Multi frequency DTMF support - sender side

webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc

 /*
  *  Copyright (c) 2012 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 1281 matching lines @@
   EXPECT_FALSE(voice_activity);
 }
 
 // As RFC4733, named telephone events are carried as part of the audio stream
 // and must use the same sequence number and timestamp base as the regular
 // audio channel.
 // This test checks the marker bit for the first packet and the consequent
 // packets of the same telephone event. Since it is specifically for DTMF
 // events, ignoring audio packets and sending kEmptyFrame instead of those.
 TEST_F(RtpSenderAudioTest, CheckMarkerBitForTelephoneEvents) {
-  char payload_name[RTP_PAYLOAD_NAME_SIZE] = "telephone-event";
-  uint8_t payload_type = 126;
-  ASSERT_EQ(0,
-            rtp_sender_->RegisterPayload(payload_name, payload_type, 0, 0, 0));
+  const char* kDtmfPayloadName = "telephone-event";
+  const uint32_t kPayloadFrequency = 8000;
+  const uint8_t kPayloadType = 126;
+  ASSERT_EQ(0, rtp_sender_->RegisterPayload(kDtmfPayloadName, kPayloadType,
+                                            kPayloadFrequency, 0, 0));
   // For Telephone events, payload is not added to the registered payload list,
   // it will register only the payload used for audio stream.
   // Registering the payload again for audio stream with different payload name.
-  const char kPayloadName[] = "payload_name";
-  ASSERT_EQ(
-      0, rtp_sender_->RegisterPayload(kPayloadName, payload_type, 8000, 1, 0));
+  const char* kPayloadName = "payload_name";
+  ASSERT_EQ(0, rtp_sender_->RegisterPayload(kPayloadName, kPayloadType,
+                                            kPayloadFrequency, 1, 0));
   int64_t capture_time_ms = fake_clock_.TimeInMilliseconds();
   // DTMF event key=9, duration=500 and attenuationdB=10
   rtp_sender_->SendTelephoneEvent(9, 500, 10);
   // During start, it takes the starting timestamp as last sent timestamp.
   // The duration is calculated as the difference of current and last sent
   // timestamp. So for first call it will skip since the duration is zero.
-  ASSERT_TRUE(rtp_sender_->SendOutgoingData(kEmptyFrame, payload_type,
-                                                capture_time_ms, 0, nullptr, 0,
-                                                nullptr, nullptr, nullptr));
+  ASSERT_TRUE(rtp_sender_->SendOutgoingData(kEmptyFrame, kPayloadType,
+                                            capture_time_ms, 0, nullptr, 0,
+                                            nullptr, nullptr, nullptr));
   // DTMF Sample Length is (Frequency/1000) * Duration.
   // So in this case, it is (8000/1000) * 500 = 4000.
   // Sending it as two packets.
   ASSERT_TRUE(rtp_sender_->SendOutgoingData(
-                      kEmptyFrame, payload_type, capture_time_ms + 2000, 0,
+                      kEmptyFrame, kPayloadType, capture_time_ms + 2000, 0,
                       nullptr, 0, nullptr, nullptr, nullptr));
 
   // Marker Bit should be set to 1 for first packet.
   EXPECT_TRUE(transport_.last_sent_packet().Marker());
 
   ASSERT_TRUE(rtp_sender_->SendOutgoingData(
-                      kEmptyFrame, payload_type, capture_time_ms + 4000, 0,
+                      kEmptyFrame, kPayloadType, capture_time_ms + 4000, 0,
                       nullptr, 0, nullptr, nullptr, nullptr));
   // Marker Bit should be set to 0 for rest of the packets.
   EXPECT_FALSE(transport_.last_sent_packet().Marker());
 }
 
 TEST_F(RtpSenderTestWithoutPacer, BytesReportedCorrectly) {
   const char* kPayloadName = "GENERIC";
   const uint8_t kPayloadType = 127;
   rtp_sender_->SetSSRC(1234);
   rtp_sender_->SetRtxSsrc(4321);
@@ skipping 194 matching lines @@
   EXPECT_CALL(mock_overhead_observer, OnOverheadChanged(_)).Times(1);
   rtp_sender_->SetTransportOverhead(28);
   rtp_sender_->SetTransportOverhead(28);
 
   EXPECT_CALL(mock_overhead_observer, OnOverheadChanged(_)).Times(1);
   SendGenericPayload();
   SendGenericPayload();
 }
 
 }  // namespace webrtc