Roll chromium_revision 212f976fef..61ed337cfe (387882:388120)

webrtc/modules/audio_coding/codecs/audio_encoder.cc

 /*
  *  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.
  */
 
 #include "webrtc/modules/audio_coding/codecs/audio_encoder.h"
 
 #include "webrtc/base/checks.h"
 #include "webrtc/base/trace_event.h"
 
 namespace webrtc {
 
 AudioEncoder::EncodedInfo::EncodedInfo() = default;
 
+AudioEncoder::EncodedInfo::EncodedInfo(const EncodedInfo&) = default;
+
 AudioEncoder::EncodedInfo::~EncodedInfo() = default;
 
 int AudioEncoder::RtpTimestampRateHz() const {
   return SampleRateHz();
 }
 
 AudioEncoder::EncodedInfo AudioEncoder::Encode(
     uint32_t rtp_timestamp,
     rtc::ArrayView<const int16_t> audio,
     rtc::Buffer* encoded) {
@@ skipping 24 matching lines @@
 void AudioEncoder::SetProjectedPacketLossRate(double fraction) {}
 
 void AudioEncoder::SetTargetBitrate(int target_bps) {}
 
 size_t AudioEncoder::MaxEncodedBytes() const {
   RTC_CHECK(false);
   return 0;
 }
 
 }  // namespace webrtc