Clean up logging in AudioSendStream::SetupSendCodec().

webrtc/audio/audio_send_stream.cc

Index: webrtc/audio/audio_send_stream.cc
diff --git a/webrtc/audio/audio_send_stream.cc b/webrtc/audio/audio_send_stream.cc
index 8925b042b5603a0a1641237a87e0dbf7659a7ce0..5c08c9b8a2ce7cedc52bdd7ddf278b90eba138d1 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;
       }
     }