Relanding https://codereview.webrtc.org/1715883002/ in pieces.

webrtc/voice_engine/channel.cc

Index: webrtc/voice_engine/channel.cc
diff --git a/webrtc/voice_engine/channel.cc b/webrtc/voice_engine/channel.cc
index 2c747af12ba5c71cb75202cff489735043718755..8f1d0c18bda724f40c6fd6a82ccc7b6c07347c41 100644
--- a/webrtc/voice_engine/channel.cc
+++ b/webrtc/voice_engine/channel.cc
@@ -361,7 +361,7 @@ void Channel::OnPlayTelephoneEvent(uint8_t event,
                " volume=%u)",
                event, lengthMs, volume);
 
-  if (!_playOutbandDtmfEvent || (event > 15)) {
+  if (!_playOutbandDtmfEvent || event > 15) {
     // Ignore callback since feedback is disabled or event is not a
     // Dtmf tone event.
     return;
@@ -2227,18 +2227,14 @@ int Channel::SendTelephoneEventOutband(int event, int duration_ms) {
   return 0;
 }
 
-int Channel::SetSendTelephoneEventPayloadType(unsigned char type) {
+int Channel::SetSendTelephoneEventPayloadType(int payload_type) {
   WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
                "Channel::SetSendTelephoneEventPayloadType()");
-  if (type > 127) {
-    _engineStatisticsPtr->SetLastError(
-        VE_INVALID_ARGUMENT, kTraceError,
-        "SetSendTelephoneEventPayloadType() invalid type");
-    return -1;
-  }
-  CodecInst codec = {};
+  RTC_DCHECK_LE(0, payload_type);
+  RTC_DCHECK_GE(127, payload_type);
+  CodecInst codec = {0};

[tlegrand-webrtc, 2016/03/14 12:11:13]

By removing the return -1 here, you have changed behavior in case of faulty
input. Is that handled later in  RegisterSendPayload()?

[the sun, 2016/03/14 13:21:00]

Very good point. I thought it was already capped, when I look through the SDP
handling code it becomes apparent that the payload type range is never
validated. I've added the necessary validation in WVoMC, where it should be, and
keeping the DCHECKs here.

   codec.plfreq = 8000;
-  codec.pltype = type;
+  codec.pltype = payload_type;
   memcpy(codec.plname, "telephone-event", 16);
   if (_rtpRtcpModule->RegisterSendPayload(codec) != 0) {
     _rtpRtcpModule->DeRegisterSendPayload(codec.pltype);