Changed AudioEncoder::Encode to take an rtc::Buffer* instead of uint8_t* and a maximum size.

webrtc/modules/audio_coding/neteq/tools/neteq_quality_test.cc

 /*
  *  Copyright (c) 2014 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 231 matching lines @@
   } else {
     // Open a pcm file.
     output_.reset(new webrtc::test::OutputAudioFile(out_filename));
   }
 
   NetEq::Config config;
   config.sample_rate_hz = out_sampling_khz_ * 1000;
   neteq_.reset(NetEq::Create(config));
   max_payload_bytes_ = in_size_samples_ * channels_ * sizeof(int16_t);
   in_data_.reset(new int16_t[in_size_samples_ * channels_]);
-  payload_.reset(new uint8_t[max_payload_bytes_]);
   out_data_.reset(new int16_t[out_size_samples_ * channels_]);
 }
 
 NetEqQualityTest::~NetEqQualityTest() {
   log_file_.close();
 }
 
 bool NoLoss::Lost() {
   return false;
 }
@@ skipping 110 matching lines @@
                                    &rtp_header_);
   Log() << "Packet of size "
         << payload_size_bytes_
         << " bytes, for frame at "
         << packet_input_time_ms
         << " ms ";
   if (payload_size_bytes_ > 0) {
     if (!PacketLost()) {
       int ret = neteq_->InsertPacket(
           rtp_header_,
-          rtc::ArrayView<const uint8_t>(payload_.get(), payload_size_bytes_),
+          rtc::ArrayView<const uint8_t>(payload_.data(), payload_size_bytes_),
           packet_input_time_ms * in_sampling_khz_);
       if (ret != NetEq::kOK)
         return -1;
       Log() << "was sent.";
     } else {
       Log() << "was lost.";
     }
   }
   Log() << std::endl;
   return packet_input_time_ms;
@@ skipping 15 matching lines @@
   }
 }
 
 void NetEqQualityTest::Simulate() {
   int audio_size_samples;
 
   while (decoded_time_ms_ < FLAGS_runtime_ms) {
     // Assume 10 packets in packets buffer.
     while (decodable_time_ms_ - 10 * block_duration_ms_ < decoded_time_ms_) {
       ASSERT_TRUE(in_file_->Read(in_size_samples_ * channels_, &in_data_[0]));
+      payload_.Clear();
       payload_size_bytes_ = EncodeBlock(&in_data_[0],
-                                        in_size_samples_, &payload_[0],
+                                        in_size_samples_, &payload_,
                                         max_payload_bytes_);
       total_payload_size_bytes_ += payload_size_bytes_;
       decodable_time_ms_ = Transmit() + block_duration_ms_;
     }
     audio_size_samples = DecodeBlock();
     if (audio_size_samples > 0) {
       decoded_time_ms_ += audio_size_samples / out_sampling_khz_;
     }
   }
   Log() << "Average bit rate was "
         << 8.0f * total_payload_size_bytes_ / FLAGS_runtime_ms
         << " kbps"
         << std::endl;
 }
 
 }  // namespace test
 }  // namespace webrtc