Is there a way to normalize the audio output of video files played with MediaPlayer?
I saw the AutomaticGainControl class but it's only for input (mic in)...
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Hi all!
Now I'm working on small audio library for PocketPC. Library using waveform audio interface (waveIn* and waveOut* functions). I have big problem with full duplex audio. Whem I'm open audio device for input, the sound from microphone is playing on the speaker. If I start playing some another sounds - they are mixing with input sounds. I want to separate input audio stream from output audio stream. Device doesn't have mixer, mixerGetNumDevs returnes 0, and mixerOpen also fails.
I have just the same problem. One workaround is to lower the sound volume level, the second is to use some echo canceller. The only echo canceller source code I have uses lots of floating point functions, so it is extremely slow on Pocket.
Is there a way to change our audio output to 24-bit 192khz audio output? I believe it can be changed from audio_policy.conf file have anyone tried that? Or is there another way to do so?
Has anyone noticed any issues with audio lag? When using MAME and MD.emu I see a delay between the action on screen and the sound effects, possibly 150-200ms.
I know my TV introduces a delay when processing sound in AC-3 (I had to set my Blu-ray player to PCM audio to stop this). Is the MOJO sending audio in AC-3 by default, and if so is it possible to force the MOJO to send PCM audio via HDMI?
I'm only connected to a stereo amp, so have no need for surround audio.
So is nobody else getting any audio lag? This is really frustrating me, to the point where I can't use it for what I intended (accurate emulation) and I'm considering selling it. Shame, as it has so much promise.
I just plugged in a cheap pair of headphones and wow, the sound is amazing using the hifi dac. I don't know much about it but I have a much better pair of Sony bluetooth headphones, is there any work around to use the hifi with bluetooth?
I don't believe so. I could be wrong but I very much doubt it.
Sent from my LG-LS997 using Tapatalk
This is a post from reddit user /u/codefyre explaining why the DAC can't be used over bluetooth (this was related to the v10, but applies to the v20 as well):
Not possible. Nutshell explanation:
All audio, the audio you hear with your actual ear, is analog. Sound is an analog vibration through the air.
The audio recordings in your phone are digital. MP3, MP4, whatever. Unless you're listening to an old record or cassette recording, you're playback source is digital.
The DAC is a Digital Analog Converter. A high quality DAC converts high quality digital audio into high quality analog audio. A higher quality DAC generates a better reconstruction of the original analog signal. Because wired headphones are analog, they simply transform the analog electrical signal into analog speaker vibrations.
If you think about this for a moment, you should understand why it can't be used with Bluetooth. BT is a digital protocol, so the connection between your phone and your Bluetooth receiver is digital. Bluetooth receivers typically perform an on-chip conversion of the digital signal to an analog output during playback. Some high quality BT speakers have onboard 24 bit and 32 bit DAC's to improve the fidelity of their audio output.
So, if the V10 allowed the DAC to be used with Bluetooth, your signal would be Digital Source - > Analog 32 bit HiFi DAC -> Digital Bluetooth -> Analog Output. Because your audio quality is limited by the lowest fidelity step in the reproduction chain, your final output quality is never going to exceed the fidelity of the conversion in your Bluetooth device. And because the analog -> digital conversion actually lowers audio quality, the highest quality Bluetooth playback can be achieved by keeping the signal digital until it reaches the output device. Running the signal through the HiFi DAC would actually result in lower quality Bluetooth audio.
If you want to match the quality of the HiFi DAC using Bluetooth, you'll need to buy a Bluetooth receiver with a high quality onboard DAC of its own.
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Couldn't have got a better answer, thank you.
I'd like to find out whether the device can do more than this:
(+) Audio (aac 2ch 48000Hz) @ 96kbps
/system/etc/media_profiles.xml has a dedicated entry for the audio quality in aac and it shows that the max. bps is 96000, which is what all video files use as their audio encoders.
I have played around with some professional audio equipment and managed to connect a Steinberg/Yamaha audio interface to the OP5 via USB to go and it provided enough power to drive the external condenser mics with 48V phantom power.
The native OP camera app doesn't enable external microphones when plugged in, but 'OpenCamera' does and I've made a video of myself playing the piano:
https://www.youtube.com/watch?v=by_9NInRfzE
In the original file the audio only has the specified bitrate as shown in this entry of the media_profile.xml:
Code:
<AudioEncoderCap name="aac" enabled="true"
minBitRate="8000" maxBitRate="96000"
minSampleRate="8000" maxSampleRate="48000"
minChannels="1" maxChannels="6" />
Is there a way to specify a distinctly higher bitrate for the audio track of a video, specifically with 3rd party applications that allow for external microphones?