Makes padding prefer video SSRCs instead of audio.

Some clients will not count audio packets into the bandwidth estimate
despite negotiating e.g. abs-send-time for that SSRC.
If padding is sent on such an RTP module, we might get stuck in a low
resolution.

This CL works around that by preferring to send padding on video SSRCs.

Bug: webrtc:11196
Change-Id: I1ff503a31a85bc32315006a4f15f8b08e5d4e883
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161941
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30066}

modules/rtp_rtcp/include/rtp_rtcp.h

@@ -250,6 +250,9 @@ class RtpRtcp : public Module, public RtcpFeedbackSenderInterface {
   // Returns current media sending status.
   virtual bool SendingMedia() const = 0;
 
+  // Returns whether audio is configured (i.e. Configuration::audio = true).
+  virtual bool IsAudioConfigured() const = 0;
+
   // Indicate that the packets sent by this module should be counted towards the
   // bitrate estimate since the stream participates in the bitrate allocation.
   virtual void SetAsPartOfAllocation(bool part_of_allocation) = 0;