This a pre-step for improving perfomance of the RTCPReceiver
- rest of the ReceiveInfo is tmmbr related and
can be handled only when tmmbr is explicitly enabled.
BUG=webrtc:5565
Review-Url: https://codereview.webrtc.org/2681003003
Cr-Commit-Position: refs/heads/master@{#16667}
We support multiple payload types, and one which matches the audio codec the closest, is picked (or the one with lowest clock rate, if no perfect match is found).
The exact clock rate is then ignored and DTMF packets are time stamped with the rate of the current audio codec. This is exactly the way the code has worked up to this point, but until now we have been under the impression that we were in fact sending 8k DTMF.
In other words, this is an improvement over the current situation, since we will most likely find a payload type which matches the codec clock rate.
This CL also does a little cleaning of the DTMFQueue and RTPSenderAudio classes.
BUG=webrtc:2795
Review-Url: https://codereview.webrtc.org/2392883002
Cr-Commit-Position: refs/heads/master@{#15129}
rtcp_utility, rtp_utility, tmmbr_help, rtcp_receiver, rtcp_receiver_help are explicetly excluded from the cleanup becaues there are short plans (or cls) to do a deeper cleaning there.
BUG=webrtc:5277
R=pbos@webrtc.org, mflodman@webrtc.org
Review URL: https://codereview.webrtc.org/1512493002
Cr-Commit-Position: refs/heads/master@{#10966}
This fixes a variety of MSVC warnings about value truncations when implicitly
storing the 64-bit values we get back from e.g. TimeTicks in 32-bit objects, and
removes the need for a number of explicit casts.
This also moves a number of constants so they're declared right where they're used, which is easier to read and maintain, and makes some of them of integral type rather than using the "enum hack".
BUG=chromium:81439
TEST=none
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/33649004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7905 4adac7df-926f-26a2-2b94-8c16560cd09d
