Fix bug in DTMF generation where events with level > 36 would be ignored.

webrtc/modules/audio_coding/neteq/neteq_impl_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 363 matching lines @@
   // Insert one more packet and make sure the buffer got flushed. That is, it
   // should only hold one single packet.
   EXPECT_EQ(NetEq::kOK,
             neteq_->InsertPacket(rtp_header, payload, kReceiveTime));
   EXPECT_EQ(1u, packet_buffer_->NumPacketsInBuffer());
   const RTPHeader* test_header = packet_buffer_->NextRtpHeader();
   EXPECT_EQ(rtp_header.header.timestamp, test_header->timestamp);
   EXPECT_EQ(rtp_header.header.sequenceNumber, test_header->sequenceNumber);
 }
 
+TEST_F(NetEqImplTest, TestDtmfPacket) {
+  UseNoMocks();
+  CreateInstance();
+  const size_t kPayloadLength = 4;
+  const uint8_t kPayloadType = 110;
+  const uint32_t kReceiveTime = 17;
+  const int kSampleRateHz = 8000;
+  // Event: 2, E bit, Volume: 63, Length: 4176.
+  uint8_t payload[kPayloadLength] = { 0x02, 0x80 + 0x3F, 0x10, 0xF0 };
+  WebRtcRTPHeader rtp_header;
+  rtp_header.header.payloadType = kPayloadType;
+  rtp_header.header.sequenceNumber = 0x1234;
+  rtp_header.header.timestamp = 0x12345678;
+  rtp_header.header.ssrc = 0x87654321;
+
+  EXPECT_EQ(NetEq::kOK, neteq_->RegisterPayloadType(
+      NetEqDecoder::kDecoderAVT, "telephone-event", kPayloadType));
+
+  // Insert one packet.
+  EXPECT_EQ(NetEq::kOK,
+            neteq_->InsertPacket(rtp_header, payload, kReceiveTime));
+
+  // Pull audio once.
+  const size_t kMaxOutputSize = static_cast<size_t>(10 * kSampleRateHz / 1000);
+  AudioFrame output;
+  bool muted;
+  EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output, &muted));
+  ASSERT_FALSE(muted);
+  ASSERT_EQ(kMaxOutputSize, output.samples_per_channel_);
+  EXPECT_EQ(1u, output.num_channels_);
+  EXPECT_EQ(AudioFrame::kNormalSpeech, output.speech_type_);
+}
+
 // This test verifies that timestamps propagate from the incoming packets
 // through to the sync buffer and to the playout timestamp.
 TEST_F(NetEqImplTest, VerifyTimestampPropagation) {
   UseNoMocks();
   CreateInstance();
 
   const uint8_t kPayloadType = 17;   // Just an arbitrary number.
   const uint32_t kReceiveTime = 17;  // Value doesn't matter for this test.
   const int kSampleRateHz = 8000;
   const size_t kPayloadLengthSamples =
@@ skipping 1031 matching lines @@
   EXPECT_CALL(*mock_delay_manager_, BufferLimits(_, _))
       .Times(1)
       .WillOnce(DoAll(SetArgPointee<0>(1), SetArgPointee<1>(2)));
 
   bool muted;
   EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output_, &muted));
   EXPECT_EQ(kAccelerate, neteq_->last_operation_for_test());
 }
 
 }// namespace webrtc