diff options
author | Rodeo <[email protected]> | 2012-09-03 12:37:16 +0000 |
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committer | Rodeo <[email protected]> | 2012-09-03 12:37:16 +0000 |
commit | 8d9969febdcec59fee453888ed0bb6a64adf7679 (patch) | |
tree | dbe81e961c2ed4c0f3e38d44c5afa86cc475693a /gtk/src/audiohandler.h | |
parent | 56bf9b80b9b55ecaec9855addda41898261d26b1 (diff) |
Audio improvements.
New supported samplerates: 8, 11.025, 12, 16 kHz.
Now 8, 11.025, 12, 16, 22.05, 24, 42, 44.1, 48 Khz are supported.
Unsupported samplerates are sanitized to the closest samplerate for all encoders.
Samplerates < 32 kHz are now forbidden for AC3 encoding (sanitized to 32 kHz). Most AC3 decoders don't support such samplerates.
New upmixing: 3.0 (Front Left, Right & Center) can now be upmixed to 5.1 to preserve the center channel.
New mixdowns:
6.1 (Front Left, Right & Center, Surround Left, Right & Center, LFE)
7.1 (Front Left, Right & Center, Surround Left & Right, Rear Left & Right, LFE)
-> available to Vorbis & FLAC encoders for compatible input channel layouts
7.1 (Front Left, Right & Center, Front Left & Right of Center, Surround Left & Right, LFE)
-> available to AAC encoders (ca_aac, ca_haac, faac) for compatible input channel layouts
Mono (Left Only): Stereo to Mono by discarding the Right channel
Mono (Right Only): Stereo to Mono by discarding the Left channel
-> available to all encoders for non-Dolby Stereo input
The "6-channel discrete" mixdown becomes "5.1 Channels".
New bitrates: 960 - 1536 Kbps.
This lets users work around poor audio quality in crappy encoders by throwing more bits at them.
Bitrate limits have been re-worked and re-tested for all encoders.
git-svn-id: svn://svn.handbrake.fr/HandBrake/trunk@4930 b64f7644-9d1e-0410-96f1-a4d463321fa5
Diffstat (limited to 'gtk/src/audiohandler.h')
-rw-r--r-- | gtk/src/audiohandler.h | 1 |
1 files changed, 1 insertions, 0 deletions
diff --git a/gtk/src/audiohandler.h b/gtk/src/audiohandler.h index 0a3f98c3f..7bd300cc5 100644 --- a/gtk/src/audiohandler.h +++ b/gtk/src/audiohandler.h @@ -34,6 +34,7 @@ void ghb_set_audio(signal_user_data_t *ud, GValue *settings); const gchar* ghb_get_user_audio_lang(GValue *settings, gint titleindex, gint track); void ghb_audio_list_refresh_selected(signal_user_data_t *ud); gint ghb_select_audio_codec(gint mux, hb_audio_config_t *aconfig, gint acodec, gint fallback_acodec, gint copy_mask); +int ghb_select_fallback( GValue *settings, int mux, int acodec ); int ghb_get_copy_mask(GValue *settings); void ghb_audio_list_refresh(signal_user_data_t *ud); char * ghb_format_quality( const char *prefix, int codec, double quality ); |