My android apps occasionally emit an alarmingly loud hissing sound,
presumably when some other sound was supposed to be played. This
suggests some underlying race condition in the sound manager; but
I wonder if anyone else has encountered this ?
My sounds are all prerecorded .wav files.
I would suggest switching to MP3 or AAC both of which are smaller and more deterministic in playback. As Scott mentioned there is probably a discrepancy in the bit depth which is less likely in an encoded file than in a raw file.
Related
Ok, so I found out that Blender has this really cool video-editing interface and I was beginning to love it. Until, I created this awesome project composition and when I exported the animation as a video file, the audio was out of sync :(.
Actual Problem
Audio is in-sync with video when the animation is played in Blender but is out-of-sync in the rendered video.
Solutions I tried out and failed
I used the 'Audio-Sync' option in the sequencer but that made no difference.
Then I thought that my scene audio frequency might have been an issue since it was initially 48kHz and my videos were at 24kHz, so I changed the scene audio frequency to 24kHz, this still failed to solve the issue.
Initially, I was combining videos with different frame rates and thought that might have been an issue (although animation played as expected in Blender), so I recreated the source videos to ensure all videos I was using in my project had the same frame rate, but this also did not work.
Someone online suggested exporting the video and audio separately and then combining them using a command-line tool like FFMPEG, this also failed.
What's really frustrating
This lag (audio is a few frames ahead of the video) is noticeable only in longer videos (>12 mins, my video is 1 hr long) suggesting a very small rendered rate difference between the video and the audio.
Also, note that the animation plays absolutely fine in Blender, so all I could figure out was that this was a rendering issue.
So if anyone figured this out please let me know. I am a noob in video/audio codecs so please forgive me if I used some incorrect nomenclature above.
I encountered this issue on OBS capture (a 13 minute clip) with Blender 2.93.3. OBS capture is constant framerate at 60 fps, I did try Handbrake conversion to 60 fps constant framerate also with no help. Workaround to solve the issue is to set Blender rendering fps to 59.94, sequencer shows audio track extending over video track but after render everything matches perfectly. Unfortunately you cannot edit the video in 59.94 fps mode, so you need to switch back to 60 fps for editing.
In case your video is 24 fps then use 23.98 fps preset and for 30 fps you can use the 29.97 fps preset.
May 2021. Blender v2.92.0 - I experienced the same as described out-of-sync problem with rendered videos that were over five minutes long. Source was as-is (3.6GB, 10mins) file from Canon EOS 5DMKII, which is an old camera, so pretty much any software can handle the encoding.
In Blender's preview mode everything looks in-sync. Audio and Video tracks are of the same length. I didn't even cut or merged any segments of the source video. I tried running rendering after a clean boot, gave Blender highest resource priority in Win10, allocated more memory to caching, etc. Source and output was on SSD. Rendered result still didn't match what GUI showed. Very frustrating, and a lot of wasted time.
What worked better for me is the following:
Change Video Codec to "FFmpeg video codec #1". This produces a lossless file that is about 27 times bigger (13.8GB for 10mins) than H.264 codec file (0.5GB). However, the audio remains in sync all the way through.
Use HandBrake open source video transcoder to convert FFmpeg file into H.264 (or H.265). End result produces a smallish-size file with A/V that is in sync.
This workaround is relatively painless and produces good-quality results because there's only a single lossy compression step. The time required to get to final file more than triples though. I believe the issue continues to be with the way H.264 rendering in Blender is implemented. I also experienced similar out-of-sync issues in ShotCut a year ago while working with cheap action cam H.265 files. I also found ShotCut to be less stable than Blender.
So after a lot of online searching, I did find an answer to fix this problem, but not in Blender. If you are like me and would like to use Blender for video editing and still get around the issue, then I found a workaround, but you need Shotcut for this. Shotcut is another great free and open-source video editor
Export the entire long video from Blender (the rendered video has desync issues as expected).
Open the video in Shotcut and detach the audio from it.
Use the audio properties to make very fine adjustments to the audio playback speed to suit your requirements (make fine adjustments until video and audio are in sync).
Follow the GIF attached.
(I am using a shorter video in the GIF but you get the idea)
Explanation
Blender has issues while rendering long videos and I noticed that the video is exported at 1.0x speed but the audio is sometimes faster (1.00400x or something like that) and hence the rendered video has audio not in sync with the video.
Another bad thing is that Blender does not really allow very fine playback speed adjustment just to the audio.
One trick is to adjust the pitch of the audio in Blender which in turn changes the playback speed but this is only allowed up to 2 decimal places (not what we want for long videos) and it makes the audio sound funny (since it actually changes the pitch).
Shotcut is a great tool that allows fine playback adjustment, and it also has a pitch compensation feature so that your pitch is kind of unaffected (since we don't want the characters to be sounding funny in our edited video).
Shotcut allows playback speed adjustment up to 6 decimal places.
I landed at this thread because of the same issue happening in a video that I have just finished. The "View animation CTRL F11" command starts an internal player that has sync issues with long videos. Opening the same video file on "Videos" in Fedora, it plays perfectly synchronized.
Recently, I discover that my tutorial videos could be seen at 1.5x playback speed without losses in quality (they are actually better to see, as I normally speak slowly). My problem is that if I change the speed of the video when using a video editor, like Kdenlive, the audio becomes distorted and turns into a mess (higher pitch, I believe).
How could I obtain the same quality as VLC "playback fast" and Youtube "playback speed 1.5" for the audio track? I'm a layman in audio/video editing, so I'm also satisfied with partial answers, like the identification of which terms I should search for in this case.
It might be better to take your audio track and use something like Sound Forge to automatically remove silence. Just be sure to add a pad to that (built into sound forge) otherwise the speech will sound way to chopped and fast.
Aside from that, you could also use Vegas to (then) chop the video to keep pace with your new speech rate. Vegas is a video editing program that is best for this kind of down and dirty editing.
I am uploading an audacity project with 2 tracks, the 1st one contains a "bitbit" sound resulted from Speex echo cancellation. I tried to remove the sound using Audacity noise cancellation, didn't work. Tried equalizer to cut off some high frequency sounds, worked but somehow degraded the sound quality. Please help how I can clear the noisy audio without significantly degrading quality.
If Audacity doesn't work, any C/C++ library can also be used.
Audacity Project
I think you should use the Noice Reduction effect from the effect menu.
I have built a source client using Portaudio and LAME which streams the microphone input to an Icecast server to be listened to online via the HTML5 tag. I have managed to (supposedly) get the quality of the stream to MP3 320kbps at 44.1kHz and am looking for a way to confirm this using tests and or benchmarks.
I have an indication that these stats are somewhat correct from looking at stream inspectors in software such as iTunes and VLC, but I am looking to get a more in-depth data set.
What I basically want is to be able to test how much of the original file is being lost over the stream and if or how much the quality changes depending on environmental conditions of the broadcaster or streamer.
Does anyone know of any tools, frameworks to get some hard numbers or representations of this data?
If VLC tells you the stream is 320kbit CBR, then it is.
It sounds like what you're looking for is a comparison of the actual audio content. This is highly subjective. MP3 is built to use features of how our hearing works to save bandwidth. For example, quiet sounds are masked by loud sounds. High frequencies are harder to hear and are simply rolled off.
You can compare the spectral analysis between the original PCM-sampled waveform and the MP3 decoded waveform, but this doesn't tell you how humans interpret that sound. For that, you would have to survey humans.
I searched many questions - but no one seems to be giving simplest, most uniform approach, hence please do not close as duplicate.
My requirement is simple: I have quiz app.
I want to include:
background music that plays continually - probably more than one
audio.
I need occassional sounds played at specific events - they
are very short in duration. Maybe 4-5 in number.
What sound format do I use? [aac etc]
How do I produce it? (optionally, get it from internet, if free)
What is the best approach to incorporate it? [audioplayback, openal etc)
Forgive me if this is quite stupid, but I am going very generic here and can't seem to find it.
Thanks for the help!
For sound format, use AAC or uncompressed 16-bit little endian in a CAF container (avoid mp3 since it's difficult to make it loop cleanly). You can convert using the command line tool 'afconvert':
Compressed:
afconvert -f caff -d aac sourcefile.wav destfile.caf
Uncompressed 16-bit:
afconvert -f caff -d LEI16 sourcefile.wav destfile.caf
For production, either record it yourself (using an audio program such as Audacity), get a professional to do it, or buy royalty free sounds/music.
To incorporate it, use AVAudioPlayer for music and OpenAL for sounds. OpenAL is difficult to use and doesn't decode compressed audio on its own, so you may want to use an audio library such as https://github.com/kstenerud/ObjectAL-for-iPhone