Simplifying audio network adaptor by moving receiver frame length range to ctor.

webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h

Index: webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h
diff --git a/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h b/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h
index 342668e78002976165d3d453256cc321680fb606..81ca17f7e08bf14d4577de9332f49f9871eb0cf5 100644
--- a/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h
+++ b/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h
@@ -49,6 +49,7 @@ class AudioEncoderOpus final : public AudioEncoder {
     int max_playback_rate_hz = 48000;
     int complexity = kDefaultComplexity;
     bool dtx_enabled = false;
+    std::vector<int> supported_frame_lengths_ms;
     const Clock* clock = nullptr;
 
    private:
@@ -102,6 +103,9 @@ class AudioEncoderOpus final : public AudioEncoder {
   void OnReceivedRtt(int rtt_ms) override;
   void SetReceiverFrameLengthRange(int min_frame_length_ms,
                                    int max_frame_length_ms) override;
+  rtc::ArrayView<const int> supported_frame_lengths_ms() const {
+    return config_.supported_frame_lengths_ms;
+  }
 
   // Getters for testing.
   double packet_loss_rate() const { return packet_loss_rate_; }