Roll chromium_revision 33a7a547b9..0e44c5e141 (452838:453130)

webrtc/modules/audio_coding/neteq/packet_buffer.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 25 matching lines @@
 
  private:
   const Packet& new_packet_;
 };
 
 // Returns true if both payload types are known to the decoder database, and
 // have the same sample rate.
 bool EqualSampleRates(uint8_t pt1,
                       uint8_t pt2,
                       const DecoderDatabase& decoder_database) {
-  auto di1 = decoder_database.GetDecoderInfo(pt1);
-  auto di2 = decoder_database.GetDecoderInfo(pt2);
+  auto* di1 = decoder_database.GetDecoderInfo(pt1);
+  auto* di2 = decoder_database.GetDecoderInfo(pt2);
   return di1 && di2 && di1->SampleRateHz() == di2->SampleRateHz();
 }
 }  // namespace
 
 PacketBuffer::PacketBuffer(size_t max_number_of_packets,
                            const TickTimer* tick_timer)
     : max_number_of_packets_(max_number_of_packets), tick_timer_(tick_timer) {}
 
 // Destructor. All packets in the buffer will be destroyed.
 PacketBuffer::~PacketBuffer() {
@@ skipping 208 matching lines @@
   }
   return num_samples;
 }
 
 void PacketBuffer::BufferStat(int* num_packets, int* max_num_packets) const {
   *num_packets = static_cast<int>(buffer_.size());
   *max_num_packets = static_cast<int>(max_number_of_packets_);
 }
 
 }  // namespace webrtc