How do I "crank up"/increse the volume?

[tigerb]

Format : MPEG-4
Format profile : Base Media
Codec ID : isom
File size : 1.44 GiB
Duration : 46mn 11s
Overall bit rate : 4 471 Kbps
Writing application : Lavf54.63.104

Video
ID : 1
Format : AVC
Format/Info : Advanced Video Codec
Format profile : Baseline@L3.1
Format settings, CABAC : No
Format settings, ReFrames : 1 frame
Codec ID : avc1
Codec ID/Info : Advanced Video Coding
Duration : 46mn 11s
Bit rate : 4 300 Kbps
Width : 1 280 pixels
Height : 720 pixels
Display aspect ratio : 16:9
Frame rate mode : Constant
Frame rate : 50.000 fps
Color space : YUV
Chroma subsampling : 4:2:0
Bit depth : 8 bits
Scan type : Progressive
Bits/(Pixel*Frame) : 0.093
Stream size : 1.39 GiB (96%)
Writing library : x264 core 142 r2479 dd79a61
Encoding settings : cabac=0 / ref=1 / deblock=1:0:0 / analyse=0x1:0x111 / me=hex / subme=2 / psy=1 / psy_rd=1.00:0.00 / mixed_ref=0 / me_range=16 / chroma_me=1 / trellis=0 / 8x8dct=0 / cqm=0 / deadzone=21,11 / fast_pskip=1 / chroma_qp_offset=0 / threads=12 / lookahead_threads=4 / sliced_threads=0 / nr=0 / decimate=1 / interlaced=0 / bluray_compat=0 / constrained_intra=0 / bframes=0 / weightp=0 / keyint=500 / keyint_min=50 / scenecut=40 / intra_refresh=0 / rc_lookahead=10 / rc=abr / mbtree=1 / bitrate=4300 / ratetol=1.0 / qcomp=0.60 / qpmin=0 / qpmax=69 / qpstep=4 / vbv_maxrate=14000 / vbv_bufsize=14000 / nal_hrd=none / filler=0 / ip_ratio=1.40 / aq=1:1.00
Color primaries : BT.709
Transfer characteristics : BT.709
Matrix coefficients : BT.709
Color range : Limited

Audio
ID : 2
Format : AAC
Format/Info : Advanced Audio Codec
Format profile : LC
Codec ID : 40
Duration : 46mn 11s
Bit rate mode : Constant
Bit rate : 160 Kbps
Channel(s) : 2 channels
Channel positions : Front: L R
Sampling rate : 48.0 KHz
Compression mode : Lossy
Stream size : 52.9 MiB (4%)
Language : unk

All the soccer matches in the above audio format is driving me crazy, when playing them on my PC using MPC-HC/MPC-BE, they are OK, but when playing on my TV, the volume is just dead low. Normally when playing video clips on my TV, my volume level is between 7 - 12, but when playing video clips in the above audio format, even when I adjust the volume level to 35 - 40, the volume is still low.

What can I do?

Thanks

[videobruger]

Drop your MP4 file on AUdacity with FFMpeg library installed, adjust the volume, save as AAC and mux it into your MP4 with YAMB replacing the existing audio.

[Cornucopia]

I think it's the "adjust the volume" part that he's/she's/you're having trouble with.

You could Normalize, but I'm guessing that is NOT what will fix it.

Pretend this is a graph of your waveform:
......................................

That is just "silence" (with a minute amount of noise thrown in, as usual).

.o.o....o.....ooo....oo...oo..o...o.o.

This is what your waveform might be like if it has some stronger elements but not using the full dynamic range.

oOo0oooo0ooooo000oooo00ooo00oo0ooo0o0o

This is the same waveform, but Normalized. Notice the larger elements are larger still - up to the DR ceiling - but the small noise elements are ALSO larger. That's how it works.

.o.o....|.....ooo....oo...oo..|...o.o.

But if your waveform is like this, where there a short spikes of sound called "transients", it has already hit the DR ceiling and has nowhere to go, so Normalize cannot work.


You want to use an editor that has the capability to operate on dynamic range with DR filter/adjustment plugins, such as a LIMITER, which clamps down those transients, and maybe a Compressor, but also an Expander/NoiseGate, which adjusts the middle & lower sections, to give you something that looks maybe like this:

.O.0....I.....000....00...00..Iooo0.0.

To get this done right, you'll need to learn some audio principles.

Scott

[tigerb]

Drop your MP4 file on AUdacity with FFMpeg library installed, adjust the volume, save as AAC and mux it into your MP4 with YAMB replacing the existing audio.

Thanks videobruge, the volume already @ 98%, even after adjusting to the max @ 100%, not much difference, what else can I try?

[tigerb]

I think it's the "adjust the volume" part that he's/she's/you're having trouble with.

You could Normalize, but I'm guessing that is NOT what will fix it.

Pretend this is a graph of your waveform:
......................................

That is just "silence" (with a minute amount of noise thrown in, as usual).

.o.o....o.....ooo....oo...oo..o...o.o.

This is what your waveform might be like if it has some stronger elements but not using the full dynamic range.

oOo0oooo0ooooo000oooo00ooo00oo0ooo0o0o

This is the same waveform, but Normalized. Notice the larger elements are larger still - up to the DR ceiling - but the small noise elements are ALSO larger. That's how it works.

.o.o....|.....ooo....oo...oo..|...o.o.

But if your waveform is like this, where there a short spikes of sound called "transients", it has already hit the DR ceiling and has nowhere to go, so Normalize cannot work.


You want to use an editor that has the capability to operate on dynamic range with DR filter/adjustment plugins, such as a LIMITER, which clamps down those transients, and maybe a Compressor, but also an Expander/NoiseGate, which adjusts the middle & lower sections, to give you something that looks maybe like this:

.O.0....I.....000....00...00..Iooo0.0.

To get this done right, you'll need to learn some audio principles.

Scott

Hi Cornucopia, thanks.

This is the audio file, what do I do?

[hech54]

32bit? That must be an Audacity "thing"....another reason I use GoldWave...

[videobruger]

http://thebest3d.com/audacity/tutorials/make-louder/audacity-tutorial-make-louder-more...amplitude.html

[tigerb]

http://thebest3d.com/audacity/tutorials/make-louder/audacity-tutorial-make-louder-more...amplitude.html

Bingo videobruger, you are great!

Using the "Normalize filter" default option, I got great volume now.

Thank you!

[Detmek]

32bit? That must be an Audacity "thing"....another reason I use GoldWave...

I don't see a problem with 32-bit float. Source is AAC so it is decoded to 32-bit float as it should for editing. AAC, as most lossy codecs does not have native bitdepht.

[popopipo]

Drop your MP4 file on AUdacity with FFMpeg library installed, adjust the volume, save as AAC and mux it into your MP4 with YAMB replacing the existing audio.

Thanks for sharing the info. Very Useful. I had a similar problem with her.

[hello_hello]

32bit? That must be an Audacity "thing"....another reason I use GoldWave...

How does GoldWave import it??

[hello_hello]

There's another option for adjusting AAC volume that's easier, faster than Audacity and lossless (no demuxing, importing, normalising, exporting while re-encoding and remuxing), but you might need to became a little familiar with foobar2000 first. Once you are, the whole process is very quick and simple.

Foobar2000 is an audio player, but it will open MP4s containing video. Add one (or many) to a playlist, highlight them all if there's more than one, right click and select "ReplayGain/Scan Per File Track Gain". It's pretty fast and if you have a multi-core CPU it'll scan more than one at a time. When it's done, it'll offer to save the ReplayGain info to the files. Let it.
Next, right click again and choose "ReplayGain/Apply Track ReplayGain to File Content". Foobar2000 will physically adjust the volume of the AAC audio in each file (no re-encoding).

The process is not the same as normalising. It should be better. Normalising just increases the volume until the peaks are at maximum. If two tracks have the same average volume, but one has a much higher peak somewhere, after peak normalising the average volumes will be different. ReplayGain strives to adjust each track to the same volume, and it works very well.

I'm pretty sure the method used to adjust the audio volume this way can only be applied to MP3 and AAC audio. For AAC it needs to be in an MP4/M4A container, but you could adjust and remux it into another container later if need be. You can scan with ReplayGain and save the info for lots of formats and apply it when re-encoding, but the "adjusting" option only works for AAC in an MP4/M4A container or MP3. It's the same method used by MP3Gain to adjust MP3 volume. In fact MP3Gain will do the same thing if you update the exe with AACGain (instructions on the MP3Gain site) but foobar2000 tends to be much faster.

In case anyone's got this far reading my post and is interested in trying the foobar2000 method of adjusting volume, here's some extra info it might be worth knowing:
MP3 and AAC audio can store values above "maximum", or 0dB. When their volumes are increased, the peaks can therefore exceed 0dB. They're not clipped, there's no damage done, but they may get clipped on playback, depending on the hardware. I've never noticed peaks of a few db clip on playback, but it's possible (I'm referring to peak values such as +1.5dB or +3dB). There's on option buried in foobar2000's preferences "Advanced/Tools/ReplayGain Scanner" that prevents the audio being increased enough to cause clipping. I think it's unchecked by default.
There's another option for setting the target volume. 89dB is the standard for music (audio ripped from CDs etc) but it doesn't provide enough headroom for "soundtrack" audio. The standard target volume there is around 83dB. A little quieter, but it's pretty likely the volume can always be adjusted to 83dB without the "prevent clipping" option errrrr...... preventing it. I find having the target volume buried in preferences a little inconvenient as I sometimes forget to change it according to the type of audio I'm adjusting, but at least it's possible.

There's another program called R128Gain that also might be worth a look. It strives to do everything I've described above with different presets for scanning and adjusting. You need to download the full version of ffmpeg yourself to work with multiple file types, but I don't know if it's capable of adjusting without re-encoding etc. I've barely played with it.
And for the record..... the original ReplayGain algorithm is obsolete, in that the newer EBU R128 algorithm is apparently better (although I've not found them to differ by all that much 99% of the time), so the R128Gain GUI has two ReplayGain presets. "ReplayGain2" uses the EBU R128 algorithm. As does foobar2000's ReplayGain scanner. I don't think the original ReplayGain supported multi-channel audio either, whereas the new method does.

That turned into more typing than I thought. Maybe I should re-write it as a guide at some stage.