Opus is a state-of-the-art royalty-free lossy audio codec convering more applications than any other single audio codec— from low latency VoIP to high fidelity music storage. After five years of open development, including contributions from Xiph.Org, Skype/Microsoft, Mozilla, Broadcom, and many individual developers, Opus was standardized in 2012 by the IETF in RFC 6716 and has since been deployed to hundreds of millions of computers and devices.
Daala is a new open effort to build a state-of-the-art video codec targeting compression performance beyond HEVC and VP9. Leveraging the experience we had with Opus we are building a new technical framework for video coding the ground up to avoid patent thickets and be royalty free: By breaking from the common design pattern of block based transform codecs we avoid many licensing complications and create an opportunity to better resolve some of the weaknesses of existing formats.
If you've been watching streaming video for as many years as I have, then no doubt you agree that one of the most remarkable changes has been the quality of video delivered. It wasn't that long ago when postage stamp sized windows with audio and video out of synch were the norm, whereas today, we can watch on big screen TVs, with buffering a random occurrence.
But the quality bar is getting even higher as this morning eyeIO ("I-I-O"), an early stage video processing technology company, is announcing it has been awarded THX certification for digital cinema HD video quality, a first for an online video encoder. The certification program objectively tests picture quality using 46 data points in 6 categories that were developed by the major Hollywood studios. A score over 90, on a scale of 100, is viewed as "THX Excellent Quality." eyeIO achieved a score of 95.528 for its H.264 first-generation encoder, when streaming at a rate of 5.8 mbps.
As expected, hardware and software tools for CALM Act loudness compliance were at the fore at this year’s NAB Show. But the audio buzzword of the show was MADI: the AES10 standard, introduced in 1991 and updated in 2003, has become increasingly popular in broadcast audio products over recent years.
Presently, the audio playback of various streams from different sources causes quite often an annoying experience for the listeners. The reason behind that is that the loudness levels of these different streams can be completely different and do not match a comfortable listening experience on the end users’ devices.
Measurements in the past have shown that even similar audio streams which are broadcast via the broadcast network and simultaneously on the Web, e.g. Internet radio services, might have very different loudness levels. Even more different levels can be expected if various streams from completely different sources are collected to form a Rich Media Service.
Why was this of interest to NoTube?
In the NoTube platform, media coming from different sources are presented to the end user. This brings along a high risk of annoying loudness jumps, among others between news items that are provided by different broadcasters. The use of a loudness module in the NoTube platform could eliminate this effect by normalising the loudness levels.
Loudness, if you will, increasingly has been making noise in the broadcast industry. Standards organizations have been working long and hard to publish recommendations, best practices and specifications to guide the industry and ensure consistent measurements.
Governments are introducing legislation to promote and enforce better regulation. Equipment vendors are writing articles to discuss the situation. Experts have penned books on the subject. Yet, despite all the available information, there remains a great deal of uncertainty as to how to best deal with the issue.
The loudness paradox lies at the heart of the matter. On one side, there is a desire to maintain the artistic integrity of the original program (translation: don't modify audio). On the other, there is a need to minimize viewer frustration with regard to loudness inconsistencies across different content and channels. This often requires some level of audio processing.
The paradox is this: How can loudness be kept consistent if the audio is not modified?
Dolby Laboratories today announced that Adobe AIR 3 will support Dolby Digital Plus to deliver surround sound to connected devices including televisions, set-top boxes, and Blu-ray Disc players. Content aggregators will be able to build Flash-based applications that deliver the complete high-definition entertainment experience with HD video and multichannel Dolby surround sound.
Adobe AIR 3 supports streaming of HD video with Dolby Digital Plus for up to 7.1 channels on select connected televisions, Blu-ray Disc players, and set-top boxes.
Broadcasters are exploring immersive audio systems to keep pace with the advances made in 3DTV and higher resolution displays.
Stefan Meltzer a technical consultant who represents the Fraunhofer IIS at DVB level says there are different degrees of interest among European broadcasters about adding a third dimension to audio.
“The big question is how to do it technically and also how many additional speakers will user’s accept,” he says.
Pieter Schillebeeckx, head of R&D at Soundfield has met with broadcasters who have moved from SD to HD and 3D on the video side but “feel the audio has stagnated at 5.1 and they are asking what next?” he says.
The 5.1 surround format has its limitations, the most obvious being reproduction of audio restricted to the horizontal plane. The 7.1 format adds two further channels in an attempt to improve the sense of envelopment in the horizontal plane, but again does not reproduce a sensation of height.
“There are problems with discrete channel formats such as 5.1, including the effect of non-standard speaker layouts at the user’s end which are beyond the control of the broadcaster,” says BBC R&D’s Chris Dunn.
The next generation of Dolby’s audio format Dolby Digital Plus (DD+), is capable of providing channel configurations in 7.1, 9.1, even to 22.2. The latest DVB transmission systems also include this capability supporting DD+ and HE-AAC/AAC. The BBC for example is using AAC and HE-AAC for Freeview HD. AAC/HE-AAC supports in its bit stream up to 48 channels.
A major advantage of HTTP Live Streaming, and it is still unbeaten by any other standard, is that you can use any technology to compose multimedia content. You don’t need expensive media servers to handle media streams. You can easily delegate media composition to some simple PHP / Python / Ruby / Node.js modules. As a long-time Flash lover I was somewhat prejudiced against the Apple’s video delivery standard. But as a developer, every day I use it to solve tasks which have been almost unattainable with the previous stack (or required expensive software). Adobe HDS playlists have been really hard to deal with. Just think of binary data in f4m playlists. They require much more time to develop & debug the solution. MPEG DASH is also far from being intuitive.
In this post, we are going to discuss how to make an alternative audio track for your video. Although HTTP Live Streaming can streamline this task, yet there are some limitations, so you need to make certain hacks on the client side. In our Togetherproject, we had to implement alternate sound tracks for user videos. Luckily, we use HTTP Live Streaming throughout the system.
The EBU Media, Radio & Television sessions (10-11 July) during the LSM will show EBU Members' and other media professionals' Open Source authoring, recording, ingest, editing, management, storage, encoding, play-out, streaming, broadcast transmission and user devices software projects, how they are used in practice and what its features/limitations are. Colleagues from EBU Members and from the industry will share their experiences and outlook for the future. Live demonstrations of the tools will be available too, including a completely open radio transmission platform. A series of workshops will provide a deeper insight into selected tools.
Visitors to CES and the Detroit Auto Show can sample 5.1 surround sound streamed at only 64 kbps bandwidth. That’s because those are the two locations where Fraunhofer is publicly debuting its MPEG Surround audio codec.
Surround sound audio typically takes 160- to 400 kbps for a stream, notes a Fraunhofer representative, but with MPEG Surround the requirement is far less. That gives content providers the ability to stream to new locations, such as cars using a 3G connection.
Web browsers are becoming more and more powerful, and new APIs like the Web Audio API and Audio Data API are making the web an ever more interesting and dynamic place. At ofmlabs we are invested in improving the state of audio on the web, and with JSMad showed that decoding audio purely in JavaScript is possible thanks to these APIs and the hard work of browser makers and spec authors. And now we've done it again. Introducing alac.js, a port of the recently open sourced Apple Lossless decoder to JavaScript. Now it is possible to play MP3 and Apple Lossless even in browsers without native support.
DTS announced today its collaboration with Adobe to bring the DTS surround sound experience to upcoming connected TVs and Blu-ray Players powered by Adobe AIR 3. Adobe’s latest version of AIR, released today, will enable Flash-based application experiences with high definition (HD) video and up to 7.1 channels of best-in-class DTS surround sound. This premium experience will be demonstrated at Adobe MAX, Adobe’s annual developer conference, which kicked off today.
AIR 3 is able to deliver all profiles of the DTS-HD codec, from higher bit-rate DTS-HD Master Audio, down to the extremely efficient DTS Express, optimized for “over-the-top” and web-connected ecosystems. DTS Express provides content distributors with the ability to deliver stereo, 5.1, and 7.1 surround sound at superior quality and flexibility, with bitrates ranging from 48 kbps up to 512 kbps, a “sweet-spot” for connected entertainment.
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