Delete remnants of RTX support in voice_engine.

Receive logic in voe::Channel attempted to handle RTX if
RTPPayloadRegistry::IsRtx returns true, but the audio code never calls
the config methods (SetRtxSsrc or SetRtxPayloadType) required for that
to ever happen.

BUG=webrtc:7135

Review-Url: https://codereview.webrtc.org/3006913002
Cr-Original-Commit-Position: refs/heads/master@{#19633}
Cr-Mirrored-From: https://chromium.googlesource.com/external/webrtc
Cr-Mirrored-Commit: da194e79c4a86869e8afccf9827c716f0b073fb1
diff --git a/voice_engine/channel.cc b/voice_engine/channel.cc
index 5732e8e..901d80c 100644
--- a/voice_engine/channel.cc
+++ b/voice_engine/channel.cc
@@ -905,7 +905,6 @@
       transport_overhead_per_packet_(0),
       rtp_overhead_per_packet_(0),
       _outputSpeechType(AudioFrame::kNormalSpeech),
-      restored_packet_in_use_(false),
       rtcp_observer_(new VoERtcpObserver(this)),
       associate_send_channel_(ChannelOwner(nullptr)),
       pacing_enabled_(config.enable_voice_pacing),
@@ -1747,9 +1746,6 @@
                             size_t packet_length,
                             const RTPHeader& header,
                             bool in_order) {
-  if (rtp_payload_registry_->IsRtx(header)) {
-    return HandleRtxPacket(packet, packet_length, header);
-  }
   const uint8_t* payload = packet + header.headerLength;
   assert(packet_length >= header.headerLength);
   size_t payload_length = packet_length - header.headerLength;
@@ -1762,35 +1758,6 @@
                                           payload_specific, in_order);
 }
 
-bool Channel::HandleRtxPacket(const uint8_t* packet,
-                              size_t packet_length,
-                              const RTPHeader& header) {
-  if (!rtp_payload_registry_->IsRtx(header))
-    return false;
-
-  // Remove the RTX header and parse the original RTP header.
-  if (packet_length < header.headerLength)
-    return false;
-  if (packet_length > kVoiceEngineMaxIpPacketSizeBytes)
-    return false;
-  if (restored_packet_in_use_) {
-    WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVoice, _channelId,
-                 "Multiple RTX headers detected, dropping packet");
-    return false;
-  }
-  if (!rtp_payload_registry_->RestoreOriginalPacket(
-          restored_packet_, packet, &packet_length, rtp_receiver_->SSRC(),
-          header)) {
-    WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVoice, _channelId,
-                 "Incoming RTX packet: invalid RTP header");
-    return false;
-  }
-  restored_packet_in_use_ = true;
-  bool ret = OnRecoveredPacket(restored_packet_, packet_length);
-  restored_packet_in_use_ = false;
-  return ret;
-}
-
 bool Channel::IsPacketInOrder(const RTPHeader& header) const {
   StreamStatistician* statistician =
       rtp_receive_statistics_->GetStatistician(header.ssrc);
@@ -1801,9 +1768,6 @@
 
 bool Channel::IsPacketRetransmitted(const RTPHeader& header,
                                     bool in_order) const {
-  // Retransmissions are handled separately if RTX is enabled.
-  if (rtp_payload_registry_->RtxEnabled())
-    return false;
   StreamStatistician* statistician =
       rtp_receive_statistics_->GetStatistician(header.ssrc);
   if (!statistician)
diff --git a/voice_engine/channel.h b/voice_engine/channel.h
index 9ec3eba..884845e 100644
--- a/voice_engine/channel.h
+++ b/voice_engine/channel.h
@@ -426,9 +426,6 @@
                      size_t packet_length,
                      const RTPHeader& header,
                      bool in_order);
-  bool HandleRtxPacket(const uint8_t* packet,
-                       size_t packet_length,
-                       const RTPHeader& header);
   bool IsPacketInOrder(const RTPHeader& header) const;
   bool IsPacketRetransmitted(const RTPHeader& header, bool in_order) const;
   int ResendPackets(const uint16_t* sequence_numbers, int length);
@@ -495,7 +492,6 @@
   uint32_t playout_timestamp_rtp_ GUARDED_BY(video_sync_lock_);
   uint32_t playout_delay_ms_ GUARDED_BY(video_sync_lock_);
   uint16_t send_sequence_number_;
-  uint8_t restored_packet_[kVoiceEngineMaxIpPacketSizeBytes];
 
   rtc::CriticalSection ts_stats_lock_;
 
@@ -529,8 +525,6 @@
   rtc::CriticalSection overhead_per_packet_lock_;
   // VoENetwork
   AudioFrame::SpeechType _outputSpeechType;
-  // DTX.
-  bool restored_packet_in_use_;
   // RtcpBandwidthObserver
   std::unique_ptr<VoERtcpObserver> rtcp_observer_;
   // An associated send channel.
diff --git a/voice_engine/voice_engine_defines.h b/voice_engine/voice_engine_defines.h
index 1c2d55d..3b86c6a 100644
--- a/voice_engine/voice_engine_defines.h
+++ b/voice_engine/voice_engine_defines.h
@@ -34,8 +34,6 @@
 // Max scale factor for output volume panning
 const float kMaxOutputVolumePanning = 1.0f;
 
-enum { kVoiceEngineMaxIpPacketSizeBytes = 1500 };  // assumes Ethernet
-
 // Audio processing
 const NoiseSuppression::Level kDefaultNsMode = NoiseSuppression::kModerate;
 const GainControl::Mode kDefaultAgcMode =