Multi frequency DTMF support - sender side

webrtc/media/engine/webrtcvoiceengine.cc

 /*
  *  Copyright (c) 2004 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 1265 matching lines @@
     max_send_bitrate_bps_ = bps;
 
     if (config_.send_codec_spec.codec_inst.rate != *send_rate) {
       // Recreate AudioSendStream with new bit rate.
       config_.send_codec_spec.codec_inst.rate = *send_rate;
       RecreateAudioSendStream();
     }
     return true;
   }
 
-  bool SendTelephoneEvent(int payload_type, int event, int duration_ms) {
+  bool SendTelephoneEvent(int payload_type, int payload_freq, int event,
+                          int duration_ms) {
     RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
     RTC_DCHECK(stream_);
-    return stream_->SendTelephoneEvent(payload_type, event, duration_ms);
+    return stream_->SendTelephoneEvent(payload_type, payload_freq, event,
+                                       duration_ms);
   }
 
   void SetSend(bool send) {
     RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
     send_ = send;
     UpdateSendState();
   }
 
   void SetMuted(bool muted) {
     RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
@@ skipping 569 matching lines @@
   }
   return result;
 }
 
 // Utility function called from SetSendParameters() to extract current send
 // codec settings from the given list of codecs (originally from SDP). Both send
 // and receive streams may be reconfigured based on the new settings.
 bool WebRtcVoiceMediaChannel::SetSendCodecs(
     const std::vector<AudioCodec>& codecs) {
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
-  // TODO(solenberg): Validate input - that payload types don't overlap, are
-  //                  within range, filter out codecs we don't support,
-  //                  redundant codecs etc - the same way it is done for
-  //                  RtpHeaderExtensions.
+  dtmf_payload_type_ = rtc::Optional<int>();
+  dtmf_payload_freq_ = -1;
 
-  // Find the DTMF telephone event "codec" payload type.
-  dtmf_payload_type_ = rtc::Optional<int>();
+  // Validate supplied codecs list.
   for (const AudioCodec& codec : codecs) {
-    if (IsCodec(codec, kDtmfCodecName)) {
-      if (codec.id < kMinPayloadType || codec.id > kMaxPayloadType) {
-        return false;
-      }
-      dtmf_payload_type_ = rtc::Optional<int>(codec.id);
-      break;
+    // TODO(solenberg): Validate more aspects of input - that payload types
+    //                  don't overlap, remove redundant/unsupported codecs etc -
+    //                  the same way it is done for RtpHeaderExtensions.
+    if (codec.id < kMinPayloadType || codec.id > kMaxPayloadType) {
+      LOG(LS_WARNING) << "Codec payload type out of range: " << ToString(codec);
+      return false;
     }
   }
 
+  // Find PT of telephone-event codec with lowest clockrate, as a fallback, in
+  // case we don't have a DTMF codec with a rate matching the send codec's, or
+  // if this function returns early.
+  std::vector<AudioCodec> dtmf_codecs;
+  for (const AudioCodec& codec : codecs) {
+    if (IsCodec(codec, kDtmfCodecName)) {
+      dtmf_codecs.push_back(codec);
+      if (!dtmf_payload_type_ || codec.clockrate < dtmf_payload_freq_) {
+        dtmf_payload_type_ = rtc::Optional<int>(codec.id);
+        dtmf_payload_freq_ = codec.clockrate;
+      }
+    }
+  }
+
   // Scan through the list to figure out the codec to use for sending, along
   // with the proper configuration for VAD, CNG, NACK and Opus-specific
   // parameters.
   // TODO(solenberg): Refactor this logic once we create AudioEncoders here.
   webrtc::AudioSendStream::Config::SendCodecSpec send_codec_spec;
   {
     send_codec_spec.nack_enabled = send_codec_spec_.nack_enabled;
 
     // Find send codec (the first non-telephone-event/CN codec).
     const AudioCodec* codec = WebRtcVoiceCodecs::GetPreferredCodec(
@@ skipping 53 matching lines @@
           default:
             LOG(LS_WARNING) << "CN frequency " << codec.clockrate
                             << " not supported.";
             continue;
         }
         send_codec_spec.cng_payload_type = codec.id;
         send_codec_spec.cng_plfreq = cng_plfreq;
         break;
       }
     }
+
+    // Find the telephone-event PT exactly matching the preferred send codec.
+    for (const AudioCodec& dtmf_codec : dtmf_codecs) {
+      if (dtmf_codec.clockrate == codec->clockrate) {
+        dtmf_payload_type_ = rtc::Optional<int>(dtmf_codec.id);
+        dtmf_payload_freq_ = dtmf_codec.clockrate;
+        break;
+      }
+    }
   }
 
   // Apply new settings to all streams.
   if (send_codec_spec_ != send_codec_spec) {
     send_codec_spec_ = std::move(send_codec_spec);
     for (const auto& kv : send_streams_) {
       kv.second->RecreateAudioSendStream(send_codec_spec_);
     }
   }
 
@@ skipping 387 matching lines @@
   if (event < kMinTelephoneEventCode ||
       event > kMaxTelephoneEventCode) {
     LOG(LS_WARNING) << "DTMF event code " << event << " out of range.";
     return false;
   }
   if (duration < kMinTelephoneEventDuration ||
       duration > kMaxTelephoneEventDuration) {
     LOG(LS_WARNING) << "DTMF event duration " << duration << " out of range.";
     return false;
   }
-  return it->second->SendTelephoneEvent(*dtmf_payload_type_, event, duration);
+  RTC_DCHECK_NE(-1, dtmf_payload_freq_);
+  return it->second->SendTelephoneEvent(*dtmf_payload_type_, dtmf_payload_freq_,
+                                        event, duration);
 }
 
 void WebRtcVoiceMediaChannel::OnPacketReceived(
     rtc::CopyOnWriteBuffer* packet, const rtc::PacketTime& packet_time) {
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
 
   const webrtc::PacketTime webrtc_packet_time(packet_time.timestamp,
                                               packet_time.not_before);
   webrtc::PacketReceiver::DeliveryStatus delivery_result =
       call_->Receiver()->DeliverPacket(webrtc::MediaType::AUDIO,
@@ skipping 216 matching lines @@
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   const auto it = send_streams_.find(ssrc);
   if (it != send_streams_.end()) {
     return it->second->channel();
   }
   return -1;
 }
 }  // namespace cricket
 
 #endif  // HAVE_WEBRTC_VOICE