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

webrtc/modules/audio_coding/main/acm2/acm_receiver.h

 /*
  *  Copyright (c) 2013 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_MAIN_ACM2_ACM_RECEIVER_H_
 #define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_RECEIVER_H_
 
 #include <map>
 #include <vector>
 
+#include "webrtc/base/array_view.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/thread_annotations.h"
 #include "webrtc/common_audio/vad/include/webrtc_vad.h"
 #include "webrtc/engine_configurations.h"
 #include "webrtc/modules/audio_coding/main/include/audio_coding_module.h"
 #include "webrtc/modules/audio_coding/main/acm2/acm_codec_database.h"
 #include "webrtc/modules/audio_coding/main/acm2/acm_resampler.h"
 #include "webrtc/modules/audio_coding/main/acm2/call_statistics.h"
 #include "webrtc/modules/audio_coding/main/acm2/initial_delay_manager.h"
 #include "webrtc/modules/audio_coding/neteq/include/neteq.h"
@@ skipping 32 matching lines @@
   //   - rtp_header           : RTP header for the incoming payload containing
   //                            information about payload type, sequence number,
   //                            timestamp, SSRC and marker bit.
   //   - incoming_payload     : Incoming audio payload.
   //   - length_payload       : Length of incoming audio payload in bytes.
   //
   // Return value             : 0 if OK.
   //                           <0 if NetEq returned an error.
   //
   int InsertPacket(const WebRtcRTPHeader& rtp_header,
-                   const uint8_t* incoming_payload,
-                   size_t length_payload);
+                   rtc::ArrayView<const uint8_t> incoming_payload);
 
   //
   // Asks NetEq for 10 milliseconds of decoded audio.
   //
   // Input:
   //   -desired_freq_hz       : specifies the sampling rate [Hz] of the output
   //                            audio. If set -1 indicates to resampling is
   //                            is required and the audio returned at the
   //                            sampling rate of the decoder.
   //
@@ skipping 191 matching lines @@
   // Return value           : list of packets to be retransmitted.
   //
   std::vector<uint16_t> GetNackList(int64_t round_trip_time_ms) const;
 
   //
   // Get statistics of calls to GetAudio().
   void GetDecodingCallStatistics(AudioDecodingCallStats* stats) const;
 
  private:
   const Decoder* RtpHeaderToDecoder(const RTPHeader& rtp_header,
-                                    const uint8_t* payload) const
+                                    uint8_t payload_type) const
       EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
 
   uint32_t NowInTimestamp(int decoder_sampling_rate) const;
 
   rtc::scoped_ptr<CriticalSectionWrapper> crit_sect_;
   int id_;  // TODO(henrik.lundin) Make const.
   const Decoder* last_audio_decoder_ GUARDED_BY(crit_sect_);
   AudioFrame::VADActivity previous_audio_activity_ GUARDED_BY(crit_sect_);
   int current_sample_rate_hz_ GUARDED_BY(crit_sect_);
   ACMResampler resampler_ GUARDED_BY(crit_sect_);
   // Used in GetAudio, declared as member to avoid allocating every 10ms.
   // TODO(henrik.lundin) Stack-allocate in GetAudio instead?
   rtc::scoped_ptr<int16_t[]> audio_buffer_ GUARDED_BY(crit_sect_);
   rtc::scoped_ptr<int16_t[]> last_audio_buffer_ GUARDED_BY(crit_sect_);
   CallStatistics call_stats_ GUARDED_BY(crit_sect_);
   NetEq* neteq_;
   // Decoders map is keyed by payload type
   std::map<uint8_t, Decoder> decoders_ GUARDED_BY(crit_sect_);
   bool vad_enabled_;
   Clock* clock_;  // TODO(henrik.lundin) Make const if possible.
   bool resampled_last_output_frame_ GUARDED_BY(crit_sect_);
 };
 
 }  // namespace acm2
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_RECEIVER_H_