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• WebRTC predictions for 2016

  17 février 2016, par silvia

  I wrote these predictions in the first week of January and meant to publish them as encouragement to think about where WebRTC still needs some work. I’d like to be able to compare the state of WebRTC in the browser a year from now. Therefore, without further ado, here are my thoughts.

  WebRTC Browser support

  I’m quite optimistic when it comes to browser support for WebRTC. We have seen Edge bring in initial support last year and Apple looking to hire engineers to implement WebRTC. My prediction is that we will see the following developments in 2016 :

  • Edge will become interoperable with Chrome and Firefox, i.e. it will publish VP8/VP9 and H.264/H.265 support
  • Firefox of course continues to support both VP8/VP9 and H.264/H.265
  • Chrome will follow the spec and implement H.264/H.265 support (to add to their already existing VP8/VP9 support)
  • Safari will enter the WebRTC space but only with H.264/H.265 support

  Codec Observations

  With Edge and Safari entering the WebRTC space, there will be a larger focus on H.264/H.265. It will help with creating interoperability between the browsers.

  However, since there are so many flavours of H.264/H.265, I expect that when different browsers are used at different endpoints, we will get poor quality video calls because of having to negotiate a common denominator. Certainly, baseline will work interoperably, but better encoding quality and lower bandwidth will only be achieved if all endpoints use the same browser.

  Thus, we will get to the funny situation where we buy ourselves interoperability at the cost of video quality and bandwidth. I’d call that a “degree of interoperability” and not the best possible outcome.

  I’m going to go out on a limb and say that at this stage, Google is going to consider strongly to improve the case of VP8/VP9 by improving its bandwidth adaptability : I think they will buy themselves some SVC capability and make VP9 the best quality codec for live video conferencing. Thus, when Safari eventually follows the standard and also implements VP8/VP9 support, the interoperability win of H.264/H.265 will become only temporary overshadowed by a vastly better video quality when using VP9.

  The Enterprise Boundary

  Like all video conferencing technology, WebRTC is having a hard time dealing with the corporate boundary : firewalls and proxies get in the way of setting up video connections from within an enterprise to people outside.

  The telco world has come up with the concept of SBCs (session border controller). SBCs come packed with functionality to deal with security, signalling protocol translation, Quality of Service policing, regulatory requirements, statistics, billing, and even media service like transcoding.

  SBCs are a total overkill for a world where a large number of Web applications simply want to add a WebRTC feature – probably mostly to provide a video or audio customer support service, but it could be a live training session with call-in, or an interest group conference all.

  We cannot install a custom SBC solution for every WebRTC service provider in every enterprise. That’s like saying we need a custom Web proxy for every Web server. It doesn’t scale.

  Cloud services thrive on their ability to sell directly to an individual in an organisation on their credit card without that individual having to ask their IT department to put special rules in place. WebRTC will not make progress in the corporate environment unless this is fixed.

  We need a solution that allows all WebRTC services to get through an enterprise firewall and enterprise proxy. I think the WebRTC standards have done pretty well with firewalls and connecting to a TURN server on port 443 will do the trick most of the time. But enterprise proxies are the next frontier.

  What it takes is some kind of media packet forwarding service that sits on the firewall or in a proxy and allows WebRTC media packets through – maybe with some configuration that is necessary in the browsers or the Web app to add this service as another type of TURN server.

  I don’t have a full understanding of the problems involved, but I think such a solution is vital before WebRTC can go mainstream. I expect that this year we will see some clever people coming up with a solution for this and a new type of product will be born and rolled out to enterprises around the world.

  Summary

  So these are my predictions. In summary, they address the key areas where I think WebRTC still has to make progress : interoperability between browsers, video quality at low bitrates, and the enterprise boundary. I’m really curious to see where we stand with these a year from now.

  —

  It’s worth mentioning Philipp Hancke’s tweet reply to my post :

  https://datatracker.ietf.org/doc/draft-ietf-rtcweb-return/ … — we saw some clever people come up with a solution already. Now it needs to be implemented

  The post WebRTC predictions for 2016 first appeared on ginger’s thoughts.

• aaccoder : Implement Perceptual Noise Substitution for AAC

  15 avril 2015, par Rostislav Pehlivanov

  aaccoder : Implement Perceptual Noise Substitution for AAC
  This commit implements the perceptual noise substitution AAC extension. This is a proof of concept
  
  implementation, and as such, is not enabled by default. This is the fourth revision of this patch,
  
  made after some problems were noted out. Any changes made since the previous revisions have been indicated.
  In order to extend the encoder to use an additional codebook, the array holding each codebook has been
  
  modified with two additional entries - 13 for the NOISE_BT codebook and 12 which has a placeholder function.
  
  The cost system was modified to skip the 12th entry using an array to map the input and outputs it has. It
  
  also does not accept using the 13th codebook for any band which is not marked as containing noise, thereby
  
  restricting its ability to arbitrarily choose it for bands. The use of arrays allows the system to be easily
  
  extended to allow for intensity stereo encoding, which uses additional codebooks.
  The 12th entry in the codebook function array points to a function which stops the execution of the program
  
  by calling an assert with an always ’false’ argument. It was pointed out in an email discussion with
  
  Claudio Freire that having a ’NULL’ entry can result in unexpected behaviour and could be used as
  
  a security hole. There is no danger of this function being called during encoding due to the codebook maps introduced.
  Another change from version 1 of the patch is the addition of an argument to the encoder, ’-aac_pns’ to
  
  enable and disable the PNS. This currently defaults to disable the PNS, as it is experimental.
  
  The switch will be removed in the future, when the algorithm to select noise bands has been improved.
  
  The current algorithm simply compares the energy to the threshold (multiplied by a constant) to determine
  
  noise, however the FFPsyBand structure contains other useful figures to determine which bands carry noise more accurately.
  Some of the sample files provided triggered an assertion when the parameter to tune the threshold was set to
  
  a value of ’2.2’. Claudio Freire reported the problem’s source could be in the range of the scalefactor
  
  indices for noise and advised to measure the minimal index and clip anything above the maximum allowed
  
  value. This has been implemented and all the files which used to trigger the asserion now encode without error.
  The third revision of the problem also removes unneded variabes and comparisons. All of them were
  
  redundant and were of little use for when the PNS implementation would be extended.
  The fourth revision moved the clipping of the noise scalefactors outside the second loop of the two-loop
  
  algorithm in order to prevent their redundant calculations. Also, freq_mult has been changed to a float
  
  variable due to the fact that rounding errors can prove to be a problem at low frequencies.
  
  Considerations were taken whether the entire expression could be evaluated inside the expression
  
  , but in the end it was decided that it would be for the best if just the type of the variable were
  
  to change. Claudio Freire reported the two problems. There is no change of functionality
  
  (except for low sampling frequencies) so the spectral demonstrations at the end of this commit’s message were not updated.
  Finally, the way energy values are converted to scalefactor indices has changed since the first commit,
  
  as per the suggestion of Claudio Freire. This may still have some drawbacks, but unlike the first commit
  
  it works without having redundant offsets and outputs what the decoder expects to have, in terms of the
  
  ranges of the scalefactor indices.
  Some spectral comparisons : https://trac.ffmpeg.org/attachment/wiki/Encode/AAC/Original.png (original),
  
  https://trac.ffmpeg.org/attachment/wiki/Encode/AAC/PNS_NO.png (encoded without PNS),
  
  https://trac.ffmpeg.org/attachment/wiki/Encode/AAC/PNS1.2.png (encoded with PNS, const = 1.2),
  
  https://trac.ffmpeg.org/attachment/wiki/Encode/AAC/Difference1.png (spectral difference).
  
  The constant is the value which multiplies the threshold when it gets compared to the energy, larger
  
  values means more noise will be substituded by PNS values. Example when const = 2.2 :
  
  https://trac.ffmpeg.org/attachment/wiki/Encode/AAC/PNS_2.2.png
  Reviewed-by : Claudio Freire <klaussfreire@gmail.com>
  
  Signed-off-by : Michael Niedermayer <michaelni@gmx.at>
  

  • [DH] libavcodec/aaccoder.c
  • [DH] libavcodec/aacenc.c
  • [DH] libavcodec/aacenc.h

• WebRTC predictions for 2016

  17 février 2016, par silvia

  I wrote these predictions in the first week of January and meant to publish them as encouragement to think about where WebRTC still needs some work. I’d like to be able to compare the state of WebRTC in the browser a year from now. Therefore, without further ado, here are my thoughts.

  WebRTC Browser support

  I’m quite optimistic when it comes to browser support for WebRTC. We have seen Edge bring in initial support last year and Apple looking to hire engineers to implement WebRTC. My prediction is that we will see the following developments in 2016 :

  • Edge will become interoperable with Chrome and Firefox, i.e. it will publish VP8/VP9 and H.264/H.265 support
  • Firefox of course continues to support both VP8/VP9 and H.264/H.265
  • Chrome will follow the spec and implement H.264/H.265 support (to add to their already existing VP8/VP9 support)
  • Safari will enter the WebRTC space but only with H.264/H.265 support

  Codec Observations

  With Edge and Safari entering the WebRTC space, there will be a larger focus on H.264/H.265. It will help with creating interoperability between the browsers.

  However, since there are so many flavours of H.264/H.265, I expect that when different browsers are used at different endpoints, we will get poor quality video calls because of having to negotiate a common denominator. Certainly, baseline will work interoperably, but better encoding quality and lower bandwidth will only be achieved if all endpoints use the same browser.

  Thus, we will get to the funny situation where we buy ourselves interoperability at the cost of video quality and bandwidth. I’d call that a “degree of interoperability” and not the best possible outcome.

  I’m going to go out on a limb and say that at this stage, Google is going to consider strongly to improve the case of VP8/VP9 by improving its bandwidth adaptability : I think they will buy themselves some SVC capability and make VP9 the best quality codec for live video conferencing. Thus, when Safari eventually follows the standard and also implements VP8/VP9 support, the interoperability win of H.264/H.265 will become only temporary overshadowed by a vastly better video quality when using VP9.

  The Enterprise Boundary

  Like all video conferencing technology, WebRTC is having a hard time dealing with the corporate boundary : firewalls and proxies get in the way of setting up video connections from within an enterprise to people outside.

  The telco world has come up with the concept of SBCs (session border controller). SBCs come packed with functionality to deal with security, signalling protocol translation, Quality of Service policing, regulatory requirements, statistics, billing, and even media service like transcoding.

  SBCs are a total overkill for a world where a large number of Web applications simply want to add a WebRTC feature – probably mostly to provide a video or audio customer support service, but it could be a live training session with call-in, or an interest group conference all.

  We cannot install a custom SBC solution for every WebRTC service provider in every enterprise. That’s like saying we need a custom Web proxy for every Web server. It doesn’t scale.

  Cloud services thrive on their ability to sell directly to an individual in an organisation on their credit card without that individual having to ask their IT department to put special rules in place. WebRTC will not make progress in the corporate environment unless this is fixed.

  We need a solution that allows all WebRTC services to get through an enterprise firewall and enterprise proxy. I think the WebRTC standards have done pretty well with firewalls and connecting to a TURN server on port 443 will do the trick most of the time. But enterprise proxies are the next frontier.

  What it takes is some kind of media packet forwarding service that sits on the firewall or in a proxy and allows WebRTC media packets through – maybe with some configuration that is necessary in the browsers or the Web app to add this service as another type of TURN server.

  I don’t have a full understanding of the problems involved, but I think such a solution is vital before WebRTC can go mainstream. I expect that this year we will see some clever people coming up with a solution for this and a new type of product will be born and rolled out to enterprises around the world.

  Summary

  So these are my predictions. In summary, they address the key areas where I think WebRTC still has to make progress : interoperability between browsers, video quality at low bitrates, and the enterprise boundary. I’m really curious to see where we stand with these a year from now.

  —

  It’s worth mentioning Philipp Hancke’s tweet reply to my post :

  https://datatracker.ietf.org/doc/draft-ietf-rtcweb-return/ … — we saw some clever people come up with a solution already. Now it needs to be implemented