Full impl of NnChannel::SetSendParameters and NnChannel::SetRecvParameters

talk/media/webrtc/webrtcvoiceengine.cc

 /*
  * libjingle
  * Copyright 2004 Google Inc.
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions are met:
  *
  *  1. Redistributions of source code must retain the above copyright notice,
  *     this list of conditions and the following disclaimer.
  *  2. Redistributions in binary form must reproduce the above copyright notice,
@@ skipping 563 matching lines @@
   voe_wrapper_->base()->Terminate();
   desired_local_monitor_enable_ = false;
 }
 
 int WebRtcVoiceEngine::GetCapabilities() {
   return AUDIO_SEND | AUDIO_RECV;
 }
 
 VoiceMediaChannel* WebRtcVoiceEngine::CreateChannel(
     const AudioOptions& options) {
-  WebRtcVoiceMediaChannel* ch = new WebRtcVoiceMediaChannel(this);
+  WebRtcVoiceMediaChannel* ch = new WebRtcVoiceMediaChannel(this, options);
   if (!ch->valid()) {
     delete ch;
     return nullptr;
   }
-  if (!ch->SetOptions(options)) {
-    LOG(LS_WARNING) << "Failed to set options while creating channel.";
-  }
   return ch;
 }
 
 bool WebRtcVoiceEngine::SetOptions(const AudioOptions& options) {
   if (!ApplyOptions(options)) {
     return false;
   }
   options_ = options;
   return true;
 }
@@ skipping 1080 matching lines @@
   // Raw pointer to AudioRenderer owned by LocalAudioTrackHandler.
   // PeerConnection will make sure invalidating the pointer before the object
   // goes away.
   AudioRenderer* renderer_;
 
   // Protects |renderer_| in Start(), Stop() and OnClose().
   rtc::CriticalSection lock_;
 };
 
 // WebRtcVoiceMediaChannel
-WebRtcVoiceMediaChannel::WebRtcVoiceMediaChannel(WebRtcVoiceEngine* engine)
+WebRtcVoiceMediaChannel::WebRtcVoiceMediaChannel(WebRtcVoiceEngine* engine,
+                                                 const AudioOptions& options)
     : engine_(engine),
       voe_channel_(engine->CreateMediaVoiceChannel()),
       send_bitrate_setting_(false),
       send_bitrate_bps_(0),
       options_(),
       dtmf_allowed_(false),
       desired_playout_(false),
       nack_enabled_(false),
       playout_(false),
       typing_noise_detected_(false),
       desired_send_(SEND_NOTHING),
       send_(SEND_NOTHING),
       call_(nullptr),
       default_receive_ssrc_(0) {
   engine->RegisterChannel(this);
   LOG(LS_VERBOSE) << "WebRtcVoiceMediaChannel::WebRtcVoiceMediaChannel "
                   << voe_channel();
   ConfigureSendChannel(voe_channel());
+  SetOptions(options);
 }
 
 WebRtcVoiceMediaChannel::~WebRtcVoiceMediaChannel() {
   LOG(LS_VERBOSE) << "WebRtcVoiceMediaChannel::~WebRtcVoiceMediaChannel "
                   << voe_channel();
   DCHECK(receive_streams_.empty() || call_);
 
   // Remove any remaining send streams, the default channel will be deleted
   // later.
   while (!send_channels_.empty())
@@ skipping 365 matching lines @@
           return false;
         }
       }
     }
   }
   return true;
 }
 
 bool WebRtcVoiceMediaChannel::SetSendCodecs(
     const std::vector<AudioCodec>& codecs) {
+  // Don't change any state if codec list is empty.
+  if (codecs.empty()) {
+    return true;
+  }

[pthatcher1, 2015/09/14 19:42:02]

Same here

+
   dtmf_allowed_ = false;
   for (const AudioCodec& codec : codecs) {
     // Find the DTMF telephone event "codec".
     if (IsCodec(codec, kDtmfCodecName)) {
       dtmf_allowed_ = true;
     }
   }
 
   // Cache the codecs in order to configure the channel created later.
   send_codecs_ = codecs;
@@ skipping 1533 matching lines @@
 
 int WebRtcSoundclipStream::Rewind() {
   mem_.Rewind();
   // Return -1 to keep VoiceEngine from looping.
   return (loop_) ? 0 : -1;
 }
 
 }  // namespace cricket
 
 #endif  // HAVE_WEBRTC_VOICE