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

webrtc/modules/audio_coding/codecs/audio_encoder.h

 /*
  *  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.
  */
 
 #ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_AUDIO_ENCODER_H_
 #define WEBRTC_MODULES_AUDIO_CODING_CODECS_AUDIO_ENCODER_H_
 
 #include <algorithm>
 #include <vector>
 
+#include "webrtc/base/array_view.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
 
 // This is the interface class for encoders in AudioCoding module. Each codec
 // type must have an implementation of this class.
 class AudioEncoder {
  public:
   struct EncodedInfoLeaf {
     size_t encoded_bytes = 0;
@@ skipping 57 matching lines @@
 
   // Accepts one 10 ms block of input audio (i.e., SampleRateHz() / 100 *
   // NumChannels() samples). Multi-channel audio must be sample-interleaved.
   // The encoder produces zero or more bytes of output in |encoded| and
   // returns additional encoding information.
   // The caller is responsible for making sure that |max_encoded_bytes| is
   // not smaller than the number of bytes actually produced by the encoder.
   // Encode() checks some preconditions, calls EncodeInternal() which does the
   // actual work, and then checks some postconditions.
   EncodedInfo Encode(uint32_t rtp_timestamp,
-                     const int16_t* audio,
-                     size_t num_samples_per_channel,
+                     rtc::ArrayView<const int16_t> audio,
                      size_t max_encoded_bytes,
                      uint8_t* encoded);
 
   virtual EncodedInfo EncodeInternal(uint32_t rtp_timestamp,
-                                     const int16_t* audio,
+                                     rtc::ArrayView<const int16_t> audio,
                                      size_t max_encoded_bytes,
                                      uint8_t* encoded) = 0;
 
   // Resets the encoder to its starting state, discarding any input that has
   // been fed to the encoder but not yet emitted in a packet.
   virtual void Reset() = 0;
 
   // Enables or disables codec-internal FEC (forward error correction). Returns
   // true if the codec was able to comply. The default implementation returns
   // true when asked to disable FEC and false when asked to enable it (meaning
@@ skipping 23 matching lines @@
   // does nothing.
   virtual void SetProjectedPacketLossRate(double fraction);
 
   // Tells the encoder what average bitrate we'd like it to produce. The
   // encoder is free to adjust or disregard the given bitrate (the default
   // implementation does the latter).
   virtual void SetTargetBitrate(int target_bps);
 };
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_AUDIO_ENCODER_H_