This commit removes the experimental flag from the native AAC Encoder
and thus makes it the default.
After a lot of work, done by myself and Claudio Freire, the quality of
this encoder rivals and surpasses libfdk_aac in some situations. The
encoder had instability issues earlier which prevented it from having
its experimental flag removed, however the last commit done by Claudio
removed the last known source of instability and solved a lot of
problems which were previously observed. The issues were caused by the
various coding tools interfering with the scalefactor indices. Thus,
with these problems solved, it should now be possible to declare this
encoder as the default and recommend that the users should use it
instead of others provided by external libraries, as it is both faster
and has a subjectively higher quality with selected tracks.
The encoder has still yet to be fine tuned for every possible audio file
type like music or voice, so it is hoped that with the experimental flag
removed the users should be able to provide feedback and make the
encoder better than the alternatives for every type of audio.
The documentation will be edited and commited with a later commit.
---
libavcodec/aacenc.c | 3 +--
1 file changed, 1 insertion(+), 2 deletions(-)
diff --git a/libavcodec/aacenc.c b/libavcodec/aacenc.c
index 971f8ab..2473a14 100644
--- a/libavcodec/aacenc.c
+++ b/libavcodec/aacenc.c
@@ -1039,8 +1039,7 @@ AVCodec ff_aac_encoder = {
.close = aac_encode_end,
.supported_samplerates = mpeg4audio_sample_rates,
.caps_internal = FF_CODEC_CAP_INIT_THREADSAFE,
- .capabilities = AV_CODEC_CAP_SMALL_LAST_FRAME | AV_CODEC_CAP_DELAY |
- AV_CODEC_CAP_EXPERIMENTAL,
+ .capabilities = AV_CODEC_CAP_SMALL_LAST_FRAME | AV_CODEC_CAP_DELAY,
.sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_FLTP,
AV_SAMPLE_FMT_NONE },
.priv_class = &aacenc_class,
--
2.6.2
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