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

webrtc/modules/audio_coding/main/acm2/audio_coding_module_unittest_oldapi.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 638 matching lines @@
       last_packet_number_ = num_calls;
     }
     ASSERT_GT(last_payload_vec_.size(), 0u);
     ASSERT_EQ(
         0,
         acm_->IncomingPacket(
             &last_payload_vec_[0], last_payload_vec_.size(), rtp_header_));
   }
 
   void InsertAudio() {
-    memcpy(input_frame_.data_, audio_loop_.GetNextBlock(), kNumSamples10ms);
+    // TODO(kwiberg): Use std::copy here. Might be complications because AFAICS
+    // this call confuses the number of samples with the number of bytes, and
+    // ends up copying only half of what it should.
+    memcpy(input_frame_.data_, audio_loop_.GetNextBlock().data(),
+           kNumSamples10ms);
     AudioCodingModuleTestOldApi::InsertAudio();
   }
 
   // Override the verification function with no-op, since iSAC produces variable
   // payload sizes.
   void VerifyEncoding() override {}
 
   // This method is the same as AudioCodingModuleMtTestOldApi::TestDone(), but
   // here it is using the constants defined in this class (i.e., shorter test
   // run).
@@ skipping 97 matching lines @@
       CriticalSectionScoped lock(crit_sect_.get());
       if (clock_->TimeInMilliseconds() < next_insert_packet_time_ms_) {
         return true;
       }
       next_insert_packet_time_ms_ += kPacketSizeMs;
       ++receive_packet_count_;
 
       // Encode new frame.
       uint32_t input_timestamp = rtp_header_.header.timestamp;
       while (info.encoded_bytes == 0) {
-        info = isac_encoder_->Encode(
-            input_timestamp, audio_loop_.GetNextBlock(), kNumSamples10ms,
-            max_encoded_bytes, encoded.get());
+        info =
+            isac_encoder_->Encode(input_timestamp, audio_loop_.GetNextBlock(),
+                                  max_encoded_bytes, encoded.get());
         input_timestamp += 160;  // 10 ms at 16 kHz.
       }
       EXPECT_EQ(rtp_header_.header.timestamp + kPacketSizeSamples,
                 input_timestamp);
       EXPECT_EQ(rtp_header_.header.timestamp, info.encoded_timestamp);
       EXPECT_EQ(rtp_header_.header.payloadType, info.payload_type);
     }
     // Now we're not holding the crit sect when calling ACM.
 
     // Insert into ACM.
@@ skipping 969 matching lines @@
   Run(16000, 8000, 1000);
 }
 
 TEST_F(AcmSwitchingOutputFrequencyOldApi, Toggle8KhzTo16Khz) {
   Run(8000, 16000, 1000);
 }
 
 #endif
 
 }  // namespace webrtc