Add set_sender_codec

[mickel8]

This PR allows to set codec that should be used for sending with the following remarks:

• when the first RTP packet is sent, the clock rate used by RTP sender is "locked" meaning that any subsequent call to the set_sender_codec must use a codec with the same clock rate. This is because using a new clock rate requires a new SSRC. Also, ReportRecorder, which generates RTCP reports relies on clock rate so changing it after we started sending packets could result in incorrect reports
• when a codec was selected, and as a result of renegotiation it is no longer supported, user must call set_sender_codec with a new codec that is supported

It also:

• moves initialization of ReportRecorder so that now, it is initialized when the first packet is sent
• adds checking FMTP parameters when negotiating codecs (if codec's sdp_fmtp_line is not nil, otherwise sdp_fmtp_line is always assumed to be the same)

RFC considerations:

• Codecs listed in the remote description are in the preference order i.e. the first codec is the most preferred to receive by the remote side, and we as a sender SHOULD use it for sending.
• However, RFC 3264 (sec 6.1 and 7.0) allows for changing codec on the fly (e.g. during silence periods although Opus solves this problem and I have never seen anyone doing this) and receiver MUST be prepared to handle this situation
• Relying on these requirements, we allow user to set the codec used for sending

[codecov]

Codecov Report

Attention: Patch coverage is 89.85507% with 7 lines in your changes missing coverage. Please review.

Project coverage is 84.88%. Comparing base (7f3b1bd) to head (61e3a04).

Files with missing lines	Patch %	Lines
lib/ex_webrtc/rtp_sender.ex	90.32%	3 Missing ⚠️
lib/ex_webrtc/rtp_transceiver.ex	66.66%	3 Missing ⚠️
lib/ex_webrtc/peer_connection.ex	91.66%	1 Missing ⚠️

Additional details and impacted files

@@            Coverage Diff             @@
##           master     #190      +/-   ##
==========================================
+ Coverage   84.75%   84.88%   +0.13%     
==========================================
  Files          49       49              
  Lines        2512     2554      +42     
==========================================
+ Hits         2129     2168      +39     
- Misses        383      386       +3     

Files with missing lines	Coverage Δ
lib/ex_webrtc/peer_connection/configuration.ex	95.20% <100.00%> (+0.31%)	⬆️
lib/ex_webrtc/rtp_sender/report_recorder.ex	97.05% <100.00%> (+0.08%)	⬆️
lib/ex_webrtc/peer_connection.ex	86.12% <91.66%> (+0.07%)	⬆️
lib/ex_webrtc/rtp_sender.ex	93.51% <90.32%> (+0.11%)	⬆️
lib/ex_webrtc/rtp_transceiver.ex	89.41% <66.66%> (-0.95%)	⬇️

... and 1 file with indirect coverage changes

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[mickel8]

Clock rate of ReportRecorder shouldn't change as it may result in incorrect RTCP reports. Instead, we probably should generate a new ssrc and start sending new RTCP reports. Anyways, we don't allow for codec change that has different clock rate so we can safely delete this code