Clean up logging in AudioSendStream::SetupSendCodec().

webrtc/api/call/audio_send_stream.h

 /*
  *  Copyright (c) 2015 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 12 matching lines @@
 namespace webrtc {
 
 // WORK IN PROGRESS
 // This class is under development and is not yet intended for for use outside
 // of WebRtc/Libjingle. Please use the VoiceEngine API instead.
 // See: https://bugs.chromium.org/p/webrtc/issues/detail?id=4690
 
 class AudioSendStream {
  public:
   struct Stats {
+    Stats();
+
     // TODO(solenberg): Harmonize naming and defaults with receive stream stats.
     uint32_t local_ssrc = 0;
     int64_t bytes_sent = 0;
     int32_t packets_sent = 0;
     int32_t packets_lost = -1;
     float fraction_lost = -1.0f;
     std::string codec_name;
     int32_t ext_seqnum = -1;
     int32_t jitter_ms = -1;
     int64_t rtt_ms = -1;
     int32_t audio_level = -1;
     float aec_quality_min = -1.0f;
     int32_t echo_delay_median_ms = -1;
     int32_t echo_delay_std_ms = -1;
     int32_t echo_return_loss = -100;
     int32_t echo_return_loss_enhancement = -100;
     float residual_echo_likelihood = -1.0f;
     bool typing_noise_detected = false;
   };
 
   struct Config {
     Config() = delete;
-    explicit Config(Transport* send_transport)
-        : send_transport(send_transport) {}
-
+    explicit Config(Transport* send_transport);
     std::string ToString() const;
 
     // Send-stream specific RTP settings.
     struct Rtp {
+      Rtp();
+      ~Rtp();
       std::string ToString() const;
 
       // Sender SSRC.
       uint32_t ssrc = 0;
 
       // RTP header extensions used for the sent stream.
       std::vector<RtpExtension> extensions;
 
       // See NackConfig for description.
       NackConfig nack;
@@ skipping 12 matching lines @@
     // of Call.
     int voe_channel_id = -1;
 
     // Bitrate limits used for variable audio bitrate streams. Set both to -1 to
     // disable audio bitrate adaptation.
     // Note: This is still an experimental feature and not ready for real usage.
     int min_bitrate_kbps = -1;
     int max_bitrate_kbps = -1;
 
     struct SendCodecSpec {
-      SendCodecSpec() {
-        webrtc::CodecInst empty_inst = {0};
-        codec_inst = empty_inst;
-        codec_inst.pltype = -1;
-      }
-      bool operator==(const SendCodecSpec& rhs) const {
-        {
-          if (nack_enabled != rhs.nack_enabled) {
-            return false;
-          }
-          if (transport_cc_enabled != rhs.transport_cc_enabled) {
-            return false;
-          }
-          if (enable_codec_fec != rhs.enable_codec_fec) {
-            return false;
-          }
-          if (enable_opus_dtx != rhs.enable_opus_dtx) {
-            return false;
-          }
-          if (opus_max_playback_rate != rhs.opus_max_playback_rate) {
-            return false;
-          }
-          if (cng_payload_type != rhs.cng_payload_type) {
-            return false;
-          }
-          if (cng_plfreq != rhs.cng_plfreq) {
-            return false;
-          }
-          if (codec_inst != rhs.codec_inst) {
-            return false;
-          }
-          return true;
-        }
-      }
+      SendCodecSpec();
+      std::string ToString() const;
+
+      bool operator==(const SendCodecSpec& rhs) const;
       bool operator!=(const SendCodecSpec& rhs) const {
         return !(*this == rhs);
       }
 
       bool nack_enabled = false;
       bool transport_cc_enabled = false;
       bool enable_codec_fec = false;
       bool enable_opus_dtx = false;
       int opus_max_playback_rate = 0;
       int cng_payload_type = -1;
@@ skipping 16 matching lines @@
   virtual void SetMuted(bool muted) = 0;
 
   virtual Stats GetStats() const = 0;
 
  protected:
   virtual ~AudioSendStream() {}
 };
 }  // namespace webrtc
 
 #endif  // WEBRTC_API_CALL_AUDIO_SEND_STREAM_H_