Set the audio MID before creating send streams

The MID reached WebRtcVoiceSendChannel only with the send parameters. An
offerer applies those with the remote answer, but adds its send streams
with the local offer, so its audio send streams were created without a
MID. The MID was then set on the RTP module later, which is one of the
reasons why RTPSender's MID is mutable.

Add VoiceMediaSendChannelInterface::SetMid() and call it from
BaseChannel::SetLocalContent_w() before the send streams are added, so
that they're always created with the MID. A channel's MID never changes,
so this prepares for passing the MID to the RTP module at construction.

Video doesn't need this. It creates the underlying send stream once the
send parameters, including the MID, have been applied, and recreates it
when the MID changes.

Bug: webrtc:42223727
Change-Id: Iea1cbcd9b2bedd77b7cdd3673263f9491800eca9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/507360
Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#48844}