AcmReceiver::InsertPacket and NetEq::InsertPacket: Take ArrayView arguments

webrtc/modules/audio_coding/neteq/neteq_impl.h

 /*
  *  Copyright (c) 2012 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 61 matching lines @@
             PreemptiveExpandFactory* preemptive_expand_factory,
             bool create_components = true);
 
   ~NetEqImpl() override;
 
   // Inserts a new packet into NetEq. The |receive_timestamp| is an indication
   // of the time when the packet was received, and should be measured with
   // the same tick rate as the RTP timestamp of the current payload.
   // Returns 0 on success, -1 on failure.
   int InsertPacket(const WebRtcRTPHeader& rtp_header,
-                   const uint8_t* payload,
-                   size_t length_bytes,
+                   rtc::ArrayView<const uint8_t> payload,
                    uint32_t receive_timestamp) override;
 
   // Inserts a sync-packet into packet queue. Sync-packets are decoded to
   // silence and are intended to keep AV-sync intact in an event of long packet
   // losses when Video NACK is enabled but Audio NACK is not. Clients of NetEq
   // might insert sync-packet when they observe that buffer level of NetEq is
   // decreasing below a certain threshold, defined by the application.
   // Sync-packets should have the same payload type as the last audio payload
   // type, i.e. they cannot have DTMF or CNG payload type, nor a codec change
   // can be implied by inserting a sync-packet.
@@ skipping 106 matching lines @@
  protected:
   static const int kOutputSizeMs = 10;
   static const size_t kMaxFrameSize = 2880;  // 60 ms @ 48 kHz.
   // TODO(hlundin): Provide a better value for kSyncBufferSize.
   static const size_t kSyncBufferSize = 2 * kMaxFrameSize;
 
   // Inserts a new packet into NetEq. This is used by the InsertPacket method
   // above. Returns 0 on success, otherwise an error code.
   // TODO(hlundin): Merge this with InsertPacket above?
   int InsertPacketInternal(const WebRtcRTPHeader& rtp_header,
-                           const uint8_t* payload,
-                           size_t length_bytes,
+                           rtc::ArrayView<const uint8_t> payload,
                            uint32_t receive_timestamp,
                            bool is_sync_packet)
       EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
 
   // Delivers 10 ms of audio data. The data is written to |output|, which can
   // hold (at least) |max_length| elements. The number of channels that were
   // written to the output is provided in the output variable |num_channels|,
   // and each channel contains |samples_per_channel| elements. If more than one
   // channel is written, the samples are interleaved.
   // Returns 0 on success, otherwise an error code.
@@ skipping 176 matching lines @@
   bool enable_fast_accelerate_ GUARDED_BY(crit_sect_);
   rtc::scoped_ptr<Nack> nack_ GUARDED_BY(crit_sect_);
   bool nack_enabled_ GUARDED_BY(crit_sect_);
 
  private:
   RTC_DISALLOW_COPY_AND_ASSIGN(NetEqImpl);
 };
 
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_AUDIO_CODING_NETEQ_NETEQ_IMPL_H_