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

webrtc/modules/audio_coding/neteq/tools/neteq_performance_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 44 matching lines @@
   int32_t time_now_ms = 0;
 
   // Get first input packet.
   WebRtcRTPHeader rtp_header;
   RtpGenerator rtp_gen(kSampRateHz / 1000);
   // Start with positive drift first half of simulation.
   rtp_gen.set_drift_factor(drift_factor);
   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();
-  if (!input_samples) exit(1);
+  auto input_samples = audio_loop.GetNextBlock();
+  if (input_samples.empty())
+    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.
   webrtc::Clock* clock = webrtc::Clock::GetRealTimeClock();
   int64_t start_time_ms = clock->TimeInMilliseconds();
   while (time_now_ms < runtime_ms) {
     while (packet_input_time_ms <= time_now_ms) {
       // Drop every N packets, where N = FLAGS_lossrate.
       bool lost = false;
       if (lossrate > 0) {
         lost = ((rtp_header.header.sequenceNumber - 1) % lossrate) == 0;
       }
       if (!lost) {
         // Insert packet.
         int error = neteq->InsertPacket(
             rtp_header, input_payload, payload_len,
             packet_input_time_ms * kSampRateHz / 1000);
         if (error != NetEq::kOK)
           return -1;
       }
 
       // Get next packet.
       packet_input_time_ms = rtp_gen.GetRtpHeader(kPayloadType,
                                                   kInputBlockSizeSamples,
                                                   &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);
+      if (input_samples.empty())
+        return -1;
+      payload_len = WebRtcPcm16b_Encode(input_samples.data(),
+                                        input_samples.size(), input_payload);
       assert(payload_len == kInputBlockSizeSamples * sizeof(int16_t));
     }
 
     // Get output audio, but don't do anything with it.
     static const int kMaxChannels = 1;
     static const size_t kMaxSamplesPerMs = 48000 / 1000;
     static const int kOutputBlockSizeMs = 10;
     static const size_t kOutDataLen =
         kOutputBlockSizeMs * kMaxSamplesPerMs * kMaxChannels;
     int16_t out_data[kOutDataLen];
@@ skipping 13 matching lines @@
       drift_flipped = true;
     }
   }
   int64_t end_time_ms = clock->TimeInMilliseconds();
   delete neteq;
   return end_time_ms - start_time_ms;
 }
 
 }  // namespace test
 }  // namespace webrtc