Revert of Remove unnecessary interface TelephoneEventHandler (patchset #3 id:40001 of https://codereview.webrtc.org/2357583002/ )

Reason for revert:
breaks downstream code

Original issue's description:
> Remove unnecessary interface TelephoneEventHandler.
>
> BUG=webrtc:2795
>
> Committed: https://crrev.com/2beb42983ca24e1326a9a7f2c06b3ad740eea2c3
> Cr-Commit-Position: refs/heads/master@{#14346}

TBR=henrik.lundin@webrtc.org,solenberg@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:2795

Review-Url: https://codereview.webrtc.org/2362673002
Cr-Commit-Position: refs/heads/master@{#14348}

webrtc/modules/rtp_rtcp/include/rtp_receiver.h

@@ -18,6 +18,21 @@ namespace webrtc {
 
 class RTPPayloadRegistry;
 
+class TelephoneEventHandler {
+ public:
+  virtual ~TelephoneEventHandler() {}
+
+  // The following three methods implement the TelephoneEventHandler interface.
+  // Forward DTMFs to decoder for playout.
+  virtual void SetTelephoneEventForwardToDecoder(bool forward_to_decoder) = 0;
+
+  // Is forwarding of outband telephone events turned on/off?
+  virtual bool TelephoneEventForwardToDecoder() const = 0;
+
+  // Is TelephoneEvent configured with payload type payload_type
+  virtual bool TelephoneEventPayloadType(const int8_t payload_type) const = 0;
+};
+
 class RtpReceiver {
  public:
   // Creates a video-enabled RTP receiver.
@@ -36,6 +51,9 @@ class RtpReceiver {
 
   virtual ~RtpReceiver() {}
 
+  // Returns a TelephoneEventHandler if available.
+  virtual TelephoneEventHandler* GetTelephoneEventHandler() = 0;
+
   // Registers a receive payload in the payload registry and notifies the media
   // receiver strategy.
   virtual int32_t RegisterReceivePayload(

webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.cc

@@ -25,7 +25,9 @@ RTPReceiverStrategy* RTPReceiverStrategy::CreateAudioStrategy(
 
 RTPReceiverAudio::RTPReceiverAudio(RtpData* data_callback)
     : RTPReceiverStrategy(data_callback),
+      TelephoneEventHandler(),
       last_received_frequency_(8000),
+      telephone_event_forward_to_decoder_(false),
       telephone_event_payload_type_(-1),
       cng_nb_payload_type_(-1),
       cng_wb_payload_type_(-1),
@@ -40,6 +42,19 @@ RTPReceiverAudio::RTPReceiverAudio(RtpData* data_callback)
   memset(current_remote_energy_, 0, sizeof(current_remote_energy_));
 }
 
+// Outband TelephoneEvent(DTMF) detection
+void RTPReceiverAudio::SetTelephoneEventForwardToDecoder(
+    bool forward_to_decoder) {
+  rtc::CritScope lock(&crit_sect_);
+  telephone_event_forward_to_decoder_ = forward_to_decoder;
+}
+
+// Is forwarding of outband telephone events turned on/off?
+bool RTPReceiverAudio::TelephoneEventForwardToDecoder() const {
+  rtc::CritScope lock(&crit_sect_);
+  return telephone_event_forward_to_decoder_;
+}
+
 bool RTPReceiverAudio::TelephoneEventPayloadType(
     int8_t payload_type) const {
   rtc::CritScope lock(&crit_sect_);
@@ -341,6 +356,10 @@ int32_t RTPReceiverAudio::ParseAudioCodecSpecific(
 
     // check if it's a DTMF event, hence something we can playout
     if (telephone_event_packet) {
+      if (!telephone_event_forward_to_decoder_) {
+        // don't forward event to decoder
+        return 0;
+      }
       std::set<uint8_t>::iterator first =
           telephone_event_reported_.begin();
       if (first != telephone_event_reported_.end() && *first > 15) {

webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.h

@@ -23,13 +23,23 @@
 namespace webrtc {
 
 // Handles audio RTP packets. This class is thread-safe.
-class RTPReceiverAudio : public RTPReceiverStrategy {
+class RTPReceiverAudio : public RTPReceiverStrategy,
+                         public TelephoneEventHandler {
  public:
   explicit RTPReceiverAudio(RtpData* data_callback);
   virtual ~RTPReceiverAudio() {}
 
+  // The following three methods implement the TelephoneEventHandler interface.
+  // Forward DTMFs to decoder for playout.
+  void SetTelephoneEventForwardToDecoder(bool forward_to_decoder) override;
+
+  // Is forwarding of outband telephone events turned on/off?
+  bool TelephoneEventForwardToDecoder() const override;
+
   // Is TelephoneEvent configured with payload type payload_type
-  bool TelephoneEventPayloadType(const int8_t payload_type) const;
+  bool TelephoneEventPayloadType(const int8_t payload_type) const override;
+
+  TelephoneEventHandler* GetTelephoneEventHandler() override { return this; }
 
   // Returns true if CNG is configured with payload type payload_type. If so,
   // the frequency and cng_payload_type_has_changed are filled in.

webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.cc

@@ -200,6 +200,10 @@ bool RtpReceiverImpl::IncomingRtpPacket(
   return true;
 }
 
+TelephoneEventHandler* RtpReceiverImpl::GetTelephoneEventHandler() {
+  return rtp_media_receiver_->GetTelephoneEventHandler();
+}
+
 bool RtpReceiverImpl::Timestamp(uint32_t* timestamp) const {
   rtc::CritScope lock(&critical_section_rtp_receiver_);
   if (!HaveReceivedFrame())

webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.h

@@ -57,6 +57,8 @@ class RtpReceiverImpl : public RtpReceiver {
 
   int32_t Energy(uint8_t array_of_energy[kRtpCsrcSize]) const override;
 
+  TelephoneEventHandler* GetTelephoneEventHandler() override;
+
  private:
   bool HaveReceivedFrame() const;
 

webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h

@@ -44,6 +44,8 @@ class RTPReceiverStrategy {
                                  int64_t timestamp_ms,
                                  bool is_first_packet) = 0;
 
+  virtual TelephoneEventHandler* GetTelephoneEventHandler() = 0;
+
   // Retrieves the last known applicable frequency.
   virtual int GetPayloadTypeFrequency() const = 0;
 

webrtc/modules/rtp_rtcp/source/rtp_receiver_video.h

@@ -33,6 +33,8 @@ class RTPReceiverVideo : public RTPReceiverStrategy {
                          int64_t timestamp,
                          bool is_first_packet) override;
 
+  TelephoneEventHandler* GetTelephoneEventHandler() override { return NULL; }
+
   int GetPayloadTypeFrequency() const override;
 
   RTPAliveType ProcessDeadOrAlive(uint16_t last_payload_length) const override;

webrtc/modules/rtp_rtcp/test/testAPI/test_api_audio.cc

@@ -162,6 +162,9 @@ TEST_F(RtpRtcpAudioTest, Basic) {
   module1->SetSSRC(test_ssrc);
   module1->SetStartTimestamp(test_timestamp);
 
+  // Test detection at the end of a DTMF tone.
+  // EXPECT_EQ(0, module2->SetTelephoneEventForwardToDecoder(true));
+
   EXPECT_EQ(0, module1->SetSendingStatus(true));
 
   // Start basic RTP test.

webrtc/voice_engine/channel.cc

@@ -809,6 +809,7 @@ Channel::Channel(int32_t channelId,
                                            this,
                                            this,
                                            rtp_payload_registry_.get())),
+      telephone_event_handler_(rtp_receiver_->GetTelephoneEventHandler()),
       _outputAudioLevel(),
       _externalTransport(false),
       // Avoid conflict with other channels by adding 1024 - 1026,
@@ -978,6 +979,7 @@ int32_t Channel::Init() {
   // disabled by the user.
   // After StopListen (when no sockets exists), RTCP packets will no longer
   // be transmitted since the Transport object will then be invalid.
+  telephone_event_handler_->SetTelephoneEventForwardToDecoder(true);
   // RTCP is enabled by default.
   _rtpRtcpModule->SetRTCPStatus(RtcpMode::kCompound);
   // --- Register all permanent callbacks

webrtc/voice_engine/channel.h

@@ -458,6 +458,7 @@ class Channel
   std::unique_ptr<ReceiveStatistics> rtp_receive_statistics_;
   std::unique_ptr<StatisticsProxy> statistics_proxy_;
   std::unique_ptr<RtpReceiver> rtp_receiver_;
+  TelephoneEventHandler* telephone_event_handler_;
   std::unique_ptr<RtpRtcp> _rtpRtcpModule;
   std::unique_ptr<AudioCodingModule> audio_coding_;
   acm2::CodecManager codec_manager_;