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

Index: webrtc/modules/audio_coding/neteq/tools/neteq_quality_test.cc
diff --git a/webrtc/modules/audio_coding/neteq/tools/neteq_quality_test.cc b/webrtc/modules/audio_coding/neteq/tools/neteq_quality_test.cc
index 9c64e0fb4823ecf80a6d4f03e456e3111e369455..7118f4ed990025de6a3104d7848b7b3e2d0d9ad6 100644
--- a/webrtc/modules/audio_coding/neteq/tools/neteq_quality_test.cc
+++ b/webrtc/modules/audio_coding/neteq/tools/neteq_quality_test.cc
@@ -249,7 +249,6 @@ NetEqQualityTest::NetEqQualityTest(int block_duration_ms,
   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_]);
 }
 
@@ -380,7 +379,7 @@ int NetEqQualityTest::Transmit() {
     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;
@@ -416,8 +415,9 @@ void NetEqQualityTest::Simulate() {
     // 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_;