Enable End-to-End Encrypted Audio Payloads.

This change integrates the FrameDecryptorInterface and the FrameEncryptorInterface into
the audio media path. If a FrameEncryptorInterface is set on an outgoing audio RTPSender
then each outgoing audio payload will first pass through the provided FrameEncryptor which
will have a chance to modify the payload contents for the purposes of encryption.

If a FrameDecryptorInterface is set on an incoming audio RtpReceiver then each incoming
audio payload will first pass through the provided FrameDecryptor which have a chance to
modify the payload contents for the purpose of decryption.

While AEAD is supported by the FrameDecryptor/FrameEncryptor interfaces this CL does not
use it and so it is left as null.

Bug: webrtc:9681
Change-Id: Ic383a9dce280528739f9d271357c2220e0a0dccf
Reviewed-on: https://webrtc-review.googlesource.com/c/101702
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Emad Omara <emadomara@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25001}

api/BUILD.gn

@@ -446,6 +446,23 @@ if (rtc_include_tests) {
     ]
   }
 
+  rtc_source_set("fake_frame_crypto") {
+    testonly = true
+    sources = [
+      "test/fake_frame_decryptor.cc",
+      "test/fake_frame_decryptor.h",
+      "test/fake_frame_encryptor.cc",
+      "test/fake_frame_encryptor.h",
+    ]
+    deps = [
+      ":array_view",
+      ":libjingle_peerconnection_api",
+      "..:webrtc_common",
+      "../rtc_base:checks",
+      "../rtc_base:rtc_base_approved",
+    ]
+  }
+
   rtc_source_set("mock_peerconnectioninterface") {
     testonly = true
     sources = [