diff --git a/webrtc/api/BUILD.gn b/webrtc/api/BUILD.gn
index 9a286025d3..69fd7f44b2 100644
--- a/webrtc/api/BUILD.gn
+++ b/webrtc/api/BUILD.gn
@@ -21,6 +21,7 @@ group("api") {
 rtc_source_set("call_api") {
   sources = [
     "call/audio_receive_stream.h",
+    "call/audio_send_stream.cc",
     "call/audio_send_stream.h",
     "call/audio_sink.h",
     "call/audio_state.h",
diff --git a/webrtc/api/api.gyp b/webrtc/api/api.gyp
index 8a7fe5aae7..b50dfd5683 100644
--- a/webrtc/api/api.gyp
+++ b/webrtc/api/api.gyp
@@ -105,6 +105,7 @@
       ],
       'sources': [
         'call/audio_receive_stream.h',
+        'call/audio_send_stream.cc',
         'call/audio_send_stream.h',
         'call/audio_sink.h',
         'call/audio_state.h',
diff --git a/webrtc/api/call/audio_send_stream.cc b/webrtc/api/call/audio_send_stream.cc
new file mode 100644
index 0000000000..06cbc545d9
--- /dev/null
+++ b/webrtc/api/call/audio_send_stream.cc
@@ -0,0 +1,118 @@
+/*
+ *  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.
+ */
+
+#include "webrtc/api/call/audio_send_stream.h"
+
+#include <string>
+
+namespace {
+
+std::string ToString(const webrtc::CodecInst& codec_inst) {
+  std::stringstream ss;
+  ss << "{pltype: " << codec_inst.pltype;
+  ss << ", plname: \"" << codec_inst.plname << "\"";
+  ss << ", plfreq: " << codec_inst.plfreq;
+  ss << ", pacsize: " << codec_inst.pacsize;
+  ss << ", channels: " << codec_inst.channels;
+  ss << ", rate: " << codec_inst.rate;
+  ss << '}';
+  return ss.str();
+}
+}  // namespace
+
+namespace webrtc {
+
+AudioSendStream::Stats::Stats() = default;
+
+AudioSendStream::Config::Config(Transport* send_transport)
+    : send_transport(send_transport) {}
+
+std::string AudioSendStream::Config::ToString() const {
+  std::stringstream ss;
+  ss << "{rtp: " << rtp.ToString();
+  ss << ", send_transport: " << (send_transport ? "(Transport)" : "nullptr");
+  ss << ", voe_channel_id: " << voe_channel_id;
+  ss << ", min_bitrate_kbps: " << min_bitrate_kbps;
+  ss << ", max_bitrate_kbps: " << max_bitrate_kbps;
+  ss << ", send_codec_spec: " << send_codec_spec.ToString();
+  ss << '}';
+  return ss.str();
+}
+
+AudioSendStream::Config::Rtp::Rtp() = default;
+
+AudioSendStream::Config::Rtp::~Rtp() = default;
+
+std::string AudioSendStream::Config::Rtp::ToString() const {
+  std::stringstream ss;
+  ss << "{ssrc: " << ssrc;
+  ss << ", extensions: [";
+  for (size_t i = 0; i < extensions.size(); ++i) {
+    ss << extensions[i].ToString();
+    if (i != extensions.size() - 1) {
+      ss << ", ";
+    }
+  }
+  ss << ']';
+  ss << ", nack: " << nack.ToString();
+  ss << ", c_name: " << c_name;
+  ss << '}';
+  return ss.str();
+}
+
+AudioSendStream::Config::SendCodecSpec::SendCodecSpec() {
+  webrtc::CodecInst empty_inst = {0};
+  codec_inst = empty_inst;
+  codec_inst.pltype = -1;
+}
+
+std::string AudioSendStream::Config::SendCodecSpec::ToString() const {
+  std::stringstream ss;
+  ss << "{nack_enabled: " << (nack_enabled ? "true" : "false");
+  ss << ", transport_cc_enabled: " << (transport_cc_enabled ? "true" : "false");
+  ss << ", enable_codec_fec: " << (enable_codec_fec ? "true" : "false");
+  ss << ", enable_opus_dtx: " << (enable_opus_dtx ? "true" : "false");
+  ss << ", opus_max_playback_rate: " << opus_max_playback_rate;
+  ss << ", cng_payload_type: " << cng_payload_type;
+  ss << ", cng_plfreq: " << cng_plfreq;
+  ss << ", codec_inst: " << ::ToString(codec_inst);
+  ss << '}';
+  return ss.str();
+}
+
+bool AudioSendStream::Config::SendCodecSpec::operator==(
+    const AudioSendStream::Config::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;
+}
+}  // namespace webrtc
diff --git a/webrtc/api/call/audio_send_stream.h b/webrtc/api/call/audio_send_stream.h
index 1956b97bd4..7ff791e62a 100644
--- a/webrtc/api/call/audio_send_stream.h
+++ b/webrtc/api/call/audio_send_stream.h
@@ -30,6 +30,8 @@ namespace webrtc {
 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;
@@ -52,13 +54,13 @@ class AudioSendStream {
 
   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.
@@ -91,40 +93,10 @@ class AudioSendStream {
     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);
       }
diff --git a/webrtc/audio/audio_send_stream.cc b/webrtc/audio/audio_send_stream.cc
index 8925b042b5..5c08c9b8a2 100644
--- a/webrtc/audio/audio_send_stream.cc
+++ b/webrtc/audio/audio_send_stream.cc
@@ -35,56 +35,11 @@ namespace {
 
 constexpr char kOpusCodecName[] = "opus";
 
-// TODO(minyue): Remove |LOG_RTCERR2|.
-#define LOG_RTCERR2(func, a1, a2, err)                      \
-  LOG(LS_WARNING) << "" << #func << "(" << a1 << ", " << a2 \
-                  << ") failed, err=" << err
-
-// TODO(minyue): Remove |LOG_RTCERR3|.
-#define LOG_RTCERR3(func, a1, a2, a3, err)                                \
-  LOG(LS_WARNING) << "" << #func << "(" << a1 << ", " << a2 << ", " << a3 \
-                  << ") failed, err=" << err
-
-std::string ToString(const webrtc::CodecInst& codec) {
-  std::stringstream ss;
-  ss << codec.plname << "/" << codec.plfreq << "/" << codec.channels << " ("
-     << codec.pltype << ")";
-  return ss.str();
-}
-
 bool IsCodec(const webrtc::CodecInst& codec, const char* ref_name) {
   return (_stricmp(codec.plname, ref_name) == 0);
 }
-
 }  // namespace
 
-std::string AudioSendStream::Config::Rtp::ToString() const {
-  std::stringstream ss;
-  ss << "{ssrc: " << ssrc;
-  ss << ", extensions: [";
-  for (size_t i = 0; i < extensions.size(); ++i) {
-    ss << extensions[i].ToString();
-    if (i != extensions.size() - 1) {
-      ss << ", ";
-    }
-  }
-  ss << ']';
-  ss << ", nack: " << nack.ToString();
-  ss << ", c_name: " << c_name;
-  ss << '}';
-  return ss.str();
-}
-
-std::string AudioSendStream::Config::ToString() const {
-  std::stringstream ss;
-  ss << "{rtp: " << rtp.ToString();
-  ss << ", voe_channel_id: " << voe_channel_id;
-  // TODO(solenberg): Encoder config.
-  ss << ", cng_payload_type: " << send_codec_spec.cng_payload_type;
-  ss << '}';
-  return ss.str();
-}
-
 namespace internal {
 AudioSendStream::AudioSendStream(
     const webrtc::AudioSendStream::Config& config,
@@ -333,11 +288,8 @@ bool AudioSendStream::SetupSendCodec() {
 
   const auto& send_codec_spec = config_.send_codec_spec;
 
-  // Set the codec immediately, since SetVADStatus() depends on whether
-  // the current codec is mono or stereo.
-  LOG(LS_INFO) << "Send channel " << channel << " selected voice codec "
-               << ToString(send_codec_spec.codec_inst)
-               << ", bitrate=" << send_codec_spec.codec_inst.rate;
+  // We set the codec first, since the below extra configuration is only applied
+  // to the "current" codec.
 
   // If codec is already configured, we do not it again.
   // TODO(minyue): check if this check is really needed, or can we move it into
@@ -346,47 +298,33 @@ bool AudioSendStream::SetupSendCodec() {
   if (codec->GetSendCodec(channel, current_codec) != 0 ||
       (send_codec_spec.codec_inst != current_codec)) {
     if (codec->SetSendCodec(channel, send_codec_spec.codec_inst) == -1) {
-      LOG_RTCERR2(SetSendCodec, channel, ToString(send_codec_spec.codec_inst),
-                  base->LastError());
+      LOG(LS_WARNING) << "SetSendCodec() failed: " << base->LastError();
       return false;
     }
   }
 
-  // FEC should be enabled after SetSendCodec.
+  // Codec internal FEC. Treat any failure as fatal internal error.
   if (send_codec_spec.enable_codec_fec) {
-    LOG(LS_INFO) << "Attempt to enable codec internal FEC on channel "
-                 << channel;
-    if (codec->SetFECStatus(channel, true) == -1) {
-      // Enable codec internal FEC. Treat any failure as fatal internal error.
-      LOG_RTCERR2(SetFECStatus, channel, true, base->LastError());
+    if (codec->SetFECStatus(channel, true) != 0) {
+      LOG(LS_WARNING) << "SetFECStatus() failed: " << base->LastError();
       return false;
     }
   }
 
+  // DTX and maxplaybackrate are only set if current codec is Opus.
   if (IsCodec(send_codec_spec.codec_inst, kOpusCodecName)) {
-    // DTX and maxplaybackrate should be set after SetSendCodec. Because current
-    // send codec has to be Opus.
-
-    // Set Opus internal DTX.
-    LOG(LS_INFO) << "Attempt to "
-                 << (send_codec_spec.enable_opus_dtx ? "enable" : "disable")
-                 << " Opus DTX on channel " << channel;
-    if (codec->SetOpusDtx(channel, send_codec_spec.enable_opus_dtx)) {
-      LOG_RTCERR2(SetOpusDtx, channel, send_codec_spec.enable_opus_dtx,
-                  base->LastError());
+    if (codec->SetOpusDtx(channel, send_codec_spec.enable_opus_dtx) != 0) {
+      LOG(LS_WARNING) << "SetOpusDtx() failed: " << base->LastError();
       return false;
     }
 
     // If opus_max_playback_rate <= 0, the default maximum playback rate
     // (48 kHz) will be used.
     if (send_codec_spec.opus_max_playback_rate > 0) {
-      LOG(LS_INFO) << "Attempt to set maximum playback rate to "
-                   << send_codec_spec.opus_max_playback_rate
-                   << " Hz on channel " << channel;
       if (codec->SetOpusMaxPlaybackRate(
-              channel, send_codec_spec.opus_max_playback_rate) == -1) {
-        LOG_RTCERR2(SetOpusMaxPlaybackRate, channel,
-                    send_codec_spec.opus_max_playback_rate, base->LastError());
+              channel, send_codec_spec.opus_max_playback_rate) != 0) {
+        LOG(LS_WARNING) << "SetOpusMaxPlaybackRate() failed: "
+                        << base->LastError();
         return false;
       }
     }
@@ -409,11 +347,9 @@ bool AudioSendStream::SetupSendCodec() {
           return false;
       }
       if (codec->SetSendCNPayloadType(channel, send_codec_spec.cng_payload_type,
-                                      cn_freq) == -1) {
-        LOG_RTCERR3(SetSendCNPayloadType, channel,
-                    send_codec_spec.cng_payload_type, cn_freq,
-                    base->LastError());
-
+                                      cn_freq) != 0) {
+        LOG(LS_WARNING) << "SetSendCNPayloadType() failed: "
+                        << base->LastError();
         // TODO(ajm): This failure condition will be removed from VoE.
         // Restore the return here when we update to a new enough webrtc.
         //
@@ -431,9 +367,8 @@ bool AudioSendStream::SetupSendCodec() {
         send_codec_spec.codec_inst.channels == 1) {
       // TODO(minyue): If CN frequency == 48000 Hz is allowed, consider the
       // interaction between VAD and Opus FEC.
-      LOG(LS_INFO) << "Enabling VAD";
-      if (codec->SetVADStatus(channel, true) == -1) {
-        LOG_RTCERR2(SetVADStatus, channel, true, base->LastError());
+      if (codec->SetVADStatus(channel, true) != 0) {
+        LOG(LS_WARNING) << "SetVADStatus() failed: " << base->LastError();
         return false;
       }
     }
diff --git a/webrtc/audio/audio_send_stream_unittest.cc b/webrtc/audio/audio_send_stream_unittest.cc
index 14ca0e076f..a2832ded00 100644
--- a/webrtc/audio/audio_send_stream_unittest.cc
+++ b/webrtc/audio/audio_send_stream_unittest.cc
@@ -219,12 +219,26 @@ TEST(AudioSendStreamTest, ConfigToString) {
       RtpExtension(RtpExtension::kAbsSendTimeUri, kAbsSendTimeId));
   config.rtp.c_name = kCName;
   config.voe_channel_id = kChannelId;
+  config.min_bitrate_kbps = 12;
+  config.max_bitrate_kbps = 34;
+  config.send_codec_spec.nack_enabled = true;
+  config.send_codec_spec.transport_cc_enabled = false;
+  config.send_codec_spec.enable_codec_fec = true;
+  config.send_codec_spec.enable_opus_dtx = false;
+  config.send_codec_spec.opus_max_playback_rate = 32000;
   config.send_codec_spec.cng_payload_type = 42;
+  config.send_codec_spec.cng_plfreq = 56;
+  config.send_codec_spec.codec_inst = kIsacCodec;
   EXPECT_EQ(
       "{rtp: {ssrc: 1234, extensions: [{uri: "
       "http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time, id: 3}], "
-      "nack: {rtp_history_ms: 0}, c_name: foo_name}, voe_channel_id: 1, "
-      "cng_payload_type: 42}",
+      "nack: {rtp_history_ms: 0}, c_name: foo_name}, send_transport: nullptr, "
+      "voe_channel_id: 1, min_bitrate_kbps: 12, max_bitrate_kbps: 34, "
+      "send_codec_spec: {nack_enabled: true, transport_cc_enabled: false, "
+      "enable_codec_fec: true, enable_opus_dtx: false, opus_max_playback_rate: "
+      "32000, cng_payload_type: 42, cng_plfreq: 56, codec_inst: {pltype: "
+      "103, plname: \"isac\", plfreq: 16000, pacsize: 320, channels: 1, rate: "
+      "32000}}}",
       config.ToString());
 }
 
diff --git a/webrtc/config.cc b/webrtc/config.cc
index 04f6a66935..281b89f46e 100644
--- a/webrtc/config.cc
+++ b/webrtc/config.cc
@@ -31,6 +31,11 @@ std::string UlpfecConfig::ToString() const {
   return ss.str();
 }
 
+FlexfecConfig::FlexfecConfig()
+    : flexfec_payload_type(-1), flexfec_ssrc(0), protected_media_ssrcs() {}
+
+FlexfecConfig::~FlexfecConfig() = default;
+
 std::string FlexfecConfig::ToString() const {
   std::stringstream ss;
   ss << "{flexfec_payload_type: " << flexfec_payload_type;
@@ -134,6 +139,8 @@ VideoEncoderConfig::VideoEncoderConfig()
       max_bitrate_bps(0),
       number_of_streams(0) {}
 
+VideoEncoderConfig::VideoEncoderConfig(VideoEncoderConfig&&) = default;
+
 VideoEncoderConfig::~VideoEncoderConfig() = default;
 
 std::string VideoEncoderConfig::ToString() const {
@@ -155,6 +162,8 @@ std::string VideoEncoderConfig::ToString() const {
   return ss.str();
 }
 
+VideoEncoderConfig::VideoEncoderConfig(const VideoEncoderConfig&) = default;
+
 void VideoEncoderConfig::EncoderSpecificSettings::FillEncoderSpecificSettings(
     VideoCodec* codec) const {
   if (codec->codecType == kVideoCodecH264) {
@@ -210,4 +219,8 @@ void VideoEncoderConfig::Vp9EncoderSpecificSettings::FillVideoCodecVp9(
   *vp9_settings = specifics_;
 }
 
+DecoderSpecificSettings::DecoderSpecificSettings() = default;
+
+DecoderSpecificSettings::~DecoderSpecificSettings() = default;
+
 }  // namespace webrtc
diff --git a/webrtc/config.h b/webrtc/config.h
index 4ec895b0dd..3b3c3d90a8 100644
--- a/webrtc/config.h
+++ b/webrtc/config.h
@@ -57,8 +57,8 @@ struct UlpfecConfig {
 // Settings for FlexFEC forward error correction.
 // Set the payload type to '-1' to disable.
 struct FlexfecConfig {
-  FlexfecConfig()
-      : flexfec_payload_type(-1), flexfec_ssrc(0), protected_media_ssrcs() {}
+  FlexfecConfig();
+  ~FlexfecConfig();
   std::string ToString() const;
 
   // Payload type of FlexFEC.
@@ -163,7 +163,7 @@ class VideoEncoderConfig {
     virtual void FillVideoCodecVp9(VideoCodecVP9* vp9_settings) const;
     virtual void FillVideoCodecH264(VideoCodecH264* h264_settings) const;
    private:
-    virtual ~EncoderSpecificSettings() {}
+    ~EncoderSpecificSettings() override {}
     friend class VideoEncoderConfig;
   };
 
@@ -211,7 +211,7 @@ class VideoEncoderConfig {
         const VideoEncoderConfig& encoder_config) = 0;
 
    protected:
-    virtual ~VideoStreamFactoryInterface() {}
+    ~VideoStreamFactoryInterface() override {}
   };
 
   VideoEncoderConfig& operator=(VideoEncoderConfig&&) = default;
@@ -221,7 +221,7 @@ class VideoEncoderConfig {
   VideoEncoderConfig Copy() const { return VideoEncoderConfig(*this); }
 
   VideoEncoderConfig();
-  VideoEncoderConfig(VideoEncoderConfig&&) = default;
+  VideoEncoderConfig(VideoEncoderConfig&&);
   ~VideoEncoderConfig();
   std::string ToString() const;
 
@@ -243,7 +243,7 @@ class VideoEncoderConfig {
  private:
   // Access to the copy constructor is private to force use of the Copy()
   // method for those exceptional cases where we do use it.
-  VideoEncoderConfig(const VideoEncoderConfig&) = default;
+  VideoEncoderConfig(const VideoEncoderConfig&);
 };
 
 struct VideoDecoderH264Settings {
@@ -252,7 +252,8 @@ struct VideoDecoderH264Settings {
 
 class DecoderSpecificSettings {
  public:
-  virtual ~DecoderSpecificSettings() {}
+  DecoderSpecificSettings();
+  virtual ~DecoderSpecificSettings();
   rtc::Optional<VideoDecoderH264Settings> h264_extra_settings;
 };
 
diff --git a/webrtc/logging/BUILD.gn b/webrtc/logging/BUILD.gn
index fa28065373..54daa20929 100644
--- a/webrtc/logging/BUILD.gn
+++ b/webrtc/logging/BUILD.gn
@@ -32,6 +32,7 @@ rtc_static_library("rtc_event_log_impl") {
   deps = [
     ":rtc_event_log_api",
     "..:webrtc_common",
+    "../api:call_api",
     "../modules/rtp_rtcp",
   ]
 
diff --git a/webrtc/webrtc.gyp b/webrtc/webrtc.gyp
index 98d9498366..1447a99183 100644
--- a/webrtc/webrtc.gyp
+++ b/webrtc/webrtc.gyp
@@ -67,6 +67,7 @@
           'dependencies': [
             'rtc_event_log_api',
             'rtc_event_log_proto',
+            '<(webrtc_root)/api/api.gyp:call_api',
           ],
           'defines': [
             'ENABLE_RTC_EVENT_LOG',