From 9e0a0cc94abc39a044085c2fc5dd89d5604534cc Mon Sep 17 00:00:00 2001 From: Jonas Smedegaard Date: Fri, 10 Oct 2014 11:26:23 +0200 Subject: Improve tuning of Vorbis and MP3, and rewrite help section on audio styles. --- localvideowebencode | 21 ++++++++++++++------- 1 file changed, 14 insertions(+), 7 deletions(-) (limited to 'localvideowebencode') diff --git a/localvideowebencode b/localvideowebencode index 33f4b2b..4eba2d1 100755 --- a/localvideowebencode +++ b/localvideowebencode @@ -31,6 +31,7 @@ # * Resolve flash player to use. # * Make choice of encoders configurable. # * Figure out how to apply application option when using opusenc. +# * Handle channels per-codec for low-bitrate joint stereo Opus speech. set -e @@ -74,9 +75,15 @@ html favoring open formats with optional non-JavaScript Flash fallback. h264 mp4 MPEG-4 H.264 AAC (default: webm,vp9,mp4) --audio Audio style: - [channels] [Vorbis] [Opus] [AAC] - music max 2 64k 64k audio 96k - speech 1 48k 32k voip 64k + [channels] [bitrate per channel] + music max. 2 ~48k-~64k + speech 1 ~32k-~64k + silence 0 + (default: channel count of first input, ~48k-~64k) + --audio Audio style: + [channels] [bitrate per channel] + music max 2 ~48k - ~64k + speech 1 ~32k - ~64k silence 0 (default: none - use channel count of first input) --audioprefilter Add audio filter before loudness @@ -366,7 +373,7 @@ fi # default per-codec-channel bitrates quality_vorbis=3 bitrate_opus=48 -bitrate_mp3=64 +quality_lame=6 bitrate_aac=64 case "$audio" in @@ -375,9 +382,9 @@ case "$audio" in ;; speech) mono=yes - quality_vorbis=1 + quality_vorbis=2 bitrate_opus=32 - bitrate_mp3=48 + quality_lame=7 opusapp=voip ;; silence) @@ -462,7 +469,7 @@ _oggenc_vorbis="$_oggenc_downmix -q $quality_vorbis" _melt_opus="$_melt_downmix acodec=libopus ab=$((channels*bitrate_opus))k${opusapp:+ application=$opusapp}" _avconv_opus="$_avconv_downmix -c:a libopus -b:a $((channels*bitrate_opus))k${opusapp:+ -application $opusapp}" _opusenc_opus="$_opusenc_downmix --bitrate $((channels*bitrate_opus))" -_melt_mp3="$_melt_downmix acodec=libmp3lame ab=$((channels*bitrate_mp3))k" +_melt_mp3="$_melt_downmix acodec=libmp3lame aq=$quality_lame" _melt_aac="$_melt_downmix acodec=aac ab=$((channels*bitrate_aac))k" # container options -- cgit v1.2.3