Pass audio to AudioEncoder::Encode() in an ArrayView

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

Index: webrtc/modules/audio_coding/neteq/tools/neteq_performance_test.cc
diff --git a/webrtc/modules/audio_coding/neteq/tools/neteq_performance_test.cc b/webrtc/modules/audio_coding/neteq/tools/neteq_performance_test.cc
index 9fe4dffa91e145dae134e2fbb677efda8b91cf70..bf2b4b7d0a4b9ff80f10ed95bc1c67cbb34823fa 100644
--- a/webrtc/modules/audio_coding/neteq/tools/neteq_performance_test.cc
+++ b/webrtc/modules/audio_coding/neteq/tools/neteq_performance_test.cc
@@ -62,11 +62,11 @@ int64_t NetEqPerformanceTest::Run(int runtime_ms,
   bool drift_flipped = false;
   int32_t packet_input_time_ms =
       rtp_gen.GetRtpHeader(kPayloadType, kInputBlockSizeSamples, &rtp_header);
-  const int16_t* input_samples = audio_loop.GetNextBlock();
+  auto input_samples = audio_loop.GetNextBlock();
   if (!input_samples) exit(1);
   uint8_t input_payload[kInputBlockSizeSamples * sizeof(int16_t)];
-  size_t payload_len =
-      WebRtcPcm16b_Encode(input_samples, kInputBlockSizeSamples, input_payload);
+  size_t payload_len = WebRtcPcm16b_Encode(input_samples.data(),
+                                           input_samples.size(), input_payload);
   assert(payload_len == kInputBlockSizeSamples * sizeof(int16_t));
 
   // Main loop.
@@ -94,9 +94,8 @@ int64_t NetEqPerformanceTest::Run(int runtime_ms,
                                                   &rtp_header);
       input_samples = audio_loop.GetNextBlock();
       if (!input_samples) return -1;
-      payload_len = WebRtcPcm16b_Encode(const_cast<int16_t*>(input_samples),
-                                        kInputBlockSizeSamples,
-                                        input_payload);
+      payload_len = WebRtcPcm16b_Encode(input_samples.data(),
+                                        input_samples.size(), input_payload);
       assert(payload_len == kInputBlockSizeSamples * sizeof(int16_t));
     }