Add new ANA stats to GetStats() to count the number of actions taken by each controller.

webrtc/api/audio_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/api/audio_codecs/audio_encoder.h"
 
 #include "webrtc/rtc_base/checks.h"
 #include "webrtc/rtc_base/trace_event.h"
 
 namespace webrtc {
 
 AudioEncoder::EncodedInfo::EncodedInfo() = default;
 AudioEncoder::EncodedInfo::EncodedInfo(const EncodedInfo&) = default;
 AudioEncoder::EncodedInfo::EncodedInfo(EncodedInfo&&) = default;
 AudioEncoder::EncodedInfo::~EncodedInfo() = default;
 AudioEncoder::EncodedInfo& AudioEncoder::EncodedInfo::operator=(
     const EncodedInfo&) = default;
 AudioEncoder::EncodedInfo& AudioEncoder::EncodedInfo::operator=(EncodedInfo&&) =
     default;
 
+AudioEncoder::AudioEncoderStats::AudioEncoderStats() = default;
+AudioEncoder::AudioEncoderStats::~AudioEncoderStats() = default;
+AudioEncoder::AudioEncoderStats::AudioEncoderStats(const AudioEncoderStats&) =
+    default;
+
 int AudioEncoder::RtpTimestampRateHz() const {
   return SampleRateHz();
 }
 
 AudioEncoder::EncodedInfo AudioEncoder::Encode(
     uint32_t rtp_timestamp,
     rtc::ArrayView<const int16_t> audio,
     rtc::Buffer* encoded) {
   TRACE_EVENT0("webrtc", "AudioEncoder::Encode");
   RTC_CHECK_EQ(audio.size(),
@@ skipping 51 matching lines @@
     int target_audio_bitrate_bps,
     rtc::Optional<int64_t> bwe_period_ms) {}
 
 void AudioEncoder::OnReceivedRtt(int rtt_ms) {}
 
 void AudioEncoder::OnReceivedOverhead(size_t overhead_bytes_per_packet) {}
 
 void AudioEncoder::SetReceiverFrameLengthRange(int min_frame_length_ms,
                                                int max_frame_length_ms) {}
 
+AudioEncoder::AudioEncoderStats AudioEncoder::GetStats() const {
+  return AudioEncoder::AudioEncoderStats();
+}
+
 }  // namespace webrtc