Revert of - Clean up unused voice engine DTMF code.

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 695aa1dee2434f85d6a1730771056b6ed9067fb6..11af45edc837fb8f94eb9da43edf9ba2be86dde9 100644
--- a/webrtc/voice_engine/channel.cc
+++ b/webrtc/voice_engine/channel.cc
@@ -40,10 +40,12 @@
 #include "webrtc/voice_engine/transmit_mixer.h"
 #include "webrtc/voice_engine/utility.h"
 
+#if defined(_WIN32)
+#include <Qos.h>
+#endif
+
 namespace webrtc {
 namespace voe {
-
-const int kTelephoneEventAttenuationdB = 10;
 
 class TransportFeedbackProxy : public TransportFeedbackObserver {
  public:
@@ -361,7 +363,7 @@
                " volume=%u)",
                event, lengthMs, volume);
 
-  if (event > 15) {
+  if (!_playOutbandDtmfEvent || (event > 15)) {
     // Ignore callback since feedback is disabled or event is not a
     // Dtmf tone event.
     return;
@@ -759,11 +761,14 @@
       _outputFilePlayerId(VoEModuleId(instanceId, channelId) + 1025),
       _outputFileRecorderId(VoEModuleId(instanceId, channelId) + 1026),
       _outputFileRecording(false),
+      _inbandDtmfQueue(VoEModuleId(instanceId, channelId)),
+      _inbandDtmfGenerator(VoEModuleId(instanceId, channelId)),
       _outputExternalMedia(false),
       _inputExternalMediaCallbackPtr(NULL),
       _outputExternalMediaCallbackPtr(NULL),
       _timeStamp(0),  // This is just an offset, RTP module will add it's own
                       // random offset
+      _sendTelephoneEventPayloadType(106),
       ntp_estimator_(Clock::GetRealTimeClock()),
       jitter_buffer_playout_timestamp_(0),
       playout_timestamp_rtp_(0),
@@ -791,6 +796,8 @@
       _panLeft(1.0f),
       _panRight(1.0f),
       _outputGain(1.0f),
+      _playOutbandDtmfEvent(false),
+      _playInbandDtmfEvent(false),
       _lastLocalTimeStamp(0),
       _lastPayloadType(0),
       _includeAudioLevelIndication(false),
@@ -823,6 +830,8 @@
       config.Get<NetEqFastAccelerate>().enabled;
   audio_coding_.reset(AudioCodingModule::Create(acm_config));
 
+  _inbandDtmfQueue.ResetDtmf();
+  _inbandDtmfGenerator.Init();
   _outputAudioLevel.Clear();
 
   RtpRtcp::Configuration configuration;
@@ -2203,18 +2212,21 @@
   return 0;
 }
 
-int Channel::SendTelephoneEventOutband(int event, int duration_ms) {
-  WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
-               "Channel::SendTelephoneEventOutband(...)");
-  RTC_DCHECK_LE(0, event);
-  RTC_DCHECK_GE(255, event);
-  RTC_DCHECK_LE(0, duration_ms);
-  RTC_DCHECK_GE(65535, duration_ms);
+int Channel::SendTelephoneEventOutband(unsigned char eventCode,
+                                       int lengthMs,
+                                       int attenuationDb,
+                                       bool playDtmfEvent) {
+  WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
+               "Channel::SendTelephoneEventOutband(..., playDtmfEvent=%d)",
+               playDtmfEvent);
   if (!Sending()) {
     return -1;
   }
-  if (_rtpRtcpModule->SendTelephoneEventOutband(
-      event, duration_ms, kTelephoneEventAttenuationdB) != 0) {
+
+  _playOutbandDtmfEvent = playDtmfEvent;
+
+  if (_rtpRtcpModule->SendTelephoneEventOutband(eventCode, lengthMs,
+                                                attenuationDb) != 0) {
     _engineStatisticsPtr->SetLastError(
         VE_SEND_DTMF_FAILED, kTraceWarning,
         "SendTelephoneEventOutband() failed to send event");
@@ -2223,14 +2235,32 @@
   return 0;
 }
 
-int Channel::SetSendTelephoneEventPayloadType(int payload_type) {
+int Channel::SendTelephoneEventInband(unsigned char eventCode,
+                                      int lengthMs,
+                                      int attenuationDb,
+                                      bool playDtmfEvent) {
+  WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
+               "Channel::SendTelephoneEventInband(..., playDtmfEvent=%d)",
+               playDtmfEvent);
+
+  _playInbandDtmfEvent = playDtmfEvent;
+  _inbandDtmfQueue.AddDtmf(eventCode, lengthMs, attenuationDb);
+
+  return 0;
+}
+
+int Channel::SetSendTelephoneEventPayloadType(unsigned char type) {
   WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
                "Channel::SetSendTelephoneEventPayloadType()");
-  RTC_DCHECK_LE(0, payload_type);
-  RTC_DCHECK_GE(127, payload_type);
-  CodecInst codec = {0};
+  if (type > 127) {
+    _engineStatisticsPtr->SetLastError(
+        VE_INVALID_ARGUMENT, kTraceError,
+        "SetSendTelephoneEventPayloadType() invalid type");
+    return -1;
+  }
+  CodecInst codec = {};
   codec.plfreq = 8000;
-  codec.pltype = payload_type;
+  codec.pltype = type;
   memcpy(codec.plname, "telephone-event", 16);
   if (_rtpRtcpModule->RegisterSendPayload(codec) != 0) {
     _rtpRtcpModule->DeRegisterSendPayload(codec.pltype);
@@ -2242,6 +2272,12 @@
       return -1;
     }
   }
+  _sendTelephoneEventPayloadType = type;
+  return 0;
+}
+
+int Channel::GetSendTelephoneEventPayloadType(unsigned char& type) {
+  type = _sendTelephoneEventPayloadType;
   return 0;
 }
 
@@ -2991,6 +3027,8 @@
     }
   }
 
+  InsertInbandDtmfTone();
+
   if (_includeAudioLevelIndication) {
     size_t length =
         _audioFrame.samples_per_channel_ * _audioFrame.num_channels_;
@@ -3307,6 +3345,64 @@
     return -1;
   }
 
+  return 0;
+}
+
+int Channel::InsertInbandDtmfTone() {
+  // Check if we should start a new tone.
+  if (_inbandDtmfQueue.PendingDtmf() && !_inbandDtmfGenerator.IsAddingTone() &&
+      _inbandDtmfGenerator.DelaySinceLastTone() >
+          kMinTelephoneEventSeparationMs) {
+    int8_t eventCode(0);
+    uint16_t lengthMs(0);
+    uint8_t attenuationDb(0);
+
+    eventCode = _inbandDtmfQueue.NextDtmf(&lengthMs, &attenuationDb);
+    _inbandDtmfGenerator.AddTone(eventCode, lengthMs, attenuationDb);
+    if (_playInbandDtmfEvent) {
+      // Add tone to output mixer using a reduced length to minimize
+      // risk of echo.
+      _outputMixerPtr->PlayDtmfTone(eventCode, lengthMs - 80, attenuationDb);
+    }
+  }
+
+  if (_inbandDtmfGenerator.IsAddingTone()) {
+    uint16_t frequency(0);
+    _inbandDtmfGenerator.GetSampleRate(frequency);
+
+    if (frequency != _audioFrame.sample_rate_hz_) {
+      // Update sample rate of Dtmf tone since the mixing frequency
+      // has changed.
+      _inbandDtmfGenerator.SetSampleRate(
+          (uint16_t)(_audioFrame.sample_rate_hz_));
+      // Reset the tone to be added taking the new sample rate into
+      // account.
+      _inbandDtmfGenerator.ResetTone();
+    }
+
+    int16_t toneBuffer[320];
+    uint16_t toneSamples(0);
+    // Get 10ms tone segment and set time since last tone to zero
+    if (_inbandDtmfGenerator.Get10msTone(toneBuffer, toneSamples) == -1) {
+      WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, _channelId),
+                   "Channel::EncodeAndSend() inserting Dtmf failed");
+      return -1;
+    }
+
+    // Replace mixed audio with DTMF tone.
+    for (size_t sample = 0; sample < _audioFrame.samples_per_channel_;
+         sample++) {
+      for (size_t channel = 0; channel < _audioFrame.num_channels_; channel++) {
+        const size_t index = sample * _audioFrame.num_channels_ + channel;
+        _audioFrame.data_[index] = toneBuffer[sample];
+      }
+    }
+
+    assert(_audioFrame.samples_per_channel_ == toneSamples);
+  } else {
+    // Add 10ms to "delay-since-last-tone" counter
+    _inbandDtmfGenerator.UpdateDelaySinceLastTone();
+  }
   return 0;
 }