwidth The bandwidth setting corresponds to the high-frequency cutoff. Specifically, it sets the highest frequency bin which is encoded. The AC-3 format uses a 512-point MDCT which gives 256 frequency levels from 0 to 1/2 of the samplerate. The formula to give the number of coded frequency bins from bandwidth setting is: (w * 3) + 73, which gives a range of 73 to 253 I hope to replace this setting with one where the user specifies the actual cutoff frequency rather than the bandwidth code. There are 2 special values, -1 and -2. When -1 is used, Aften automatically selects what it thinks is an appropriate bandwidth. This is the default setting. When -2 is used, a bandwidth is chosen for each frame based on CBR frame size and a target quality of 240. Variable bandwidth cannot be used with VBR mode. [-wmin #] Minimum bandwidth For variable bandwidth mode (-2), this option sets the minimum value for the bandwidth code. This allows the user to avoid a harsh cutoff frequency by sacrificing general audio quality. The default value is 0. [-wmax #] Maximum bandwidth For variable bandwidth mode (-2), this option sets the maximum value for the bandwidth code. This can be used to speed up encoding by using a lower value than 60, which is the default. [-m #] Stereo rematrixing Using stereo rematrixing can increase quality by removing redundant information between the left and right channels. This technique is common in audio encoding, and is sometimes called mid/side encoding. When this setting is turned on, Aften adaptively turns rematrixing on or off for each of 4 frequency bands for each block. When this setting is turned off, rematrixing is not used for any blocks. The default value is 1. [-s #] Block switching The AC-3 format allows for 2 different types of MDCT transformations to translate from time-domain to frequency-domain. The default is a 512-point transform, which gives better frequency resolution. There is also a 256-point transform, which gives better time resolution. The specification gives a suggested method for determining when to use the 256-point transform. When block switching is turned on, Aften uses the spec method for selecting the 256-point MDCT. When it is turned off, only the 512-point MDCT is used, which is faster. Block switching is turned off by default. BITSTREAM INFO METADATA [-cmix #] Center mix level When three front channels are in use, this code indicates the nominal down mix level of the center channel with respect to the left and right channels. 0 = -3.0 dB (default) 1 = -4.5 dB 2 = -6.0 dB [-smix #] Surround mix level If surround channels are in use, this code indicates the nominal down mix level of the surround channels. 0 = -3 dB (default) 1 = -6 dB 2 = 0 [-dsur #] Dolby Surround mode When operating in the two channel mode, this code indicates whether or not the program has been encoded in Dolby Surround. This information is not used by the AC-3 decoder, but may be used by other portions of the audio reproduction equipment. 0 = not indicated (default) 1 = not Dolby surround encoded 2 = Dolby surround encoded DYNAMIC RANGE COMPRESSION AND DIALOG NORMALIZATION [-dynrng #] Dynamic Range Compression profile Dynamic Range Compression allows for the final output dynamic range to be limited without sacrificing quality. The full dynamic range audio is still encoded, but a code is given for each block which tells the decoder to adjust the output volume for that block. The encoder must analyze the input audio to determine the best way to compress the dynamic range based on the loudness and type of input (film, music, speech). 0 = Film Light 1 = Film Standard 2 = Music Light 3 = Music Standard 4 = Speech 5 = None (default) [-dnorm #] Dialog normalization [0 - 31] (default: 31) The dialog normalization value sets the average dialog level. The value is typically constant for a particular audio program. The decoder has a target output dialog level of -31dB, so if the dialog level is specified as being -31dB already (default), the output volume is not altered. Otherwise, the overall output volume is decreased so that the dialog level is adjusted down to -31dB. INPUT OPTIONS By default, information about the input file, such as the channel configuration and data size, is determined by the input file wav header. However, the basic WAVE format is limited in that it can only support up to 4 GB of data and cannot specify all channel layouts possible in the AC-3 format. The acmod and lfe options allow the user to explicitly select the desired channel layout. This only controls the interpretation of the input, so no downmixing or upmixing is done. The readtoeof option overrides the header and lets the user specify that Aften should keep reading data until the end-of-file. [-acmod #] Audio coding mode (overrides wav header) 0 = 1+1 (Ch1,Ch2) 1 = 1/0 (C) 2 = 2/0 (L,R) 3 = 3/0 (L,R,C) 4 = 2/1 (L,R,S) 5 = 3/1 (L,R,C,S) 6 = 2/2 (L,R,SL,SR) 7 = 3/2 (L,R,C,SL,SR) [-lfe #] Specify use of LFE channel (overrides wav header) 0 = LFE channel is not present 1 = LFE channel is present [-raw_fmt X] Raw audio input sample format (default: s16_le) This options specifies the sample format when using raw audio input. Using this option forces Aften to treat the input as raw audio. The choices for the pre-defined sample formats are: u8, s16_le, s16_be, s20_le, s20_be, s24_le, s24_be, s32_le, s32_be, float_le, float_be, double_le, double_be [-raw_sr #] Raw audio input sample rate (default: 48000) Using this option forces Aften to treat the input as raw audio. [-raw_ch #] Raw audio input channels (default: 2) Using this option forces Aften to treat the input as raw audio. [-chmap #] Channel mapping order of input audio Some programs create WAVE files which use a channel mapping other than the standard WAVE mapping. This option allows the user to specify if the input file uses WAVE, AC-3, or MPEG channel mapping. The MPEG channel mapping is used by DTS and by MPEG-2/4 formats such as MP2 and AAC. 0 = WAVE mapping (default) 1 = AC-3 mapping 2 = MPEG mapping [-readtoeof #] Read input WAVE audio data until the end-of-file. This overrides the data size in the WAVE header, and can be useful for streaming input or files larger than 4 GB. 0 = use data size in header (default) 1 = read data until end-of-file INPUT FILTERS [-bwfilter #] Specify use of the bandwidth low-pass filter The bandwidth low-pass filter pre-filters the input audio before converting to frequency-domain. This smooths the cutoff frequency transition for slightly better quality. 0 = do not apply filter (default) 1 = apply filter [-dcfilter #] Specify use of the DC high-pass filter The DC high-pass filter is listed as optional by the AC-3 specification. The implementation, as suggested, is a single pole filter at 3 Hz. 0 = do not apply filter (default) 1 = apply filter [-lfefilter #] Specify use of the LFE low-pass filter The LFE low-pass filter is recommended by the AC-3 specification. The cutoff is 120 Hz. The specification recommends an 8th order elliptic filter, but instead, Aften uses a Butterworth 2nd order cascaded direct form II filter. 0 = do not apply filter (default) 1 = apply filter ALTERNATE BIT STREAM SYNTAX The alternate bit stream syntax allows for encoding additional metadata by adding 1 or 2 extended bitstream info sections to each frame. [-xbsi1 #] Specify use of extended bitstream info 1 Extended bitstream info 1 contains the dmixmod, ltrtcmix, ltrtsmix, lorocmix, and lorosmix fields. If this option is turned on, all these values are written to the output stream. 0 = do not write xbsi1 1 = write xbsi1 [-dmixmod #] Preferred stereo downmix mode This code indicates the type of stereo downmix preferred by the mastering engineer, and can be optionally used, overridden, or ignored by the decoder. 0 = not indicated (default) 1 = Lt/Rt downmix preferred 2 = Lo/Ro downmix preferred The 4 mix level options below all use the same code table: 0 = +3.0 dB 1 = +1.5 dB 2 = 0.0 dB 3 = -1.5 dB 4 = -3.0 dB (default) 5 = -4.5 dB 6 = -6.0 dB 7 = -inf dB [-ltrtcmix #] Lt/Rt center mix level This code indicates the nominal down mix level of the center channel with respect to the left and right channels in an Lt/Rt downmix. [-ltrtsmix #] Lt/Rt surround mix level This code indicates the nominal down mix level of the surround channels with respect to the left and right channels in an Lt/Rt downmix. [-lorocmix #] Lo/Ro center mix level This code indicates the nominal down mix level of the center channel with respect to the left and right channels in an Lo/Ro downmix. [-lorosmix #] Lo/Ro surround mix level This code indicates the nominal down mix level of the surround channels with respect to the left and right channels in an Lo/Ro downmix. [-xbsi2 #] Specify use of extended bitstream info 2 Extended bitstream info 2 contains the dsurexmod, dheadphon, and adconvtyp fields. If this option is turned on, all these values are written to the output stream. These options are not used by the AC-3 decoder, but may be used by other portions of the audio reproduction equipment. 0 = do not write xbsi2 1 = write xbsi2 [-dsurexmod #] Dolby Surround EX mode This code indicates whether or not the program has been encoded in Dolby Surround EX. 0 = not indicated (default) 1 = Not Dolby Surround EX encoded 2 = Dolby Surround EX encoded [-dheadphon #] Dolby Headphone mode This code indicates whether or not the program has been Dolby Headphone-encoded. 0 = not indicated (default) 1 = Not Dolby Headphone encoded 2 = Dolby Headphone encoded [-adconvtyp #] A/D converter type This code indicates the type of A/D converter technology used to capture the PCM audio. 0 = Standard (default) 1 = HDCD