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

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

 /*
  *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 31 matching lines @@
 
     int frame_size_ms = 20;
     size_t num_channels = 1;
     int payload_type = 120;
     ApplicationMode application = kVoip;
     rtc::Optional<int> bitrate_bps;  // Unset means to use default value.
     bool fec_enabled = false;
     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:
 #if defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS) || defined(WEBRTC_ARCH_ARM)
     // If we are on Android, iOS and/or ARM, use a lower complexity setting as
     // default, to save encoder complexity.
     static const int kDefaultComplexity = 5;
 #else
     static const int kDefaultComplexity = 9;
 #endif
@@ skipping 33 matching lines @@
   bool EnableAudioNetworkAdaptor(const std::string& config_string,
                                  const Clock* clock) override;
   void DisableAudioNetworkAdaptor() override;
   void OnReceivedUplinkBandwidth(int uplink_bandwidth_bps) override;
   void OnReceivedUplinkPacketLossFraction(
       float uplink_packet_loss_fraction) override;
   void OnReceivedTargetAudioBitrate(int target_audio_bitrate_bps) override;
   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_; }
   ApplicationMode application() const { return config_.application; }
   bool fec_enabled() const { return config_.fec_enabled; }
   size_t num_channels_to_encode() const { return num_channels_to_encode_; }
   int next_frame_length_ms() const { return next_frame_length_ms_; }
 
  protected:
   EncodedInfo EncodeImpl(uint32_t rtp_timestamp,
@@ skipping 24 matching lines @@
   std::unique_ptr<PacketLossFractionSmoother> packet_loss_fraction_smoother_;
   AudioNetworkAdaptorCreator audio_network_adaptor_creator_;
   std::unique_ptr<AudioNetworkAdaptor> audio_network_adaptor_;
 
   RTC_DISALLOW_COPY_AND_ASSIGN(AudioEncoderOpus);
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_AUDIO_ENCODER_OPUS_H_