[FFmpeg-cvslog] aacenc: use the fast coder as the default

[Rostislav Pehlivanov]

ffmpeg | branch: master | Rostislav Pehlivanov <atomnuker at gmail.com> | Sat Jan 13 11:46:29 2018 +0000| [fcb681ac3edd708f26c3b5014c06e32fd2723887] | committer: Rostislav Pehlivanov

aacenc: use the fast coder as the default

The twoloop coder sounds decent at low bitrates, however at higher bitrates
it sounds worse than the fast coder (which used to be the old twoloop coder
before October 2015) and needs quite a lot more CPU.
Change the default to fast. It has been well tested and has had little changes
over the years so its been confirmed to be quite stable.
Also change its description (not valid for more than a year) and the
documentation.

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

> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=fcb681ac3edd708f26c3b5014c06e32fd2723887
---

 doc/encoders.texi   |  9 ++++-----
 libavcodec/aacenc.c |  4 ++--
 tests/fate/aac.mak  | 20 ++++++++++----------
 3 files changed, 16 insertions(+), 17 deletions(-)

diff --git a/doc/encoders.texi b/doc/encoders.texi
index 88ef8f9b23..6a410a8cb6 100644
--- a/doc/encoders.texi
+++ b/doc/encoders.texi
@@ -64,7 +64,6 @@ 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 @option{aac_is}, @option{aac_ms} and @option{aac_pns}
 are enabled.
-This is the default choice for a coder.
 
 @item anmr
 Average noise to mask ratio (ANMR) trellis-based solution.
@@ -77,10 +76,10 @@ 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 and has no rate control or support for @option{aac_is} or
- at option{aac_pns}.
-Not recommended.
+Uses a cheaper version of twoloop algorithm that doesn't try to do as many
+clever adjustments. Worse with low bitrates (less than 64kbps), but is better
+and much faster at higher bitrates.
+This is the default choice for a coder
 
 @end table
 
diff --git a/libavcodec/aacenc.c b/libavcodec/aacenc.c
index fa7932d42d..6d94c76905 100644
--- a/libavcodec/aacenc.c
+++ b/libavcodec/aacenc.c
@@ -1118,10 +1118,10 @@ fail:
 
 #define AACENC_FLAGS AV_OPT_FLAG_ENCODING_PARAM | AV_OPT_FLAG_AUDIO_PARAM
 static const AVOption aacenc_options[] = {
-    {"aac_coder", "Coding algorithm", offsetof(AACEncContext, options.coder), AV_OPT_TYPE_INT, {.i64 = AAC_CODER_TWOLOOP}, 0, AAC_CODER_NB-1, AACENC_FLAGS, "coder"},
+    {"aac_coder", "Coding algorithm", offsetof(AACEncContext, options.coder), AV_OPT_TYPE_INT, {.i64 = AAC_CODER_FAST}, 0, AAC_CODER_NB-1, AACENC_FLAGS, "coder"},
         {"anmr",     "ANMR method",               0, AV_OPT_TYPE_CONST, {.i64 = AAC_CODER_ANMR},    INT_MIN, INT_MAX, AACENC_FLAGS, "coder"},
         {"twoloop",  "Two loop searching method", 0, AV_OPT_TYPE_CONST, {.i64 = AAC_CODER_TWOLOOP}, INT_MIN, INT_MAX, AACENC_FLAGS, "coder"},
-        {"fast",     "Constant quantizer",        0, AV_OPT_TYPE_CONST, {.i64 = AAC_CODER_FAST},    INT_MIN, INT_MAX, AACENC_FLAGS, "coder"},
+        {"fast",     "Default fast search",       0, AV_OPT_TYPE_CONST, {.i64 = AAC_CODER_FAST},    INT_MIN, INT_MAX, AACENC_FLAGS, "coder"},
     {"aac_ms", "Force M/S stereo coding", offsetof(AACEncContext, options.mid_side), AV_OPT_TYPE_BOOL, {.i64 = -1}, -1, 1, AACENC_FLAGS},
     {"aac_is", "Intensity stereo coding", offsetof(AACEncContext, options.intensity_stereo), AV_OPT_TYPE_BOOL, {.i64 = 1}, -1, 1, AACENC_FLAGS},
     {"aac_pns", "Perceptual noise substitution", offsetof(AACEncContext, options.pns), AV_OPT_TYPE_BOOL, {.i64 = 1}, -1, 1, AACENC_FLAGS},
diff --git a/tests/fate/aac.mak b/tests/fate/aac.mak
index e8cbcef54d..5bc6779c39 100644
--- a/tests/fate/aac.mak
+++ b/tests/fate/aac.mak
@@ -154,7 +154,7 @@ fate-aac-aref-encode: CMD = enc_dec_pcm adts wav s16le $(REF) -c:a aac -aac_is 0
 fate-aac-aref-encode: CMP = stddev
 fate-aac-aref-encode: REF = ./tests/data/asynth-44100-2.wav
 fate-aac-aref-encode: CMP_SHIFT = -4096
-fate-aac-aref-encode: CMP_TARGET = 669
+fate-aac-aref-encode: CMP_TARGET = 596
 fate-aac-aref-encode: SIZE_TOLERANCE = 2464
 fate-aac-aref-encode: FUZZ = 89
 
@@ -163,7 +163,7 @@ fate-aac-ln-encode: CMD = enc_dec_pcm adts wav s16le $(TARGET_SAMPLES)/audio-ref
 fate-aac-ln-encode: CMP = stddev
 fate-aac-ln-encode: REF = $(SAMPLES)/audio-reference/luckynight_2ch_44kHz_s16.wav
 fate-aac-ln-encode: CMP_SHIFT = -4096
-fate-aac-ln-encode: CMP_TARGET = 61
+fate-aac-ln-encode: CMP_TARGET = 72
 fate-aac-ln-encode: SIZE_TOLERANCE = 3560
 fate-aac-ln-encode: FUZZ = 30
 
@@ -172,7 +172,7 @@ fate-aac-ln-encode-128k: CMD = enc_dec_pcm adts wav s16le $(TARGET_SAMPLES)/audi
 fate-aac-ln-encode-128k: CMP = stddev
 fate-aac-ln-encode-128k: REF = $(SAMPLES)/audio-reference/luckynight_2ch_44kHz_s16.wav
 fate-aac-ln-encode-128k: CMP_SHIFT = -4096
-fate-aac-ln-encode-128k: CMP_TARGET = 800
+fate-aac-ln-encode-128k: CMP_TARGET = 622
 fate-aac-ln-encode-128k: SIZE_TOLERANCE = 3560
 fate-aac-ln-encode-128k: FUZZ = 5
 
@@ -181,7 +181,7 @@ fate-aac-pns-encode: CMD = enc_dec_pcm adts wav s16le $(TARGET_SAMPLES)/audio-re
 fate-aac-pns-encode: CMP = stddev
 fate-aac-pns-encode: REF = $(SAMPLES)/audio-reference/luckynight_2ch_44kHz_s16.wav
 fate-aac-pns-encode: CMP_SHIFT = -4096
-fate-aac-pns-encode: CMP_TARGET = 616
+fate-aac-pns-encode: CMP_TARGET = 655
 fate-aac-pns-encode: SIZE_TOLERANCE = 3560
 fate-aac-pns-encode: FUZZ = 74
 
@@ -190,7 +190,7 @@ fate-aac-tns-encode: CMD = enc_dec_pcm adts wav s16le $(TARGET_SAMPLES)/audio-re
 fate-aac-tns-encode: CMP = stddev
 fate-aac-tns-encode: REF = $(SAMPLES)/audio-reference/luckynight_2ch_44kHz_s16.wav
 fate-aac-tns-encode: CMP_SHIFT = -4096
-fate-aac-tns-encode: CMP_TARGET = 817
+fate-aac-tns-encode: CMP_TARGET = 637
 fate-aac-tns-encode: FUZZ = 7
 fate-aac-tns-encode: SIZE_TOLERANCE = 3560
 
@@ -199,7 +199,7 @@ fate-aac-is-encode: CMD = enc_dec_pcm adts wav s16le $(TARGET_SAMPLES)/audio-ref
 fate-aac-is-encode: CMP = stddev
 fate-aac-is-encode: REF = $(SAMPLES)/audio-reference/luckynight_2ch_44kHz_s16.wav
 fate-aac-is-encode: CMP_SHIFT = -4096
-fate-aac-is-encode: CMP_TARGET = 615
+fate-aac-is-encode: CMP_TARGET = 514
 fate-aac-is-encode: SIZE_TOLERANCE = 3560
 fate-aac-is-encode: FUZZ = 10
 
@@ -208,7 +208,7 @@ fate-aac-ms-encode: CMD = enc_dec_pcm adts wav s16le $(TARGET_SAMPLES)/audio-ref
 fate-aac-ms-encode: CMP = stddev
 fate-aac-ms-encode: REF = $(SAMPLES)/audio-reference/luckynight_2ch_44kHz_s16.wav
 fate-aac-ms-encode: CMP_SHIFT = -4096
-fate-aac-ms-encode: CMP_TARGET = 675
+fate-aac-ms-encode: CMP_TARGET = 558
 fate-aac-ms-encode: SIZE_TOLERANCE = 3560
 fate-aac-ms-encode: FUZZ = 15
 
@@ -217,7 +217,7 @@ fate-aac-ltp-encode: CMD = enc_dec_pcm adts wav s16le $(TARGET_SAMPLES)/audio-re
 fate-aac-ltp-encode: CMP = stddev
 fate-aac-ltp-encode: REF = $(SAMPLES)/audio-reference/luckynight_2ch_44kHz_s16.wav
 fate-aac-ltp-encode: CMP_SHIFT = -4096
-fate-aac-ltp-encode: CMP_TARGET = 1270
+fate-aac-ltp-encode: CMP_TARGET = 1207
 fate-aac-ltp-encode: SIZE_TOLERANCE = 3560
 fate-aac-ltp-encode: FUZZ = 17
 
@@ -227,7 +227,7 @@ fate-aac-yoraw-encode: CMD = enc_dec_pcm adts wav s16le $(TARGET_SAMPLES)/audio-
 fate-aac-yoraw-encode: CMP = stddev
 fate-aac-yoraw-encode: REF = $(SAMPLES)/audio-reference/yo.raw-short.wav
 fate-aac-yoraw-encode: CMP_SHIFT = -12288
-fate-aac-yoraw-encode: CMP_TARGET = 259
+fate-aac-yoraw-encode: CMP_TARGET = 226
 fate-aac-yoraw-encode: SIZE_TOLERANCE = 3560
 fate-aac-yoraw-encode: FUZZ = 17
 
@@ -237,7 +237,7 @@ fate-aac-pred-encode: CMD = enc_dec_pcm adts wav s16le $(TARGET_SAMPLES)/audio-r
 fate-aac-pred-encode: CMP = stddev
 fate-aac-pred-encode: REF = $(SAMPLES)/audio-reference/luckynight_2ch_44kHz_s16.wav
 fate-aac-pred-encode: CMP_SHIFT = -4096
-fate-aac-pred-encode: CMP_TARGET = 841
+fate-aac-pred-encode: CMP_TARGET = 662
 fate-aac-pred-encode: FUZZ = 12
 fate-aac-pred-encode: SIZE_TOLERANCE = 3560