Injectable audio encoders: Moved audio encoder, factory and builtin factory to api/.

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/modules/audio_coding/codecs/audio_encoder.h"
+#include "webrtc/api/audio_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(EncodedInfo&&) = default;
 AudioEncoder::EncodedInfo::~EncodedInfo() = default;
@@ skipping 34 matching lines @@
 
 bool AudioEncoder::SetApplication(Application application) {
   return false;
 }
 
 void AudioEncoder::SetMaxPlaybackRate(int frequency_hz) {}
 
 void AudioEncoder::SetTargetBitrate(int target_bps) {}
 
 rtc::ArrayView<std::unique_ptr<AudioEncoder>>
-AudioEncoder::ReclaimContainedEncoders() { return nullptr; }
+AudioEncoder::ReclaimContainedEncoders() {
+  return nullptr;
+}
 
 bool AudioEncoder::EnableAudioNetworkAdaptor(const std::string& config_string,
                                              RtcEventLog* event_log,
                                              const Clock* clock) {
   return false;
 }
 
 void AudioEncoder::DisableAudioNetworkAdaptor() {}
 
 void AudioEncoder::OnReceivedUplinkPacketLossFraction(
@@ skipping 11 matching lines @@
     rtc::Optional<int64_t> probing_interval_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) {}
 
 }  // namespace webrtc