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

webrtc/modules/audio_coding/neteq/neteq_impl.cc

 /*
  *  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 103 matching lines @@
   }
   RTC_DCHECK(!vad_->enabled());
   if (config.enable_post_decode_vad) {
     vad_->Enable();
   }
 }
 
 NetEqImpl::~NetEqImpl() = default;
 
 int NetEqImpl::InsertPacket(const WebRtcRTPHeader& rtp_header,
-                            const uint8_t* payload,
-                            size_t length_bytes,
+                            rtc::ArrayView<const uint8_t> payload,
                             uint32_t receive_timestamp) {
   CriticalSectionScoped lock(crit_sect_.get());
-  LOG(LS_VERBOSE) << "InsertPacket: ts=" << rtp_header.header.timestamp <<
-      ", sn=" << rtp_header.header.sequenceNumber <<
-      ", pt=" << static_cast<int>(rtp_header.header.payloadType) <<
-      ", ssrc=" << rtp_header.header.ssrc <<
-      ", len=" << length_bytes;
-  int error = InsertPacketInternal(rtp_header, payload, length_bytes,
-                                   receive_timestamp, false);
+  LOG(LS_VERBOSE) << "InsertPacket: ts=" << rtp_header.header.timestamp
+                  << ", sn=" << rtp_header.header.sequenceNumber
+                  << ", pt=" << static_cast<int>(rtp_header.header.payloadType)
+                  << ", ssrc=" << rtp_header.header.ssrc
+                  << ", len=" << payload.size();
+  int error =
+      InsertPacketInternal(rtp_header, payload, receive_timestamp, false);
   if (error != 0) {
     error_code_ = error;
     return kFail;
   }
   return kOK;
 }
 
 int NetEqImpl::InsertSyncPacket(const WebRtcRTPHeader& rtp_header,
                                 uint32_t receive_timestamp) {
   CriticalSectionScoped lock(crit_sect_.get());
   LOG(LS_VERBOSE) << "InsertPacket-Sync: ts="
       << rtp_header.header.timestamp <<
       ", sn=" << rtp_header.header.sequenceNumber <<
       ", pt=" << static_cast<int>(rtp_header.header.payloadType) <<
       ", ssrc=" << rtp_header.header.ssrc;
 
   const uint8_t kSyncPayload[] = { 's', 'y', 'n', 'c' };
-  int error = InsertPacketInternal(
-      rtp_header, kSyncPayload, sizeof(kSyncPayload), receive_timestamp, true);
+  int error =
+      InsertPacketInternal(rtp_header, kSyncPayload, receive_timestamp, true);
 
   if (error != 0) {
     error_code_ = error;
     return kFail;
   }
   return kOK;
 }
 
 int NetEqImpl::GetAudio(size_t max_length, int16_t* output_audio,
                         size_t* samples_per_channel, int* num_channels,
@@ skipping 274 matching lines @@
 }
 
 const SyncBuffer* NetEqImpl::sync_buffer_for_test() const {
   CriticalSectionScoped lock(crit_sect_.get());
   return sync_buffer_.get();
 }
 
 // Methods below this line are private.
 
 int NetEqImpl::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) {
-  if (!payload) {
+  if (payload.empty()) {
     LOG_F(LS_ERROR) << "payload == NULL";

[hlundin-webrtc, 2015/11/10 10:14:35]

"payload == NULL" is not as correct as it used to be.

[kwiberg-webrtc, 2015/11/10 10:27:59]

Done.

     return kInvalidPointer;
   }
   // Sanity checks for sync-packets.
   if (is_sync_packet) {
     if (decoder_database_->IsDtmf(rtp_header.header.payloadType) ||
         decoder_database_->IsRed(rtp_header.header.payloadType) ||
         decoder_database_->IsComfortNoise(rtp_header.header.payloadType)) {
       LOG_F(LS_ERROR) << "Sync-packet with an unacceptable payload type "
                       << static_cast<int>(rtp_header.header.payloadType);
       return kSyncPacketNotAccepted;
@@ skipping 15 matching lines @@
     // Create |packet| within this separate scope, since it should not be used
     // directly once it's been inserted in the packet list. This way, |packet|
     // is not defined outside of this block.
     Packet* packet = new Packet;
     packet->header.markerBit = false;
     packet->header.payloadType = rtp_header.header.payloadType;
     packet->header.sequenceNumber = rtp_header.header.sequenceNumber;
     packet->header.timestamp = rtp_header.header.timestamp;
     packet->header.ssrc = rtp_header.header.ssrc;
     packet->header.numCSRCs = 0;
-    packet->payload_length = length_bytes;
+    packet->payload_length = payload.size();
     packet->primary = true;
     packet->waiting_time = 0;
     packet->payload = new uint8_t[packet->payload_length];
     packet->sync_packet = is_sync_packet;
     if (!packet->payload) {
       LOG_F(LS_ERROR) << "Payload pointer is NULL.";
     }
-    assert(payload);  // Already checked above.
-    memcpy(packet->payload, payload, packet->payload_length);
+    assert(!payload.empty());  // Already checked above.
+    memcpy(packet->payload, payload.data(), packet->payload_length);
     // Insert packet in a packet list.
     packet_list.push_back(packet);
     // Save main payloads header for later.
     memcpy(&main_header, &packet->header, sizeof(main_header));
   }
 
   bool update_sample_rate_and_channels = false;
   // Reinitialize NetEq if it's needed (changed SSRC or first call).
   if ((main_header.ssrc != ssrc_) || first_packet_) {
     // Note: |first_packet_| will be cleared further down in this method, once
@@ skipping 1560 matching lines @@
 
 void NetEqImpl::CreateDecisionLogic() {
   decision_logic_.reset(DecisionLogic::Create(fs_hz_, output_size_samples_,
                                               playout_mode_,
                                               decoder_database_.get(),
                                               *packet_buffer_.get(),
                                               delay_manager_.get(),
                                               buffer_level_filter_.get()));
 }
 }  // namespace webrtc