WhatsApp is one of the most popular messaging apps worldwide, with over 2 billion active users. A key feature of WhatsApp is voice calls, which allow users to make free voice and video calls over the internet. But what audio codec does WhatsApp use to encode the audio for these calls? Keep reading to find out.
The Basics of Audio Codecs
An audio codec is a software program that compresses and decompresses digital audio data. Audio codecs aim to reduce the size of audio files while maintaining sound quality. This is important for streaming and sharing audio online, as smaller files require less bandwidth to transmit.
Some key terms related to audio codecs:
- Bitrate – The amount of data transmitted per second, measured in kbps (kilobits per second)
- Sample rate – How many times per second the sound wave is sampled, measured in Hz (Hertz)
- Compression – Reducing the size of audio data by removing unnecessary or redundant information
- Decompression – Reconstructing compressed audio data back into its original form
Codecs balance these factors to optimize audio quality and file size. Higher bitrates and sample rates result in better quality but larger files. More compression means smaller files but can reduce quality if taken too far.
WhatsApp’s Choice of Audio Codec
For voice calls, WhatsApp uses the Opus audio codec to encode and transmit audio. Opus is optimized for both speech and music transmission over the internet.
Here are some key details about the Opus codec used by WhatsApp:
- Bitrates between 6 – 510 kbps
- Sampling rates from 8 – 48 kHz
- Frame sizes from 2.5 – 60 ms
- Support for up to 255 channels (though WhatsApp uses mono)
- Latency as low as 5-10 ms
Opus entered the open-source world in 2010. It was created by the Internet Engineering Task Force (IETF) as a successor to earlier codecs like MP3, Vorbis, and AAC.
Opus gained popularity because it achieves better sound quality than these codecs at similar bitrates. It’s optimized for both speech and music at a range of bitrates.
Benefits of Opus for WhatsApp
Here are some key benefits of using the Opus codec for WhatsApp voice calls:
- Low latency – Opus can achieve latency as low as 5-10 ms, which is imperceptible during a call
- Flexibility – Opus dynamically adapts to varying network conditions, adjusting the bitrate to optimize quality
- Efficient compression – Opus provides better quality than older codecs at similar bitrates
- Error resilience – Opus is robust to packet loss, allowing good call quality even on spotty connections
- Open standard – As an IETF open standard, Opus is license-free and open for anyone to implement
In summary, Opus allows WhatsApp to deliver good voice call quality across a wide range of devices, processors, and network conditions – key for a globally popular messaging app.
Opus Audio Quality
The sound quality of Opus can vary depending on factors like bitrate and network conditions. Here’s a rough guide to the audio quality you can expect at different bitrates:
| Bitrate Range | Audio Quality |
|---|---|
| 6 – 32 kbps | Low bandwidth, suitable for speech |
| 32 – 64 kbps | Good quality speech |
| 64 – 128 kbps | Very good quality speech, music starts to become good |
| 128 – 510 kbps | Excellent quality speech and music |
In perfect network conditions, WhatsApp will use bitrates up to 128 kbps, providing excellent voice call quality. The bitrate may drop lower in poor network conditions to maintain the call.
How WhatsApp Implements Opus
WhatsApp doesn’t implement the Opus codec directly. Instead, it relies on the open-source WebRTC framework to handle audio encoding and transmission.
Key details on how WhatsApp leverages WebRTC and Opus:
- WebRTC handles all the audio capturing, encoding, packetizing, and network transmission
- Opus is one of the mandatory audio codecs supported by WebRTC
- WhatsApp just interfaces with WebRTC APIs to initiate and connect calls
- WebRTC automatically selects the best codec and settings between callers
By leveraging WebRTC, WhatsApp offloads the complexity of implementing audio codecs. WebRTC handles all the Opus encoding in the background. All WhatsApp needs to do is exchange signaling data to set up and manage the calls.
WebRTC Media Codecs
Here are the codecs mandated by WebRTC that Opus competes against:
- Opus – WhatsApp’s primary codec as discussed above
- G.711 – A legacy narrowband speech codec
- G.722 – A wideband speech codec for HD voice
- iSAC – A Google adaptive speech codec
- iLBC – A popular narrowband speech codec
During WebRTC call setup, both parties share which codecs they support from this list. WebRTC will automatically select Opus in most cases as the optimal modern codec.
Conclusion
To summarize, WhatsApp relies on the advanced Opus audio codec implemented within WebRTC for its voice calling feature. Opus provides an optimized combination of sound quality, even at lower bitrates, along with low latency performance ideal for real-time communication.
Leveraging WebRTC allows WhatsApp to avoid developing its own audio stack while benefiting from the performance of Opus. The flexibility of Opus and WebRTC powers excellent voice call quality for WhatsApp’s billions of diverse users around the world.
So the next time you make a WhatsApp call, you can appreciate the technology that allows it to sound so good! Opus and WebRTC are delivering crystal clear conversations to WhatsApp users everywhere.
Some key takeaways:
- WhatsApp uses the Opus audio codec developed by the IETF
- Opus provides great quality at low bitrates, ideal for WhatsApp calling
- WebRTC handles the Opus encoding/decoding and transmission
- WebRTC chooses the optimal codec and settings between callers
- Opus allows high quality WhatsApp calls on any device and network worldwide