The codec offers a range of high quality algorithm options as well as 16 Bit 22kHz linear PCM audio at less than 12 ms encode delay for high quality, uncompressed audio.
All Bridge-IT and Bridge-IT XTRA codecs include Opus, MPEG Layer 2, G.711 and G.722 algorithms, as well as the AAC suite of algorithms and Tieline Music and MusicPLUS as standard. Music and Music PLUS are optimized for wired and wireless IP connections.
aptX® Enhanced is included in Bridge-IT XTRA and can be purchased separately as a license upgrade in Bridge-IT codecs.
Bridge-IT Algorithm Encode License Options |
Bridge-IT |
Bridge-IT XTRA |
AAC-LD, AAC-ELD, LC-AAC, HE-AAC v.1 and HE-AAC v.2 algorithms |
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16 bit and 24 bit aptX® Enhanced algorithm |
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Included
* Option available for purchase separately if required.
Note: Bridge-IT has a range of default connection profiles that make it very simple to easily program your codec to connect using all available algorithms. See Choosing Dialing Profiles for more details.
Overview of Tieline Algorithms
1.The Tieline Music algorithm is optimized for audio bit rates as low as 19.2kbps with only a 20 millisecond encode delay. It offers 15 kHz mono from 24Kbps to 48Kbps.
2.Tieline MusicPLUS delivers up to 20 kHz mono from 48kbps upwards. It can also deliver up to 20 kHz stereo from 96kbps upwards, offering huge savings on your IP data bills and outstanding audio quality.
Overview of AAC Algorithms
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Important Notes: AAC algorithms are only available in Bridge-IT if the AAC license has been purchased and uploaded into the Bridge-IT codec. For more information see Installing Software Licenses. |
AAC-LC
LC-AAC is optimized for audio bit-rates of 64Kbps per channel or higher using a sample rate of 48kHz. Tieline recommends using LC-AAC instead of HE-AAC if bandwidth of 64Kbps or higher per channel is available, to optimism audio quality. If lower than 64Kbps is available, consider using HE-AAC, Tieline Music or Tieline MusicPLUS.
AAC-HE
Codecs include both HE-AAC v.1 and HE-AAC v.2, which are optimized for low bit rate connections. Selection of HE-AAC v.1 and v.2 is automatically managed within the codec, so only AAC-HE is displayed on the screen. When used for mono connections, HE-AAC v.1 performs best at bit rates of 24kbps per channel or higher. HE-AAC v.1 is also used for stereo connections when audio connection bandwidth is 48kbps or higher.
HE-AAC v.2 is used for stereo connections when audio connection bandwidth is below 48kbps and is capable of delivering 15kHz quality stereo audio at audio bit rates as low as 24kbps.
A sample rate of 32kHz is used in the codec's default profiles to achieve ultra-low bit-rate connections, but this is adjustable to 44.1kHz or 48kHz if required.
AAC-LD
AAC-LD (Low Delay AAC), AAC-ELD (Enhanced Low Delay AAC) and AAC-ELDv 2 are optimized for low latency real-time communication. AAC-LD is suited to bit rates of 96kbps or higher for stereo audio.
AAC-ELD
AAC-ELD is optimized for high quality stereo connections from 48 - 96kbps and performs better at these bit rates when compared with AAC-LD.
AAC-ELD v 2
For stereo connections below 48kbps AAC-ELD v2 will deliver better performance than AAC-ELD down to 24kbps.
Overview of aptX® Enhanced Audio Coding
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Important Notes: aptX® Enhanced is only available if the aptX® Enhanced license has been purchased and uploaded into the codec. For more information see Installing Software Licenses. |
aptX® Enhanced audio coding is used by thousands of radio stations to deliver very low delay audio for studio to transmitter links, audio distribution and remote broadcasts. It delivers outstanding audio quality with exceptionally low delay across a range of IP networks. It is ideal for high quality studio-to-transmitter links and audio distribution.
32kHz, 44.1kHz or 48kHz sampling rates are available at either 16 bit or 24 bits per sample. aptX® Enhanced has a minimum connection bit-rate of 128Kbps per channel and offers 10Hz to 24kHz frequency response. 24 bit, 48kHz aptX® Enhanced at the maximum bit-rate of 576Kbps delivers >120dB of dynamic range.
Overview of Opus Algorithm
Opus is a highly versatile open source audio coding algorithm. It incorporates technology from the well-known SILK and CELT codecs to create a low latency speech and audio codec. It is a variable bit rate algorithm ideal for live broadcast situations because of its capacity to deliver high quality, real-time Audio over IP (AoIP) at low bit rates. Visit http://www.opus-codec.org for more info.
There are three Opus algorithm configurations available:
Algorithm |
Recommended connection for on-air use |
Opus Voice |
High quality low bit rate remotes (9.6kbps -64kbps) |
Opus Mono |
Very high quality mono remotes, STLs and audio distribution (48kbps -128kbps) |
Opus Stereo |
Very high quality stereo remotes, STLs and audio distribution (64kbps -256kbps) |
Configuring an Algorithm in the Codec
1.Press the HOME 
button to return to the Home screen.
2.Use the navigation buttons on the front panel to select Connect and press the 
button.
3.Select IP and press the button.
4.Select your preferred IP Session mode and press the button.
5.Use the down 
navigation button to select Setup and press the button.
6.Navigate to Algor'm and press .
7.Select the mono or stereo algorithm that you want to connect with and press .
How do I choose the right algorithm?
The algorithm you select will not only affect the quality of the broadcast but it will also contribute to the amount of latency or delay introduced. For example, if MP2 algorithms are used, program delays will be much longer than when using Tieline Music or MusicPLUS algorithms. This is due to the additional inherent encoding delays involved when using MP2 algorithms. This can be a major consideration for live applications that integrate remote-crosses into a broadcast. The algorithm you choose to connect with will also depend upon:
•The codecs you are connecting to (Tieline versus non-Tieline)
•Whether you are creating multi-unicast connections.
•Whether you are connecting using SIP or not.
•The uplink bandwidth capability of your broadband connection.
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Important Notes: Music and MusicPLUS algorithms cannot be used over SIP connections. Use MP2 algorithms at 64kbps mono or 128kbps stereo for high quality connections when using SIP, or use G.711 and G.722 if required. Tieline G3 codecs do not support connections using AAC, aptX Enhanced and Opus algorithms and will default to MPEG Layer 2 if an incoming connection is configured to use these algorithms. |
It can be a good idea to listen to the quality of your program signal using each algorithm and to see how it sounds when it is sent at different connection bit-rates (as well as different FEC and jitter-buffer millisecond settings). This will assist you to determine what the best algorithm setting is for the connection you are setting up. Please see the following table for details on the connection requirements of the different algorithms available.
Algorithm |
Audio Band- width |
Algor-ithmic Delay |
IP bit rate per channel |
IP over-head per connection |
Audio Quality and Features |
Recommended applications for on-air use |
PCM/Linear (Uncom- pressed) |
16/24 bit up to 24kHz |
0ms |
sample rate x bits per sample x no. channels |
80kbps |
•Full bandwidth, perfect audio quality for voice and music •No error concealment/correction or artifacts |
•Extremely high quality PCM linear uncompressed audio for STLs and audio distribution. •Ideal for fiber or high bandwidth links. |
Tieline Music |
Up to 15kHz |
20ms |
24 kbps minimum |
16kbps |
•High quality voice and music •Very low delay at low bit rates |
•Great for live voice or music remotes as well as STLs and audio distribution with limited connection bandwidth (e.g. 3G wireless) •Suitable when bidirectional communication between announcers is required |
Tieline Music- PLUS |
Up to 22kHz |
20ms |
48 kbps minimum (Optimized for 64kbps per audio channel) |
16kbps |
•Very high quality voice and music •Very low delay at low to moderate bit-rates |
•Very high quality, very low delay STLs and audio distribution •Remote connections able to achieve 48kbps for each audio channel •Suitable when bidirectional communication between announcers is required |
G.711 |
3kHz |
1ms |
64kbps minimum |
80kbps |
•Low quality 3kHz POTS phone quality audio •Very low delay at moderate bit rates |
•Highly compatible with other brands of audio codec •Low quality and used generally for compatibility |
G.722 |
7kHz |
1ms |
64kbps minimum |
80kbps |
•Good quality 7kHz voice •Better quality than a standard POTS phone call •Very low delay at moderate bit rates |
•Highly compatible with other brands of audio codec •Good voice quality audio for remotes and other voice quality applications |
MPEG Layer 2 |
Up to 22kHz |
24 to 36ms |
64kbps minimum |
8.5 - 13.3kbps |
•Very high quality voice and music •Low to moderate delay at moderate to high bit rates |
•Highly compatible with other brands of audio codec •Very high quality audio for remotes, STLs and audio distribution |
LC-AAC |
Up to 15kHz |
64ms |
64kbps |
15kbps |
•High quality voice and music at lowest bit rate; better quality at higher bit rates •Moderate delay at moderate to high bit rates |
•Voice or music remotes as well as STLs and audio distribution where some delay is tolerable •Tieline Music or MusicPLUS deliver lower delay |
HE-AAC v.1 |
Up to 15kHz |
128ms |
48kbps |
7.4kbps |
•High quality voice and music at the lowest bit rate; better quality at higher bit rates •Low to Moderate bit rates •High delay |
•Live voice or music remotes as well as STLs and audio distribution with limited connection bandwidth •Use when bidirectional communication between announcers is not required |
HE-AAC v.2 |
Up to 15kHz |
128ms |
Minimum 16kbps (Mono); 24kbps (stereo) |
7.4kbps |
•High quality voice and music •Low bit rates •High delay |
•Used for DAB+ radio streaming •Ideal for low bit rate remotes •Use when bidirectional communication between announcers is not required |
AAC-LD |
Up to 20kHz |
20ms at 48kHz |
48kbps minimum |
30kbps |
•Very high quality voice and music •Very low delay at low to moderate bit rates |
•Very high quality, very low delay STLs and audio distribution •Remote connections able to achieve 48kbps for each audio channel requiring •Suitable when bidirectional communication between announcers is required |
AAC-ELD |
Up to 20kHz |
15-30ms |
24 kbps minimum |
15-30kbps |
•Very high quality voice and music •Very low delay at low bit rates |
•Great for live voice or music remotes •Suitable when bidirectional communication between announcers is required |
AAC-ELDv.2 |
Up to 20kHz |
35ms |
Pending release |
Pending release |
•High quality voice and music •Low delay at low bit rates |
•Great for live voice or music remotes where limited connection bandwidth is available •Suitable when bidirectional communication between announcers is required |
aptX Enhanced |
10Hz-24kHz |
2.5ms at 48kHz |
128kbps minimum (16bit; 32kHz) to 288kbps (24bit;48kHz) |
80kbps |
•Very high quality voice and music •Extremely low delay at high bit rates •Highly cascade resilient |
•Ideal for STLs and audio distribution where high connection bandwidth is available and very low delay is highly desirable. •Resilient with multiple encodes/decodes when required |
Opus |
4Hz-20kHz |
20ms |
9.6-256kbps |
16kbps |
•Very high quality voice and music •Very low delay at low bit rates
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•"Opus Voice" is ideal for high quality, and low delay voice quality remotes at extremely low bit rates. •"Opus Mono" and "Opus Stereo" are perfect for high fidelity remotes, STLs and audio distribution at higher bit rates |
Algorithm Selection Guide
Algorithm |
Very Low Delay |
Moder- ate to High Delay |
Excell- ent Perfor-mance at Low Bit rates |
Preferr- ed for Live Remotes |
Preferred for STLs and Audio Distribu-tion |
Highly Compat- ible with other Codecs |
Linear/PCM |
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Opus |
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Tieline Music |
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Tieline MusicPLUS |
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apt-X Enhanced |
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LC-AAC |
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HE-AACv1 |
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HE-AACv2 |
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AAC-LD |
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AAC-ELD |
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AAC-ELDv2 |
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MPEG Layer 2 |
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G.722 |
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G.711 |
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* Use with caution for remotes due to high delay; not suitable when bidirectional communications is required.
Sampling Rates
When selecting linear (PCM) uncompressed audio or AAC, MPEG and aptX® Enhanced algorithms, it is possible to select different either 32kHz, 44.1kHz and 48kHz sample rates as required. Tieline Music runs at 32kHz sampling and MusicPLUS runs at 48kHz sampling. G.711 and G.722 will always run at a 32kHz sampling rate (down-sampled to 8kHz and 16kHz respectively).
