How to Use Opus Codec for Bitrate Scalability in Audio Streaming

Learn how to use Opus codec for bitrate scalability in audio streaming, and how to adjust the bitrate dynamically according to the network conditions and the audio content.

Opus is a highly versatile audio codec that supports both speech and music transmission over the Internet. Opus is designed to provide high-quality audio at low bitrates, and to adapt to different network conditions and audio content. One of the key features of Opus is bitrate scalability, which means that the bitrate of the encoded audio can be changed during a stream without affecting the quality or the synchronization. Bitrate scalability can be useful for audio streaming applications, such as VoIP, video conferencing, or online gaming, where the network bandwidth may vary or the audio content may change. In this article, we will explain how to use Opus codec for bitrate scalability in audio streaming, and how to adjust the bitrate dynamically according to the network conditions and the audio content.

What is Opus codec and why is it suitable for bitrate scalability?

Opus is an open, royalty-free, and standardized audio codec that was developed by the Internet Engineering Task Force (IETF) as RFC 6716. Opus is based on two existing codecs: SILK, which is optimized for speech, and CELT, which is optimized for music. Opus can switch between these two modes, or use a hybrid mode, depending on the audio content. Opus can also adjust the sampling rate, the frame size, and the number of channels, to provide the best audio quality for the given bitrate.

Opus is suitable for bitrate scalability because it supports a wide range of bitrates, from 6 kbps to 510 kbps, and can change the bitrate on the fly without affecting the quality or the synchronization. Opus also supports variable bitrate (VBR), which means that the bitrate can vary according to the complexity of the audio content. For example, a simple speech signal may require a lower bitrate than a complex music signal. Opus also supports constant bitrate (CBR), which means that the bitrate is fixed regardless of the audio content. CBR can be useful for applications that require a predictable bandwidth usage.

How to use Opus codec for bitrate scalability in audio streaming?

To use Opus codec for bitrate scalability in audio streaming, you need to encode and decode the audio using the Opus library, which is available for various platforms and languages. The Opus library provides a high-level API for encoding and decoding Opus streams, as well as a low-level API for accessing the Opus parameters and functions. The Opus library also provides some tools and examples for testing and experimenting with Opus.

1. To encode and decode Opus streams, you need to create an Opus encoder and an Opus decoder, and configure them with the desired parameters, such as the bitrate, the sampling rate, the frame size, the number of channels, and the mode. You can also use the default parameters, which are suitable for most applications. You can then feed the raw audio data to the encoder, and get the encoded Opus packets. You can then send the Opus packets over the network, and receive them at the decoder. You can then feed the Opus packets to the decoder, and get the decoded raw audio data.
2. To adjust the bitrate dynamically during a stream, you can use the opus_encoder_ctl function, which allows you to control the encoder parameters at runtime. You can use the OPUS_SET_BITRATE request to set the target bitrate for the encoder, and the OPUS_GET_BITRATE request to get the current bitrate of the encoder. You can also use the OPUS_SET_VBR and OPUS_SET_VBR_CONSTRAINT requests to enable or disable the VBR mode, and the OPUS_SET_COMPLEXITY request to set the encoding complexity, which affects the CPU usage and the quality.
3. To determine the optimal bitrate for a stream, you can use various methods, such as:
   • Measuring the network bandwidth and the packet loss rate, and adjusting the bitrate accordingly. You can use the opus_packet_get_nb_samples function to get the number of samples per channel in an Opus packet, and the opus_packet_get_bandwidth function to get the bandwidth of an Opus packet. You can then calculate the bitrate of an Opus packet as: bitrate = (packet_size * 8) / (number_of_samples / sampling_rate)

1. You can then compare the bitrate of the Opus packets with the available network bandwidth, and increase or decrease the bitrate accordingly. You can also monitor the packet loss rate, and decrease the bitrate if the packet loss rate is too high.
2. Analyzing the audio content and the listener’s preference, and adjusting the bitrate accordingly. You can use the opus_encoder_get_analysis_info function to get some information about the audio content, such as the voice activity, the music activity, and the speech probability. You can then use this information to adjust the bitrate according to the audio content. For example, you can use a lower bitrate for speech than for music, or a higher bitrate for high-quality music than for low-quality music. You can also use the listener’s preference, such as the desired quality or the bandwidth limit, to adjust the bitrate accordingly.

Frequently Asked Questions (FAQs)

Question: How do I encode and decode Opus streams using the command-line tools?

Answer: You can use the opusenc and opusdec tools, which are included in the Opus library, to encode and decode Opus streams using the command-line. The syntax of the opusenc tool is:

opusenc [options] input_file output_file.opus

The input_file can be a WAV, AIFF, FLAC, Ogg/FLAC, or raw PCM file. The output_file.opus is the encoded Opus file. You can use various options to specify the parameters for the encoder, such as the bitrate, the sampling rate, the frame size, the number of channels, and the mode. You can use the –help option to see the full list of options.

The syntax of the opusdec tool is:

opusdec [options] input_file.opus output_file

The input_file.opus is the encoded Opus file. The output_file can be a WAV, AIFF, or raw PCM file. You can use various options to specify the parameters for the decoder, such as the sampling rate, the number of channels, and the gain. You can use the –help option to see the full list of options.

Question: How do I play Opus files on various platforms and applications?

Answer: Opus is supported by various platforms and applications, such as:

• Web browsers: Opus is supported by most modern web browsers, such as Chrome, Firefox, Edge, Safari, and Opera. You can use the HTML5 audio element to play Opus files on web pages, or use the Web Audio API to process Opus streams in web applications.
• Mobile devices: Opus is supported by most Android and iOS devices, either natively or through third-party applications. You can use the Android MediaCodec API or the iOS AVFoundation framework to play Opus files on mobile applications, or use the Opus library to encode and decode Opus streams on mobile devices.
• Desktop applications: Opus is supported by various desktop applications, such as VLC, Foobar2000, Winamp, Audacity, and Skype. You can use these applications to play Opus files on your desktop, or use the Opus library to encode and decode Opus streams on your desktop applications.

Summary

In this article, we learned how to use Opus codec for bitrate scalability in audio streaming, and how to adjust the bitrate dynamically according to the network conditions and the audio content. We also learned how to use the Opus library, the command-line tools, and the various platforms and applications that support Opus. We hope that this article was helpful and informative, and that you can use Opus codec to achieve high-quality and adaptive audio streaming.

Disclaimer: This article is for informational purposes only, and does not constitute professional advice. We are not responsible for any damages or losses that may result from following the instructions or suggestions in this article. Please use your own discretion and judgment when applying any of the solutions or tips mentioned in this article.