There are so many different codecs for phone calls and many of them have very high license fees, meaning it will take a lot of time before everyone can use normal telephony with wide band audio.
Is Opus supported for VoLTE?
The usual codecs for VoLTE are AMR, AMR-WB and EVS (see links below for more info - thanks, #Mikael Dúi Bolinder).
As with most mainstream voice (and video codecs) there is IPR and licensing associate with these. However, for end users the network providers and device manufacturers have included the licensing and the codecs in their rollouts so a typical operator service will use these.
I'm not aware of any restrictions from 3GPP on using other codecs if the devices and the network support them, but the above are definitely the default and the most widely used.
If you want to create your own voice service, e.g a VoIP service running over the data connection to the phone, then in theory you can use whatever codec you want. It's worth being aware that for software based codecs, which they will be unless they are tightly integrated in the device's hardware, the efficiency is important as an inefficient implementation may impact performance, battery life etc.
For Opus in particular there are several open source projects which provide Android libraries for this, for example. Opus is also supposed to be supported on devices from Android 5+ (https://developer.android.com/guide/topics/media/media-formats).
amr-licensing-wikipedia: https://en.wikipedia.org/wiki/Adaptive_Multi-Rate_audio_codec#Licensing_and_patent_issues "AMR licensing (and issues) on Wikipedia"
amr-wb-licensing-wikipedia: https://en.wikipedia.org/wiki/Adaptive_Multi-Rate_Wideband#Licensing "AMR-WB licensing on Wikipedia"
evs-news-patent-pool: http://www.mpegla.com/Lists/MPEG%20LA%20News%20List/Attachments/97/n-16-01-20.pdf "MPEG developing a patent pool for EVS"
Related
I've built a voip app for iphone and android phone (audio only). I've been testing the call quality. I noticed that if a person has an upload speed of less than 600kbps, then his partner will have trouble hearing him clearly. His partner might hear lots of crackling, dropped sentences in speech, or nothing at all.
I am currently using GSM codec on my android, iphone and on my asterisk server.
So my question is whether upload speeds of less than 600kpbs is generally considered too slow for VOIP calls?
If so, is there anything I can do to reduce necessary bandwidth for call? I could consider the G722 codec, but from what I remember, it doesn't offer significant bandwidth improvement compared to GSM
Additional Notes
I used speedtest.net to inform me of my upload speed. I'm not sure if that's reliable way to test speeds necessary for voip service.
Also, I'm using linphone core as my SIP library.
So far, I've used many different Audio Production software on Mac and Windows platforms. Often times, I ponder on the idea of creating my own DAW, but I realize that would be an extremely difficult challenge for a single person to undertake (especially if only knowledgeable in one particular area / language of programming).
There's a flood of ideas / features that comes to my mind just by the thought of some of the other DAWs I've used. From implementing MIDI in/out, Audio Routing, Mix Buses, VST support, User Interface for a Piano Roll and Song view, etc...
So my question is...
Which roles would be required in a team of developers to create a complete Digital Audio Workstation (DAW) Software?
I think the right answer is several good developers (you don't need so many, perhaps 3) a good product manager, an ui designer/graphist a lot of testers. And a good coffee machine.
The real problem is what kind of DAW do you want, portable on mac and windows, which OSs, which formats (vst 2, 3, AU, RTAS, AAX, rack extension, DX), do you want only MIDI and adio tracks, which external MIDI devices you want to support, do you support OSC, other protocols?
What will be the features of you mixer, integrated effects? What support of audio API on windows (wasapi, asio ...) do you want some cloud feature ? community or online store integration?
What kind of breakthrough would you have compared to cubase, live, PT, DP, Logic, garage band, bitwig, studio one, sonar, fl studio ...? Do you want modular patches or just tracks? Will you have advanced integrated controls or midi modifiers?
All that is the problem...
This is a very complex question!
I'm currently using the waveInOpen set of Windows API functions to record audio for a VOIP application. I'm now being asked to add echo cancellation, and possibly noise reduction, and gain control. I know nothing about DirectSound, but while searching on "echo cancellation" on Google I came across references on MSDN to DirectSound such as CaptureAcousticEchoCancellationEffect.
If I switch to DirectSound will I get some of these features "for free"? Are they only supported if the hardware supports it, and if so, how often will that hardware be present in the average consumer PC?
Starting with Windows Vista, Microsoft provides a separate component Voice Capture DSP:
The voice capture DMO includes the following DSP components:
Acoustic echo cancellation (AEC)
Microphone array processing
Noise suppression
Automatic gain control
Voice activity detection
Applications can turn each component on and off individually.
You can use it in your DSP application to leverage EAC and NS implemented in software.
As far as I know these features are not professionally supported in DirectSound. A hardware device that has support for these features usually is equipped with a special processor/DSP and costs a lot more than the standard hardware device.
I was wondering whether it is possible to capture audio data from other sources like the system out, FM radio, bluetooth headset, etc. I'm particularly interested in capturing audio from the FM radio and already investigated all possibilities including trying to sniff the raw bluetooth communication between the phone and the radio device with no luck. It's too bad Android only allows recording audio from the MIC.
I've looked at the Android source code and couldn't find a backdoor to allow me to do that without rooting the device. Do you, at least, have any idea how to use other devices (maybe access somehow /dev/audio) say via NDK or even better - Java (maybe Reflection?) to trick the system to capture the audio stream from say, the FM radio. (in my case I'm trying to develop the app for the HTC Desire)
PS. And for those of you who are against using undocumented APIs, please don't post here - I'm writing an app that will be for my personal use or even if I ever publish it I will warn the user of possible incompatibilities.
I've spent quite some time deciphering the audio stack, and I think you may try to hijack libaudio. You'll have trouble speaking directly to the hardware (/dev/*) because many devices use proprietary audio drivers. There's no rule in this regard.
However, the audio hardware abstraction layer (HAL) provided by /system/lib/libaudio.so should expose the API described at http://source.android.com/porting/audio.html
The Android system, and especially audioflinger, uses this libaudio HAL to find available devices, deal with routing, and of course to read/write PCM data.
So, you could hijack the interaction between audioflinger and libaudio, by renaming the later, and providing your own libaudio which decorates the real one. Doing so, you should be able to log what happens and very possibly intercept FM radio output, provided that this is not directly handled by the hardware.
Of course, all this requires rooting. Please comment if you manage to do this, that interests me.
I wonder; does audio software like Cubase and Audacity use PlaySound calls??
Where can I learn about low level audio programming? As far as I've found information on the web, MCI seems to be the lowest level audio API in Windows...
Thanks
Edit: I don't ask for information specific for Windows only.
There's several audio APIs to choose from. The oldest and most widely supported is the waveOut API - look for functions starting with waveOut in MSDN. A slightly newer one is DirectSound which is geared more towards games, but it's main feature over waveOut is positional 3D sound which professional audio software doesn't use (it was also supposed to have lower latency than waveOut, but that never really materialized). For low latency audio, there is ASIO. Professional audio apps support this API, but not all drivers do (it's a standard feature in professional sound cards, but not gaming or on-board hardware). ASIO can provide much lower latency than waveOut or DirectSound. Finally, there's the kernel streaming interface, which is the lowest-level audio interface still accessible from user-mode code. This is a direct pipe into Windows's internal mixer which combines output from all apps that are currently playing sound into the signal that gets sent to the sound card. It's scarcely documented though. There's a driver called ASIO4ALL (just google it) that provides ASIO support on soundcards without ASIO drivers by implementing the ASIO API on top of the kernel streaming interface.
I'm a little late to the game here, but I posted a Windows API history last week that might add a little more context. The choice of API really depends on your needs. If you want to avoid 3rd party libraries, it really only comes down to MME, XAudio2, and Core Audio (WASAPI).
A Brief History of Windows Audio APIs
Hope this helps!
Actually, if you are looking for more than Windows-only output support, then the best way to start is to review Phil Burk's PortAudio, available as of this writing at http://www.portaudio.com/ .
ASIO is a good quality interface, but it's proprietary and owned by Steinberg.
There are many lower-level interfaces to audio output than MCI in modern Windows. These include, at least, DirectSound, XAudio and WASAPI.
I recommend avoiding the Windows APIs as much as possible, and learning PortAudio instead.