[NOT FOR REVIEW] Convert channel counts to size_t.

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 452 matching lines @@
 
   // Insert one packet.
   EXPECT_EQ(NetEq::kOK,
             neteq_->InsertPacket(
                 rtp_header, payload, kPayloadLengthBytes, kReceiveTime));
 
   // Pull audio once.
   const size_t kMaxOutputSize = static_cast<size_t>(10 * kSampleRateHz / 1000);
   int16_t output[kMaxOutputSize];
   size_t samples_per_channel;
-  int num_channels;
+  size_t num_channels;
   NetEqOutputType type;
   EXPECT_EQ(
       NetEq::kOK,
       neteq_->GetAudio(
           kMaxOutputSize, output, &samples_per_channel, &num_channels, &type));
   ASSERT_EQ(kMaxOutputSize, samples_per_channel);
-  EXPECT_EQ(1, num_channels);
+  EXPECT_EQ(1u, num_channels);
   EXPECT_EQ(kOutputNormal, type);
 
   // Start with a simple check that the fake decoder is behaving as expected.
   EXPECT_EQ(kPayloadLengthSamples,
             static_cast<size_t>(decoder_.next_value() - 1));
 
   // The value of the last of the output samples is the same as the number of
   // samples played from the decoded packet. Thus, this number + the RTP
   // timestamp should match the playout timestamp.
   uint32_t timestamp = 0;
@@ skipping 52 matching lines @@
 
   // Insert one packet.
   EXPECT_EQ(NetEq::kOK,
             neteq_->InsertPacket(
                 rtp_header, payload, kPayloadLengthBytes, kReceiveTime));
 
   // Pull audio once.
   const size_t kMaxOutputSize = static_cast<size_t>(10 * kSampleRateHz / 1000);
   int16_t output[kMaxOutputSize];
   size_t samples_per_channel;
-  int num_channels;
+  size_t num_channels;
   NetEqOutputType type;
   EXPECT_EQ(
       NetEq::kOK,
       neteq_->GetAudio(
           kMaxOutputSize, output, &samples_per_channel, &num_channels, &type));
   ASSERT_EQ(kMaxOutputSize, samples_per_channel);
-  EXPECT_EQ(1, num_channels);
+  EXPECT_EQ(1u, num_channels);
   EXPECT_EQ(kOutputNormal, type);
 
   // Insert two more packets. The first one is out of order, and is already too
   // old, the second one is the expected next packet.
   rtp_header.header.sequenceNumber -= 1;
   rtp_header.header.timestamp -= kPayloadLengthSamples;
   payload[0] = 1;
   EXPECT_EQ(NetEq::kOK,
             neteq_->InsertPacket(
                 rtp_header, payload, kPayloadLengthBytes, kReceiveTime));
@@ skipping 12 matching lines @@
                                           dummy_output + kPayloadLengthSamples),
                       SetArgPointee<5>(AudioDecoder::kSpeech),
                       Return(kPayloadLengthSamples)));
 
   // Pull audio once.
   EXPECT_EQ(
       NetEq::kOK,
       neteq_->GetAudio(
           kMaxOutputSize, output, &samples_per_channel, &num_channels, &type));
   ASSERT_EQ(kMaxOutputSize, samples_per_channel);
-  EXPECT_EQ(1, num_channels);
+  EXPECT_EQ(1u, num_channels);
   EXPECT_EQ(kOutputNormal, type);
 
   // Now check the packet buffer, and make sure it is empty, since the
   // out-of-order packet should have been discarded.
   EXPECT_TRUE(packet_buffer_->Empty());
 
   EXPECT_CALL(mock_decoder, Die());
 }
 
 // This test verifies that NetEq can handle the situation where the first
@@ skipping 19 matching lines @@
   // this packet will be rejected.
   EXPECT_EQ(NetEq::kFail,
             neteq_->InsertPacket(rtp_header, payload, kPayloadLengthBytes,
                                  kReceiveTime));
   EXPECT_EQ(NetEq::kUnknownRtpPayloadType, neteq_->LastError());
 
   // Pull audio once.
   const size_t kMaxOutputSize = static_cast<size_t>(10 * kSampleRateHz / 1000);
   int16_t output[kMaxOutputSize];
   size_t samples_per_channel;
-  int num_channels;
+  size_t num_channels;
   NetEqOutputType type;
   EXPECT_EQ(NetEq::kOK,
             neteq_->GetAudio(kMaxOutputSize, output, &samples_per_channel,
                              &num_channels, &type));
   ASSERT_LE(samples_per_channel, kMaxOutputSize);
   EXPECT_EQ(kMaxOutputSize, samples_per_channel);
-  EXPECT_EQ(1, num_channels);
+  EXPECT_EQ(1u, num_channels);
   EXPECT_EQ(kOutputPLC, type);
 
   // Register the payload type.
   EXPECT_EQ(NetEq::kOK,
             neteq_->RegisterPayloadType(kDecoderPCM16B, kPayloadType));
 
   // Insert 10 packets.
   for (size_t i = 0; i < 10; ++i) {
     rtp_header.header.sequenceNumber++;
     rtp_header.header.timestamp += kPayloadLengthSamples;
     EXPECT_EQ(NetEq::kOK,
               neteq_->InsertPacket(rtp_header, payload, kPayloadLengthBytes,
                                    kReceiveTime));
     EXPECT_EQ(i + 1, packet_buffer_->NumPacketsInBuffer());
   }
 
   // Pull audio repeatedly and make sure we get normal output, that is not PLC.
   for (size_t i = 0; i < 3; ++i) {
     EXPECT_EQ(NetEq::kOK,
               neteq_->GetAudio(kMaxOutputSize, output, &samples_per_channel,
                                &num_channels, &type));
     ASSERT_LE(samples_per_channel, kMaxOutputSize);
     EXPECT_EQ(kMaxOutputSize, samples_per_channel);
-    EXPECT_EQ(1, num_channels);
+    EXPECT_EQ(1u, num_channels);
     EXPECT_EQ(kOutputNormal, type)
         << "NetEq did not decode the packets as expected.";
   }
 }
 
 // This test verifies that NetEq can handle comfort noise and enters/quits codec
 // internal CNG mode properly.
 TEST_F(NetEqImplTest, CodecInternalCng) {
   UseNoMocks();
   CreateInstance();
@@ skipping 62 matching lines @@
   payload[0] = 1;
   rtp_header.header.sequenceNumber++;
   rtp_header.header.timestamp += kPayloadLengthSamples;
   EXPECT_EQ(NetEq::kOK,
             neteq_->InsertPacket(
                 rtp_header, payload, kPayloadLengthBytes, kReceiveTime));
 
   const size_t kMaxOutputSize = static_cast<size_t>(10 * kSampleRateKhz);
   int16_t output[kMaxOutputSize];
   size_t samples_per_channel;
-  int num_channels;
+  size_t num_channels;
   uint32_t timestamp;
   uint32_t last_timestamp;
   NetEqOutputType type;
   NetEqOutputType expected_type[8] = {
       kOutputNormal, kOutputNormal,
       kOutputCNG, kOutputCNG,
       kOutputCNG, kOutputCNG,
       kOutputNormal, kOutputNormal
   };
   int expected_timestamp_increment[8] = {
       -1,  // will not be used.
       10 * kSampleRateKhz,
       0, 0,  // timestamp does not increase during CNG mode.
       0, 0,
       50 * kSampleRateKhz, 10 * kSampleRateKhz
   };
 
   EXPECT_EQ(NetEq::kOK,
             neteq_->GetAudio(kMaxOutputSize, output, &samples_per_channel,
                              &num_channels, &type));
   EXPECT_TRUE(neteq_->GetPlayoutTimestamp(&last_timestamp));
 
   for (size_t i = 1; i < 6; ++i) {
     ASSERT_EQ(kMaxOutputSize, samples_per_channel);
-    EXPECT_EQ(1, num_channels);
+    EXPECT_EQ(1u, num_channels);
     EXPECT_EQ(expected_type[i - 1], type);
     EXPECT_TRUE(neteq_->GetPlayoutTimestamp(&timestamp));
     EXPECT_EQ(NetEq::kOK,
               neteq_->GetAudio(kMaxOutputSize, output, &samples_per_channel,
                                &num_channels, &type));
     EXPECT_TRUE(neteq_->GetPlayoutTimestamp(&timestamp));
     EXPECT_EQ(timestamp, last_timestamp + expected_timestamp_increment[i]);
     last_timestamp = timestamp;
   }
 
   // Insert third packet, which leaves a gap from last packet.
   payload[0] = 2;
   rtp_header.header.sequenceNumber += 2;
   rtp_header.header.timestamp += 2 * kPayloadLengthSamples;
   EXPECT_EQ(NetEq::kOK,
             neteq_->InsertPacket(
                 rtp_header, payload, kPayloadLengthBytes, kReceiveTime));
 
   for (size_t i = 6; i < 8; ++i) {
     ASSERT_EQ(kMaxOutputSize, samples_per_channel);
-    EXPECT_EQ(1, num_channels);
+    EXPECT_EQ(1u, num_channels);
     EXPECT_EQ(expected_type[i - 1], type);
     EXPECT_EQ(NetEq::kOK,
               neteq_->GetAudio(kMaxOutputSize, output, &samples_per_channel,
                                &num_channels, &type));
     EXPECT_TRUE(neteq_->GetPlayoutTimestamp(&timestamp));
     EXPECT_EQ(timestamp, last_timestamp + expected_timestamp_increment[i]);
     last_timestamp = timestamp;
   }
 
   // Now check the packet buffer, and make sure it is empty.
   EXPECT_TRUE(packet_buffer_->Empty());
 
   EXPECT_CALL(mock_decoder, Die());
 }
 
 TEST_F(NetEqImplTest, UnsupportedDecoder) {
   UseNoMocks();
   CreateInstance();
   static const size_t kNetEqMaxFrameSize = 2880;  // 60 ms @ 48 kHz.
-  static const int kChannels = 2;
+  static const size_t kChannels = 2;
 
   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 =
       static_cast<size_t>(10 * kSampleRateHz / 1000);  // 10 ms.
   const size_t kPayloadLengthBytes = 1;
   uint8_t payload[kPayloadLengthBytes]= {0};
   int16_t dummy_output[kPayloadLengthSamples * kChannels] = {0};
@@ skipping 55 matching lines @@
   // Insert another packet.
   payload[0] = kSecondPayloadValue;  // This will make Decode() successful.
   rtp_header.header.sequenceNumber++;
   // The second timestamp needs to be at least 30 ms after the first to make
   // the second packet get decoded.
   rtp_header.header.timestamp += 3 * kPayloadLengthSamples;
   EXPECT_EQ(NetEq::kOK,
             neteq_->InsertPacket(
                 rtp_header, payload, kPayloadLengthBytes, kReceiveTime));
 
-  const size_t kMaxOutputSize =
-      static_cast<size_t>(10 * kSampleRateHz / 1000 * kChannels);
+  const size_t kMaxOutputSize = 10 * kSampleRateHz / 1000 * kChannels;
   int16_t output[kMaxOutputSize];
   size_t samples_per_channel;
-  int num_channels;
+  size_t num_channels;
   NetEqOutputType type;
 
   EXPECT_EQ(NetEq::kFail, neteq_->GetAudio(kMaxOutputSize, output,
                                            &samples_per_channel, &num_channels,
                                            &type));
   EXPECT_EQ(NetEq::kOtherDecoderError, neteq_->LastError());
   EXPECT_EQ(kMaxOutputSize, samples_per_channel * kChannels);
   EXPECT_EQ(kChannels, num_channels);
 
   EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(kMaxOutputSize, output,
                                          &samples_per_channel, &num_channels,
                                          &type));
   EXPECT_EQ(kMaxOutputSize, samples_per_channel * kChannels);
   EXPECT_EQ(kChannels, num_channels);
 }
 
 }  // namespace webrtc