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

webrtc/modules/audio_coding/neteq/test/neteq_opus_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 86 matching lines @@
     RegisterFlagValidator(&FLAGS_sub_packets, &ValidateSubPackets);
 
 }  // namepsace
 
 class NetEqOpusQualityTest : public NetEqQualityTest {
  protected:
   NetEqOpusQualityTest();
   void SetUp() override;
   void TearDown() override;
   virtual int EncodeBlock(int16_t* in_data, size_t block_size_samples,
-                          uint8_t* payload, size_t max_bytes);
+                          rtc::Buffer* payload, size_t max_bytes);
  private:
   WebRtcOpusEncInst* opus_encoder_;
   OpusRepacketizer* repacketizer_;
   size_t sub_block_size_samples_;
   int bit_rate_kbps_;
   bool fec_;
   bool dtx_;
   int complexity_;
   int maxplaybackrate_;
   int target_loss_rate_;
@@ skipping 50 matching lines @@
 
 void NetEqOpusQualityTest::TearDown() {
   // Free memory.
   EXPECT_EQ(0, WebRtcOpus_EncoderFree(opus_encoder_));
   opus_repacketizer_destroy(repacketizer_);
   NetEqQualityTest::TearDown();
 }
 
 int NetEqOpusQualityTest::EncodeBlock(int16_t* in_data,
                                       size_t block_size_samples,
-                                      uint8_t* payload, size_t max_bytes) {
+                                      rtc::Buffer* payload, size_t max_bytes) {
   EXPECT_EQ(block_size_samples, sub_block_size_samples_ * sub_packets_);
   int16_t* pointer = in_data;
   int value;
   opus_repacketizer_init(repacketizer_);
   for (int idx = 0; idx < sub_packets_; idx++) {
-    value = WebRtcOpus_Encode(opus_encoder_, pointer, sub_block_size_samples_,
-                              max_bytes, payload);
-    Log() << "Encoded a frame with Opus mode "
-          << (value == 0 ? 0 : payload[0] >> 3)
-          << std::endl;
-    if (OPUS_OK != opus_repacketizer_cat(repacketizer_, payload, value)) {
+    payload->AppendData(max_bytes, [&] (rtc::ArrayView<uint8_t> payload) {
+        value = WebRtcOpus_Encode(opus_encoder_,
+                                  pointer, sub_block_size_samples_,
+                                  max_bytes, payload.data());
+
+        Log() << "Encoded a frame with Opus mode "
+              << (value == 0 ? 0 : payload[0] >> 3)
+              << std::endl;
+
+        return (value >= 0) ? static_cast<size_t>(value) : 0;
+      });
+
+    if (OPUS_OK != opus_repacketizer_cat(repacketizer_,
+                                         payload->data(), value)) {
       opus_repacketizer_init(repacketizer_);
       // If the repacketization fails, we discard this frame.
       return 0;
     }
     pointer += sub_block_size_samples_ * channels_;
   }
-  value = opus_repacketizer_out(repacketizer_, payload,
+  value = opus_repacketizer_out(repacketizer_, payload->data(),
                                 static_cast<opus_int32>(max_bytes));
   EXPECT_GE(value, 0);
   return value;
 }
 
 TEST_F(NetEqOpusQualityTest, Test) {
   Simulate();
 }
 
 }  // namespace test
 }  // namespace webrtc