Clean up logging in AudioSendStream::SetupSendCodec().

webrtc/api/call/audio_send_stream.cc

Index: webrtc/api/call/audio_send_stream.cc
diff --git a/webrtc/api/call/audio_send_stream.cc b/webrtc/api/call/audio_send_stream.cc
new file mode 100644
index 0000000000000000000000000000000000000000..06cbc545d9313846b057ed3432e2862c5c8b9b14
--- /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