Make AudioEncoder::GetFrameLengthRange() pure virtual.

In order for WebRTC to be able to include packet overhead in its
bitrate calculations, the AudioEncoder::GetFrameLengthRange()
function must be implemented by all audio encoders. Making this
member function pure virtual as per the following PSA:

https://groups.google.com/forum/#!topic/discuss-webrtc/qscwYr38je0

Bug: webrtc:11427
Change-Id: I30d297ef05f57453bfc257624729559057cad118
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171517
Commit-Queue: Ali Tofigh <alito@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31127}

api/audio_codecs/audio_encoder.cc

@@ -110,9 +110,4 @@ ANAStats AudioEncoder::GetANAStats() const {
   return ANAStats();
 }
 
-absl::optional<std::pair<TimeDelta, TimeDelta>>
-AudioEncoder::GetFrameLengthRange() const {
-  return absl::nullopt;
-}
-
 }  // namespace webrtc

api/audio_codecs/audio_encoder.h

@@ -244,7 +244,7 @@ class AudioEncoder {
   // information. This is used to calculated the full bitrate range, including
   // overhead.
   virtual absl::optional<std::pair<TimeDelta, TimeDelta>> GetFrameLengthRange()
-      const;
+      const = 0;
 
  protected:
   // Subclasses implement this to perform the actual encoding. Called by