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

webrtc/modules/audio_coding/audio_network_adaptor/controller_manager.cc

 /*
  *  Copyright (c) 2016 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 124 matching lines @@
                                       const Clock* clock)
     : min_reordering_time_ms(min_reordering_time_ms),
       min_reordering_squared_distance(min_reordering_squared_distance),
       clock(clock) {}
 
 ControllerManagerImpl::Config::~Config() = default;
 
 std::unique_ptr<ControllerManager> ControllerManagerImpl::Create(
     const std::string& config_string,
     size_t num_encoder_channels,
-    rtc::ArrayView<const int> encoder_frame_lengths_ms,
+    const std::vector<int>& encoder_frame_lengths_ms,

[kwiberg-webrtc, 2016/10/19 09:12:01]

Why did you make this change?

[minyue-webrtc, 2016/10/19 09:25:54]

Because it is easier to construct a vector at the current caller.

[kwiberg-webrtc, 2016/10/19 10:43:33]

Well, marginally---given an ArrayView av, you should be able to create a vector
like this:

  std::vector<int> v(av.cbegin(), av.cend());

The advantage of using ArrayView instead of const vector& for the function
argument is that you don't force the caller to hold the data in a vector. It can
be a vector, but also e.g. an array, or anything else that stores the elements
contiguously in memory.

[minyue-webrtc, 2016/10/21 08:26:29]

Ok. thanks, yes. I'd changed it back.

     size_t intial_channels_to_encode,
     int initial_frame_length_ms,
     int initial_bitrate_bps,
     bool initial_fec_enabled,
     bool initial_dtx_enabled,
     const Clock* clock) {
 #ifdef WEBRTC_AUDIO_NETWORK_ADAPTOR_DEBUG_DUMP
   audio_network_adaptor::config::ControllerManager controller_manager_config;
   controller_manager_config.ParseFromString(config_string);
 
@@ skipping 169 matching lines @@
       NormalizeUplinkBandwidth(scoring_point.uplink_bandwidth_bps) -
       NormalizeUplinkBandwidth(uplink_bandwidth_bps);
   float diff_normalized_packet_loss =
       NormalizePacketLossFraction(scoring_point.uplink_packet_loss_fraction) -
       NormalizePacketLossFraction(uplink_packet_loss_fraction);
   return std::pow(diff_normalized_bitrate_bps, 2) +
          std::pow(diff_normalized_packet_loss, 2);
 }
 
 }  // namespace webrtc