Change NetEq::GetAudio to use AudioFrame

webrtc/modules/audio_coding/neteq/neteq_stereo_unittest.cc

 /*
  *  Copyright (c) 2013 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.
  */
 
 // Test to verify correct stereo and multi-channel operation.
 
 #include <algorithm>
 #include <memory>
 #include <string>
 #include <list>
 
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/modules/audio_coding/codecs/pcm16b/pcm16b.h"
 #include "webrtc/modules/audio_coding/neteq/include/neteq.h"
 #include "webrtc/modules/audio_coding/neteq/tools/input_audio_file.h"
 #include "webrtc/modules/audio_coding/neteq/tools/rtp_generator.h"
+#include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/test/testsupport/fileutils.h"
 
 namespace webrtc {
 
 struct TestParameters {
   int frame_size;
   int sample_rate;
   size_t num_channels;
 };
 
@@ skipping 29 matching lines @@
         last_arrival_time_(0) {
     NetEq::Config config;
     config.sample_rate_hz = sample_rate_hz_;
     neteq_mono_ = NetEq::Create(config);
     neteq_ = NetEq::Create(config);
     input_ = new int16_t[frame_size_samples_];
     encoded_ = new uint8_t[2 * frame_size_samples_];
     input_multi_channel_ = new int16_t[frame_size_samples_ * num_channels_];
     encoded_multi_channel_ = new uint8_t[frame_size_samples_ * 2 *
                                          num_channels_];
-    output_multi_channel_ = new int16_t[kMaxBlockSize * num_channels_];
   }
 
   ~NetEqStereoTest() {
     delete neteq_mono_;
     delete neteq_;
     delete [] input_;
     delete [] encoded_;
     delete [] input_multi_channel_;
     delete [] encoded_multi_channel_;
-    delete [] output_multi_channel_;
   }
 
   virtual void SetUp() {
     const std::string file_name =
         webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm");
     input_file_.reset(new test::InputAudioFile(file_name));
     NetEqDecoder mono_decoder;
     NetEqDecoder multi_decoder;
     switch (sample_rate_hz_) {
       case 8000:
@@ skipping 64 matching lines @@
       return -1;
     }
     rtp_generator_.GetRtpHeader(kPayloadTypeMulti, frame_size_samples_,
                                 &rtp_header_);
     return next_send_time;
   }
 
   virtual void VerifyOutput(size_t num_samples) {
     for (size_t i = 0; i < num_samples; ++i) {
       for (size_t j = 0; j < num_channels_; ++j) {
-        ASSERT_EQ(output_[i], output_multi_channel_[i * num_channels_ + j]) <<
-            "Diff in sample " << i << ", channel " << j << ".";
+        ASSERT_EQ(output_.data_[i],
+                  output_multi_channel_.data_[i * num_channels_ + j])
+            << "Diff in sample " << i << ", channel " << j << ".";
       }
     }
   }
 
   virtual int GetArrivalTime(int send_time) {
     int arrival_time = last_arrival_time_ + (send_time - last_send_time_);
     last_send_time_ = send_time;
     last_arrival_time_ = arrival_time;
     return arrival_time;
   }
@@ skipping 27 matching lines @@
                                   next_arrival_time));
         // Get next input packets (mono and multi-channel).
         do {
           next_send_time = GetNewPackets();
           ASSERT_NE(-1, next_send_time);
           next_arrival_time = GetArrivalTime(next_send_time);
         } while (Lost());  // If lost, immediately read the next packet.
       }
       NetEqOutputType output_type;
       // Get audio from mono instance.
-      size_t samples_per_channel;
-      size_t num_channels;
-      EXPECT_EQ(NetEq::kOK,
-                neteq_mono_->GetAudio(kMaxBlockSize, output_,
-                                      &samples_per_channel, &num_channels,
-                                      &output_type));
-      EXPECT_EQ(1u, num_channels);
-      EXPECT_EQ(output_size_samples_, samples_per_channel);
+      EXPECT_EQ(NetEq::kOK, neteq_mono_->GetAudio(&output_, &output_type));
+      EXPECT_EQ(1u, output_.num_channels_);
+      EXPECT_EQ(output_size_samples_, output_.samples_per_channel_);
       // Get audio from multi-channel instance.
       ASSERT_EQ(NetEq::kOK,
-                neteq_->GetAudio(kMaxBlockSize * num_channels_,
-                                 output_multi_channel_,
-                                 &samples_per_channel, &num_channels,
-                                 &output_type));
-      EXPECT_EQ(num_channels_, num_channels);
-      EXPECT_EQ(output_size_samples_, samples_per_channel);
+                neteq_->GetAudio(&output_multi_channel_, &output_type));
+      EXPECT_EQ(num_channels_, output_multi_channel_.num_channels_);
+      EXPECT_EQ(output_size_samples_,
+                output_multi_channel_.samples_per_channel_);
       std::ostringstream ss;
       ss << "Lap number " << k << ".";
       SCOPED_TRACE(ss.str());  // Print out the parameter values on failure.
       // Compare mono and multi-channel.
       ASSERT_NO_FATAL_FAILURE(VerifyOutput(output_size_samples_));
 
       time_now += kTimeStepMs;
     }
   }
 
   const size_t num_channels_;
   const int sample_rate_hz_;
   const int samples_per_ms_;
   const int frame_size_ms_;
   const size_t frame_size_samples_;
   const size_t output_size_samples_;
   NetEq* neteq_mono_;
   NetEq* neteq_;
   test::RtpGenerator rtp_generator_mono_;
   test::RtpGenerator rtp_generator_;
   int16_t* input_;
   int16_t* input_multi_channel_;
   uint8_t* encoded_;
   uint8_t* encoded_multi_channel_;
-  int16_t output_[kMaxBlockSize];
-  int16_t* output_multi_channel_;
+  AudioFrame output_;
+  AudioFrame output_multi_channel_;
   WebRtcRTPHeader rtp_header_mono_;
   WebRtcRTPHeader rtp_header_;
   size_t payload_size_bytes_;
   size_t multi_payload_size_bytes_;
   int last_send_time_;
   int last_arrival_time_;
   std::unique_ptr<test::InputAudioFile> input_file_;
 };
 
 class NetEqStereoTestNoJitter : public NetEqStereoTest {
@@ skipping 85 matching lines @@
         frame_index_(0) {
   }
 
   virtual bool Lost() {
     return (++frame_index_) % kLossInterval == 0;
   }
 
   // TODO(hlundin): NetEq is not giving bitexact results for these cases.
   virtual void VerifyOutput(size_t num_samples) {
     for (size_t i = 0; i < num_samples; ++i) {
-      auto first_channel_sample = output_multi_channel_[i * num_channels_];
+      auto first_channel_sample =
+          output_multi_channel_.data_[i * num_channels_];
       for (size_t j = 0; j < num_channels_; ++j) {
         const int kErrorMargin = 200;
-        EXPECT_NEAR(output_[i], output_multi_channel_[i * num_channels_ + j],
+        EXPECT_NEAR(output_.data_[i],
+                    output_multi_channel_.data_[i * num_channels_ + j],
                     kErrorMargin)
             << "Diff in sample " << i << ", channel " << j << ".";
         EXPECT_EQ(first_channel_sample,
-                  output_multi_channel_[i * num_channels_ + j]);
+                  output_multi_channel_.data_[i * num_channels_ + j]);
       }
     }
   }
 
   int frame_index_;
 };
 
 TEST_P(NetEqStereoTestLosses, RunTest) {
   RunTest(100);
 }
@@ skipping 47 matching lines @@
 
 INSTANTIATE_TEST_CASE_P(MultiChannel,
                         NetEqStereoTestDelays,
                         ::testing::ValuesIn(GetTestParameters()));
 
 INSTANTIATE_TEST_CASE_P(MultiChannel,
                         NetEqStereoTestLosses,
                         ::testing::ValuesIn(GetTestParameters()));
 
 }  // namespace webrtc