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

webrtc/modules/audio_coding/codecs/audio_encoder.cc

Index: webrtc/modules/audio_coding/codecs/audio_encoder.cc
diff --git a/webrtc/modules/audio_coding/codecs/audio_encoder.cc b/webrtc/modules/audio_coding/codecs/audio_encoder.cc
deleted file mode 100644
index 92d1b253e0ce1353b4f32f2bf8305cfb7f9c936c..0000000000000000000000000000000000000000
--- a/webrtc/modules/audio_coding/codecs/audio_encoder.cc
+++ /dev/null
@@ -1,96 +0,0 @@
-/*
- *  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(EncodedInfo&&) = default;
-AudioEncoder::EncodedInfo::~EncodedInfo() = default;
-AudioEncoder::EncodedInfo& AudioEncoder::EncodedInfo::operator=(
-    const EncodedInfo&) = default;
-AudioEncoder::EncodedInfo& AudioEncoder::EncodedInfo::operator=(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) {
-  TRACE_EVENT0("webrtc", "AudioEncoder::Encode");
-  RTC_CHECK_EQ(audio.size(),
-               static_cast<size_t>(NumChannels() * SampleRateHz() / 100));
-
-  const size_t old_size = encoded->size();
-  EncodedInfo info = EncodeImpl(rtp_timestamp, audio, encoded);
-  RTC_CHECK_EQ(encoded->size() - old_size, info.encoded_bytes);
-  return info;
-}
-
-bool AudioEncoder::SetFec(bool enable) {
-  return !enable;
-}
-
-bool AudioEncoder::SetDtx(bool enable) {
-  return !enable;
-}
-
-bool AudioEncoder::GetDtx() const {
-  return false;
-}
-
-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; }
-
-bool AudioEncoder::EnableAudioNetworkAdaptor(const std::string& config_string,
-                                             RtcEventLog* event_log) {
-  return false;
-}
-
-void AudioEncoder::DisableAudioNetworkAdaptor() {}
-
-void AudioEncoder::OnReceivedUplinkPacketLossFraction(
-    float uplink_packet_loss_fraction) {}
-
-void AudioEncoder::OnReceivedUplinkRecoverablePacketLossFraction(
-    float uplink_recoverable_packet_loss_fraction) {}
-
-void AudioEncoder::OnReceivedTargetAudioBitrate(int target_audio_bitrate_bps) {
-  OnReceivedUplinkBandwidth(target_audio_bitrate_bps, rtc::Optional<int64_t>());
-}
-
-void AudioEncoder::OnReceivedUplinkBandwidth(
-    int target_audio_bitrate_bps,
-    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