Convert channel counts to size_t.

webrtc/modules/audio_coding/neteq/neteq_unittest.cc

 /*
  *  Copyright (c) 2011 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 407 matching lines @@
                            packet_->payload(), packet_->payload_length_bytes()),
                        static_cast<uint32_t>(packet_->time_ms() *
                                              (output_sample_rate_ / 1000))));
     }
     // Get next packet.
     packet_.reset(rtp_source_->NextPacket());
   }
 
   // Get audio from NetEq.
   NetEqOutputType type;
-  int num_channels;
+  size_t num_channels;
   ASSERT_EQ(0, neteq_->GetAudio(kMaxBlockSize, out_data_, out_len,
                                 &num_channels, &type));
   ASSERT_TRUE((*out_len == kBlockSize8kHz) ||
               (*out_len == kBlockSize16kHz) ||
               (*out_len == kBlockSize32kHz) ||
               (*out_len == kBlockSize48kHz));
   output_sample_rate_ = static_cast<int>(*out_len / 10 * 1000);
   EXPECT_EQ(output_sample_rate_, neteq_->last_output_sample_rate_hz());
 
   // Increase time.
@@ skipping 162 matching lines @@
     rtp_info.header.sequenceNumber = i;
     rtp_info.header.timestamp = i * kSamples;
     rtp_info.header.ssrc = 0x1234;  // Just an arbitrary SSRC.
     rtp_info.header.payloadType = 94;  // PCM16b WB codec.
     rtp_info.header.markerBit = 0;
     ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, 0));
   }
   // Pull out all data.
   for (size_t i = 0; i < num_frames; ++i) {
     size_t out_len;
-    int num_channels;
+    size_t num_channels;
     NetEqOutputType type;
     ASSERT_EQ(0, neteq_->GetAudio(kMaxBlockSize, out_data_, &out_len,
                                   &num_channels, &type));
     ASSERT_EQ(kBlockSize16kHz, out_len);
   }
 
   NetEqNetworkStatistics stats;
   EXPECT_EQ(0, neteq_->NetworkStatistics(&stats));
   // Since all frames are dumped into NetEQ at once, but pulled out with 10 ms
   // spacing (per definition), we expect the delay to increase with 10 ms for
@@ skipping 24 matching lines @@
     for (int n = 0; n < num_packets; ++n) {
       uint8_t payload[kPayloadBytes] = {0};
       WebRtcRTPHeader rtp_info;
       PopulateRtpInfo(frame_index, frame_index * kSamples, &rtp_info);
       ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, 0));
       ++frame_index;
     }
 
     // Pull out data once.
     size_t out_len;
-    int num_channels;
+    size_t num_channels;
     NetEqOutputType type;
     ASSERT_EQ(0, neteq_->GetAudio(kMaxBlockSize, out_data_, &out_len,
                                   &num_channels, &type));
     ASSERT_EQ(kBlockSize16kHz, out_len);
   }
 
   NetEqNetworkStatistics network_stats;
   ASSERT_EQ(0, neteq_->NetworkStatistics(&network_stats));
   EXPECT_EQ(-103196, network_stats.clockdrift_ppm);
 }
@@ skipping 10 matching lines @@
     for (int n = 0; n < num_packets; ++n) {
       uint8_t payload[kPayloadBytes] = {0};
       WebRtcRTPHeader rtp_info;
       PopulateRtpInfo(frame_index, frame_index * kSamples, &rtp_info);
       ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, 0));
       ++frame_index;
     }
 
     // Pull out data once.
     size_t out_len;
-    int num_channels;
+    size_t num_channels;
     NetEqOutputType type;
     ASSERT_EQ(0, neteq_->GetAudio(kMaxBlockSize, out_data_, &out_len,
                                   &num_channels, &type));
     ASSERT_EQ(kBlockSize16kHz, out_len);
   }
 
   NetEqNetworkStatistics network_stats;
   ASSERT_EQ(0, neteq_->NetworkStatistics(&network_stats));
   EXPECT_EQ(110946, network_stats.clockdrift_ppm);
 }
 
 void NetEqDecodingTest::LongCngWithClockDrift(double drift_factor,
                                               double network_freeze_ms,
                                               bool pull_audio_during_freeze,
                                               int delay_tolerance_ms,
                                               int max_time_to_speech_ms) {
   uint16_t seq_no = 0;
   uint32_t timestamp = 0;
   const int kFrameSizeMs = 30;
   const size_t kSamples = kFrameSizeMs * 16;
   const size_t kPayloadBytes = kSamples * 2;
   double next_input_time_ms = 0.0;
   double t_ms;
   size_t out_len;
-  int num_channels;
+  size_t num_channels;
   NetEqOutputType type;
 
   // Insert speech for 5 seconds.
   const int kSpeechDurationMs = 5000;
   for (t_ms = 0; t_ms < kSpeechDurationMs; t_ms += 10) {
     // Each turn in this for loop is 10 ms.
     while (next_input_time_ms <= t_ms) {
       // Insert one 30 ms speech frame.
       uint8_t payload[kPayloadBytes] = {0};
       WebRtcRTPHeader rtp_info;
@@ skipping 218 matching lines @@
   WebRtcRTPHeader rtp_info;
   PopulateRtpInfo(0, 0, &rtp_info);
   rtp_info.header.payloadType = 103;  // iSAC, but the payload is invalid.
   EXPECT_EQ(0, neteq_->InsertPacket(rtp_info, payload, 0));
   NetEqOutputType type;
   // Set all of |out_data_| to 1, and verify that it was set to 0 by the call
   // to GetAudio.
   for (size_t i = 0; i < kMaxBlockSize; ++i) {
     out_data_[i] = 1;
   }
-  int num_channels;
+  size_t num_channels;
   size_t samples_per_channel;
   EXPECT_EQ(NetEq::kFail,
             neteq_->GetAudio(kMaxBlockSize, out_data_,
                              &samples_per_channel, &num_channels, &type));
   // Verify that there is a decoder error to check.
   EXPECT_EQ(NetEq::kDecoderErrorCode, neteq_->LastError());
   // Code 6730 is an iSAC error code.
   EXPECT_EQ(6730, neteq_->LastDecoderError());
   // Verify that the first 160 samples are set to 0, and that the remaining
   // samples are left unmodified.
@@ skipping 13 matching lines @@
 }
 #endif
 
 TEST_F(NetEqDecodingTest, GetAudioBeforeInsertPacket) {
   NetEqOutputType type;
   // Set all of |out_data_| to 1, and verify that it was set to 0 by the call
   // to GetAudio.
   for (size_t i = 0; i < kMaxBlockSize; ++i) {
     out_data_[i] = 1;
   }
-  int num_channels;
+  size_t num_channels;
   size_t samples_per_channel;
   EXPECT_EQ(0, neteq_->GetAudio(kMaxBlockSize, out_data_,
                                 &samples_per_channel,
                                 &num_channels, &type));
   // Verify that the first block of samples is set to 0.
   static const int kExpectedOutputLength =
       kInitSampleRateHz / 100;  // 10 ms at initial sample rate.
   for (int i = 0; i < kExpectedOutputLength; ++i) {
     std::ostringstream ss;
     ss << "i = " << i;
@@ skipping 35 matching lines @@
         webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm"),
         10 * sampling_rate_hz,  // Max 10 seconds loop length.
         expected_samples_per_channel));
 
     // Payload of 10 ms of PCM16 32 kHz.
     uint8_t payload[kBlockSize32kHz * sizeof(int16_t)];
     WebRtcRTPHeader rtp_info;
     PopulateRtpInfo(0, 0, &rtp_info);
     rtp_info.header.payloadType = payload_type;
 
-    int number_channels = 0;
+    size_t number_channels = 0;
     size_t samples_per_channel = 0;
 
     uint32_t receive_timestamp = 0;
     for (int n = 0; n < 10; ++n) {  // Insert few packets and get audio.
       auto block = input.GetNextBlock();
       ASSERT_EQ(expected_samples_per_channel, block.size());
       size_t enc_len_bytes =
           WebRtcPcm16b_Encode(block.data(), block.size(), payload);
       ASSERT_EQ(enc_len_bytes, expected_samples_per_channel * 2);
 
       number_channels = 0;
       samples_per_channel = 0;
       ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, rtc::ArrayView<const uint8_t>(
                                                       payload, enc_len_bytes),
                                         receive_timestamp));
       ASSERT_EQ(0,
                 neteq_->GetAudio(kBlockSize32kHz,
                                  output,
                                  &samples_per_channel,
                                  &number_channels,
                                  &type));
-      ASSERT_EQ(1, number_channels);
+      ASSERT_EQ(1u, number_channels);
       ASSERT_EQ(expected_samples_per_channel, samples_per_channel);
       ASSERT_EQ(kOutputNormal, type);
 
       // Next packet.
       rtp_info.header.timestamp += expected_samples_per_channel;
       rtp_info.header.sequenceNumber++;
       receive_timestamp += expected_samples_per_channel;
     }
 
     number_channels = 0;
     samples_per_channel = 0;
 
     // Get audio without inserting packets, expecting PLC and PLC-to-CNG. Pull
     // one frame without checking speech-type. This is the first frame pulled
     // without inserting any packet, and might not be labeled as PLC.
     ASSERT_EQ(0,
               neteq_->GetAudio(kBlockSize32kHz,
                                output,
                                &samples_per_channel,
                                &number_channels,
                                &type));
-    ASSERT_EQ(1, number_channels);
+    ASSERT_EQ(1u, number_channels);
     ASSERT_EQ(expected_samples_per_channel, samples_per_channel);
 
     // To be able to test the fading of background noise we need at lease to
     // pull 611 frames.
     const int kFadingThreshold = 611;
 
     // Test several CNG-to-PLC packet for the expected behavior. The number 20
     // is arbitrary, but sufficiently large to test enough number of frames.
     const int kNumPlcToCngTestFrames = 20;
     bool plc_to_cng = false;
     for (int n = 0; n < kFadingThreshold + kNumPlcToCngTestFrames; ++n) {
       number_channels = 0;
       samples_per_channel = 0;
       memset(output, 1, sizeof(output));  // Set to non-zero.
       ASSERT_EQ(0,
                 neteq_->GetAudio(kBlockSize32kHz,
                                  output,
                                  &samples_per_channel,
                                  &number_channels,
                                  &type));
-      ASSERT_EQ(1, number_channels);
+      ASSERT_EQ(1u, number_channels);
       ASSERT_EQ(expected_samples_per_channel, samples_per_channel);
       if (type == kOutputPLCtoCNG) {
         plc_to_cng = true;
         double sum_squared = 0;
         for (size_t k = 0; k < number_channels * samples_per_channel; ++k)
           sum_squared += output[k] * output[k];
         TestCondition(sum_squared, n > kFadingThreshold);
       } else {
         EXPECT_EQ(kOutputPLC, type);
       }
@@ skipping 148 matching lines @@
   const size_t kPayloadBytes = kBlockSize16kHz * sizeof(int16_t);
   uint8_t payload[kPayloadBytes];
   int16_t decoded[kBlockSize16kHz];
   int algorithmic_frame_delay = algorithmic_delay_ms_ / 10 + 1;
   for (size_t n = 0; n < kPayloadBytes; ++n) {
     payload[n] = (rand() & 0xF0) + 1;  // Non-zero random sequence.
   }
   // Insert some packets which decode to noise. We are not interested in
   // actual decoded values.
   NetEqOutputType output_type;
-  int num_channels;
+  size_t num_channels;
   size_t samples_per_channel;
   uint32_t receive_timestamp = 0;
   for (int n = 0; n < 100; ++n) {
     ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, receive_timestamp));
     ASSERT_EQ(0, neteq_->GetAudio(kBlockSize16kHz, decoded,
                                   &samples_per_channel, &num_channels,
                                   &output_type));
     ASSERT_EQ(kBlockSize16kHz, samples_per_channel);
-    ASSERT_EQ(1, num_channels);
+    ASSERT_EQ(1u, num_channels);
 
     rtp_info.header.sequenceNumber++;
     rtp_info.header.timestamp += kBlockSize16kHz;
     receive_timestamp += kBlockSize16kHz;
   }
   const int kNumSyncPackets = 10;
 
   // Make sure sufficient number of sync packets are inserted that we can
   // conduct a test.
   ASSERT_GT(kNumSyncPackets, algorithmic_frame_delay);
   // Insert sync-packets, the decoded sequence should be all-zero.
   for (int n = 0; n < kNumSyncPackets; ++n) {
     ASSERT_EQ(0, neteq_->InsertSyncPacket(rtp_info, receive_timestamp));
     ASSERT_EQ(0, neteq_->GetAudio(kBlockSize16kHz, decoded,
                                   &samples_per_channel, &num_channels,
                                   &output_type));
     ASSERT_EQ(kBlockSize16kHz, samples_per_channel);
-    ASSERT_EQ(1, num_channels);
+    ASSERT_EQ(1u, num_channels);
     if (n > algorithmic_frame_delay) {
       EXPECT_TRUE(IsAllZero(decoded, samples_per_channel * num_channels));
     }
     rtp_info.header.sequenceNumber++;
     rtp_info.header.timestamp += kBlockSize16kHz;
     receive_timestamp += kBlockSize16kHz;
   }
 
   // We insert regular packets, if sync packet are not correctly buffered then
   // network statistics would show some packet loss.
@@ skipping 28 matching lines @@
   PopulateRtpInfo(0, 0, &rtp_info);
   const size_t kPayloadBytes = kBlockSize16kHz * sizeof(int16_t);
   uint8_t payload[kPayloadBytes];
   int16_t decoded[kBlockSize16kHz];
   for (size_t n = 0; n < kPayloadBytes; ++n) {
     payload[n] = (rand() & 0xF0) + 1;  // Non-zero random sequence.
   }
   // Insert some packets which decode to noise. We are not interested in
   // actual decoded values.
   NetEqOutputType output_type;
-  int num_channels;
+  size_t num_channels;
   size_t samples_per_channel;
   uint32_t receive_timestamp = 0;
   int algorithmic_frame_delay = algorithmic_delay_ms_ / 10 + 1;
   for (int n = 0; n < algorithmic_frame_delay; ++n) {
     ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, receive_timestamp));
     ASSERT_EQ(0, neteq_->GetAudio(kBlockSize16kHz, decoded,
                                   &samples_per_channel, &num_channels,
                                   &output_type));
     ASSERT_EQ(kBlockSize16kHz, samples_per_channel);
-    ASSERT_EQ(1, num_channels);
+    ASSERT_EQ(1u, num_channels);
     rtp_info.header.sequenceNumber++;
     rtp_info.header.timestamp += kBlockSize16kHz;
     receive_timestamp += kBlockSize16kHz;
   }
   const int kNumSyncPackets = 10;
 
   WebRtcRTPHeader first_sync_packet_rtp_info;
   memcpy(&first_sync_packet_rtp_info, &rtp_info, sizeof(rtp_info));
 
   // Insert sync-packets, but no decoding.
@@ skipping 18 matching lines @@
     rtp_info.header.timestamp += kBlockSize16kHz;
     receive_timestamp += kBlockSize16kHz;
   }
 
   // Decode.
   for (int n = 0; n < kNumSyncPackets; ++n) {
     ASSERT_EQ(0, neteq_->GetAudio(kBlockSize16kHz, decoded,
                                   &samples_per_channel, &num_channels,
                                   &output_type));
     ASSERT_EQ(kBlockSize16kHz, samples_per_channel);
-    ASSERT_EQ(1, num_channels);
+    ASSERT_EQ(1u, num_channels);
     EXPECT_TRUE(IsAllNonZero(decoded, samples_per_channel * num_channels));
   }
 }
 
 void NetEqDecodingTest::WrapTest(uint16_t start_seq_no,
                                  uint32_t start_timestamp,
                                  const std::set<uint16_t>& drop_seq_numbers,
                                  bool expect_seq_no_wrap,
                                  bool expect_timestamp_wrap) {
   uint16_t seq_no = start_seq_no;
   uint32_t timestamp = start_timestamp;
   const int kBlocksPerFrame = 3;  // Number of 10 ms blocks per frame.
   const int kFrameSizeMs = kBlocksPerFrame * kTimeStepMs;
   const int kSamples = kBlockSize16kHz * kBlocksPerFrame;
   const size_t kPayloadBytes = kSamples * sizeof(int16_t);
   double next_input_time_ms = 0.0;
   int16_t decoded[kBlockSize16kHz];
-  int num_channels;
+  size_t num_channels;
   size_t samples_per_channel;
   NetEqOutputType output_type;
   uint32_t receive_timestamp = 0;
 
   // Insert speech for 2 seconds.
   const int kSpeechDurationMs = 2000;
   int packets_inserted = 0;
   uint16_t last_seq_no;
   uint32_t last_timestamp;
   bool timestamp_wrapped = false;
@@ skipping 32 matching lines @@
       next_input_time_ms += static_cast<double>(kFrameSizeMs);
 
       seq_no_wrapped |= seq_no < last_seq_no;
       timestamp_wrapped |= timestamp < last_timestamp;
     }
     // Pull out data once.
     ASSERT_EQ(0, neteq_->GetAudio(kBlockSize16kHz, decoded,
                                   &samples_per_channel, &num_channels,
                                   &output_type));
     ASSERT_EQ(kBlockSize16kHz, samples_per_channel);
-    ASSERT_EQ(1, num_channels);
+    ASSERT_EQ(1u, num_channels);
 
     // Expect delay (in samples) to be less than 2 packets.
     EXPECT_LE(timestamp - PlayoutTimestamp(),
               static_cast<uint32_t>(kSamples * 2));
   }
   // Make sure we have actually tested wrap-around.
   ASSERT_EQ(expect_seq_no_wrap, seq_no_wrapped);
   ASSERT_EQ(expect_timestamp_wrap, timestamp_wrapped);
 }
 
@@ skipping 30 matching lines @@
   const int kFrameSizeMs = 10;
   const int kSampleRateKhz = 16;
   const int kSamples = kFrameSizeMs * kSampleRateKhz;
   const size_t kPayloadBytes = kSamples * 2;
 
   const int algorithmic_delay_samples = std::max(
       algorithmic_delay_ms_ * kSampleRateKhz, 5 * kSampleRateKhz / 8);
   // Insert three speech packets. Three are needed to get the frame length
   // correct.
   size_t out_len;
-  int num_channels;
+  size_t num_channels;
   NetEqOutputType type;
   uint8_t payload[kPayloadBytes] = {0};
   WebRtcRTPHeader rtp_info;
   for (int i = 0; i < 3; ++i) {
     PopulateRtpInfo(seq_no, timestamp, &rtp_info);
     ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, 0));
     ++seq_no;
     timestamp += kSamples;
 
     // Pull audio once.
@@ skipping 82 matching lines @@
   PopulateCng(seq_no, timestamp, &rtp_info, payload, &payload_len);
   ASSERT_EQ(
       NetEq::kOK,
       neteq_->InsertPacket(
           rtp_info, rtc::ArrayView<const uint8_t>(payload, payload_len), 0));
   ++seq_no;
   timestamp += kCngPeriodSamples;
 
   // Pull audio once and make sure CNG is played.
   size_t out_len;
-  int num_channels;
+  size_t num_channels;
   NetEqOutputType type;
   ASSERT_EQ(0, neteq_->GetAudio(kMaxBlockSize, out_data_, &out_len,
                                 &num_channels, &type));
   ASSERT_EQ(kBlockSize16kHz, out_len);
   EXPECT_EQ(kOutputCNG, type);
 
   // Insert some speech packets.
   for (int i = 0; i < 3; ++i) {
     PopulateRtpInfo(seq_no, timestamp, &rtp_info);
     ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload, 0));
     ++seq_no;
     timestamp += kSamples;
 
     // Pull audio once.
     ASSERT_EQ(0, neteq_->GetAudio(kMaxBlockSize, out_data_, &out_len,
                                   &num_channels, &type));
     ASSERT_EQ(kBlockSize16kHz, out_len);
   }
   // Verify speech output.
   EXPECT_EQ(kOutputNormal, type);
 }
 
 }  // namespace webrtc