[FFmpeg-cvslog] doc/encoders.texi: update documentation for the native AAC encoder

Rostislav Pehlivanov git at videolan.org
Sat Dec 5 17:10:23 CET 2015


ffmpeg | branch: master | Rostislav Pehlivanov <atomnuker at gmail.com> | Sat Dec  5 14:43:17 2015 +0000| [e34e3619a2b5b6fb4b4d9e68504b528c168da868] | committer: Rostislav Pehlivanov

doc/encoders.texi: update documentation for the native AAC encoder

Since the next commit removes the experimental flag from the encoder
it's better to update the documentation which has been around in its
current form for as long as the encoder itself.

Signed-off-by: Rostislav Pehlivanov <atomnuker at gmail.com>

> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=e34e3619a2b5b6fb4b4d9e68504b528c168da868
---

 doc/encoders.texi |  145 +++++++++++++++++++++++++++++++++--------------------
 1 file changed, 91 insertions(+), 54 deletions(-)

diff --git a/doc/encoders.texi b/doc/encoders.texi
index ae74449..8c8280b 100644
--- a/doc/encoders.texi
+++ b/doc/encoders.texi
@@ -30,81 +30,119 @@ follows.
 
 Advanced Audio Coding (AAC) encoder.
 
-This encoder is an experimental FFmpeg-native AAC encoder. Currently only the
-low complexity (AAC-LC) profile is supported. To use this encoder, you must set
- at option{strict} option to @samp{experimental} or lower.
-
-As this encoder is experimental, unexpected behavior may exist from time to
-time. For a more stable AAC encoder, see @ref{libvo-aacenc}. However, be warned
-that it has a worse quality reported by some users.
-
- at c todo @ref{libaacplus}
-See also @ref{libfdk-aac-enc,,libfdk_aac} and @ref{libfaac}.
+This encoder is the default AAC encoder, natively implemented into FFmpeg. Its
+quality is on par or better than libfdk_aac at the default bitrate of 128kbps.
+This encoder also implements more options, profiles and samplerates than
+other encoders (with only the AAC-HE profile pending to be implemented) so this
+encoder has become the default and is the recommended choice.
 
 @subsection Options
 
 @table @option
 @item b
 Set bit rate in bits/s. Setting this automatically activates constant bit rate
-(CBR) mode.
+(CBR) mode. If this option is unspecified it is set to 128kbps.
 
 @item q
 Set quality for variable bit rate (VBR) mode. This option is valid only using
 the @command{ffmpeg} command-line tool. For library interface users, use
 @option{global_quality}.
 
- at item stereo_mode
-Set stereo encoding mode. Possible values:
-
- at table @samp
- at item auto
-Automatically selected by the encoder.
-
- at item ms_off
-Disable middle/side encoding. This is the default.
-
- at item ms_force
-Force middle/side encoding.
- at end table
+ at item cutoff
+Set cutoff frequency. If unspecified will allow the encoder to dynamically
+adjust the cutoff to improve clarity on low bitrates.
 
 @item aac_coder
 Set AAC encoder coding method. Possible values:
 
 @table @samp
- at item faac
-FAAC-inspired method.
-
-This method is a simplified reimplementation of the method used in FAAC, which
-sets thresholds proportional to the band energies, and then decreases all the
-thresholds with quantizer steps to find the appropriate quantization with
-distortion below threshold band by band.
-
-The quality of this method is comparable to the two loop searching method
-described below, but somewhat a little better and slower.
-
- at item anmr
-Average noise to mask ratio (ANMR) trellis-based solution.
-
-This has a theoretic best quality out of all the coding methods, but at the
-cost of the slowest speed.
-
 @item twoloop
 Two loop searching (TLS) method.
 
 This method first sets quantizers depending on band thresholds and then tries
 to find an optimal combination by adding or subtracting a specific value from
 all quantizers and adjusting some individual quantizer a little.
+Will tune itself based on whether aac_is/aac_ms/aac_pns are enabled.
+This is the default choice for a coder.
 
-This method produces similar quality with the FAAC method and is the default.
+ at item anmr
+Average noise to mask ratio (ANMR) trellis-based solution.
+
+This is an experimental coder which currently produces a lower quality, is more
+unstable and is slower than the default twoloop coder but has potential.
+Currently has no support for the @option{aac_is} or @option{aac_pns} options.
+Not currently recommended.
 
 @item fast
 Constant quantizer method.
 
 This method sets a constant quantizer for all bands. This is the fastest of all
-the methods, yet produces the worst quality.
+the methods and has no rate control or support for @option{aac_is} or
+ at option{aac_pns}.
+Not recommended.
 
 @end table
 
+ at item aac_ms
+Sets mid/side coding mode. The default value of "-1" will automatically use
+M/S with bands which will benefit from such coding. Can be forced for all bands
+using the value "1", which is mainly useful for debugging or disabled using "0".
+
+ at item aac_is
+Sets intensity stereo coding tool usage. By default, it's enabled and will
+automatically toggle IS for similar pairs of stereo bands if it's benefitial.
+Can be disabled for debugging by setting the value to "0".
+
+ at item aac_pns
+Uses perceptual noise substitution to replace low entropy high frequency bands
+with imperceivable white noise during the decoding process. By default, it's
+enabled, but can be disabled for debugging purposes by using "0".
+
+ at item aac_tns
+Enables the use of a multitap FIR filter which spans through the high frequency
+bands to hide quantization noise during the encoding process and is reverted
+by the decoder. As well as decreasing unpleasant artifacts in the high range
+this also reduces the entropy in the high bands and allows for more bits to
+be used by the mid-low bands. By default it's enabled but can be disabled for
+debugging by setting the option to "0".
+
+ at item aac_ltp
+Enables the use of the long term prediction extension which increases coding
+efficiency in very low bandwidth situations such as encoding of voice or
+solo piano music by extending constant harmonic peaks in bands throughout
+frames. This option is implied by profile:a aac_low and is incompatible with
+aac_pred. Use in conjunction with @option{-ar} to decrease the samplerate.
+
+ at item aac_pred
+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.
+This option implies the aac_main profile and is incompatible with aac_ltp.
+
+ at item profile
+Sets the encoding profile, possible values:
+
+ at table @samp
+ at item aac_low
+The default, AAC "Low-complexity" profile. Is the most compatible and produces
+decent quality.
+
+ at item mpeg2_aac_low
+This profile disables aac_is and aac_pns which were introduced with the MPEG4
+specifications and could cause incompatibility with very old devices. Will
+reduce quality and is therefore not recommended unless necessary.
+
+ at item aac_ltp
+Long term prediction profile, is enabled by and will enable the aac_ltp option.
+Introduced in MPEG4.
+
+ at item aac_ltp
+Main-type prediction profile, is enabled by and will enable the aac_pred option.
+Introduced in MPEG2.
+
+If this option is unspecified it is set to @samp{aac_low}.
+ at end table
 @end table
 
 @section ac3 and ac3_fixed
@@ -578,15 +616,13 @@ and slightly improves compression.
 
 libfaac AAC (Advanced Audio Coding) encoder wrapper.
 
-Requires the presence of the libfaac headers and library during
-configuration. You need to explicitly configure the build with
- at code{--enable-libfaac --enable-nonfree}.
-
-This encoder is considered to be of higher quality with respect to the
- at ref{aacenc,,the native experimental FFmpeg AAC encoder}.
+This encoder is of much lower quality and is more unstable than any other AAC
+encoders, so it's highly recommended to instead use other encoders, like
+ at ref{aacenc,,the native FFmpeg AAC encoder}.
 
-For more information see the libfaac project at
- at url{http://www.audiocoding.com/faac.html/}.
+This encoder also requires the presence of the libfaac headers and library
+during configuration. You need to explicitly configure the build with
+ at code{--enable-libfaac --enable-nonfree}.
 
 @subsection Options
 
@@ -694,9 +730,10 @@ configuration. You need to explicitly configure the build with
 so if you allow the use of GPL, you should configure with
 @code{--enable-gpl --enable-nonfree --enable-libfdk-aac}.
 
-This encoder is considered to be of higher quality with respect to
-both @ref{aacenc,,the native experimental FFmpeg AAC encoder} and
- at ref{libfaac}.
+This encoder is considered to produce output on par or worse at 128kbps to the
+ at ref{aacenc,,the native FFmpeg AAC encoder} but can often produce better
+sounding audio at identical or lower bitrates and has support for the
+AAC-HE profiles.
 
 VBR encoding, enabled through the @option{vbr} or @option{flags
 +qscale} options, is experimental and only works with some



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