Revert of Opus implementation of the AudioEncoderFactoryTemplate API

webrtc/api/audio_codecs/opus/audio_encoder_opus_config.h

Index: webrtc/api/audio_codecs/opus/audio_encoder_opus_config.h
diff --git a/webrtc/api/audio_codecs/opus/audio_encoder_opus_config.h b/webrtc/api/audio_codecs/opus/audio_encoder_opus_config.h
deleted file mode 100644
index 1726c69d6e413863b7437c1d4b5d13b674ba1eb0..0000000000000000000000000000000000000000
--- a/webrtc/api/audio_codecs/opus/audio_encoder_opus_config.h
+++ /dev/null
@@ -1,63 +0,0 @@
-/*
- *  Copyright (c) 2017 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.
- */
-
-#ifndef WEBRTC_API_AUDIO_CODECS_OPUS_AUDIO_ENCODER_OPUS_CONFIG_H_
-#define WEBRTC_API_AUDIO_CODECS_OPUS_AUDIO_ENCODER_OPUS_CONFIG_H_
-
-#include <stddef.h>
-
-#include <vector>
-
-namespace webrtc {
-
-// NOTE: This struct is still under development and may change without notice.
-struct AudioEncoderOpusConfig {
-  static constexpr int kDefaultFrameSizeMs = 20;
-
-  // Opus API allows a min bitrate of 500bps, but Opus documentation suggests
-  // bitrate should be in the range of 6000 to 510000, inclusive.
-  static constexpr int kMinBitrateBps = 6000;
-  static constexpr int kMaxBitrateBps = 510000;
-
-  AudioEncoderOpusConfig();
-  AudioEncoderOpusConfig(const AudioEncoderOpusConfig&);
-  ~AudioEncoderOpusConfig();
-  AudioEncoderOpusConfig& operator=(const AudioEncoderOpusConfig&);
-
-  bool IsOk() const;  // Checks if the values are currently OK.
-
-  int frame_size_ms;
-  size_t num_channels;
-  enum class ApplicationMode { kVoip, kAudio };
-  ApplicationMode application;
-  int bitrate_bps;
-  bool fec_enabled;
-  bool cbr_enabled;
-  int max_playback_rate_hz;
-
-  // |complexity| is used when the bitrate goes above
-  // |complexity_threshold_bps| + |complexity_threshold_window_bps|;
-  // |low_rate_complexity| is used when the bitrate falls below
-  // |complexity_threshold_bps| - |complexity_threshold_window_bps|. In the
-  // interval in the middle, we keep using the most recent of the two
-  // complexity settings.
-  int complexity;
-  int low_rate_complexity;
-  int complexity_threshold_bps;
-  int complexity_threshold_window_bps;
-
-  bool dtx_enabled;
-  std::vector<int> supported_frame_lengths_ms;
-  int uplink_bandwidth_update_interval_ms;
-};
-
-}  // namespace webrtc
-
-#endif  // WEBRTC_API_AUDIO_CODECS_OPUS_AUDIO_ENCODER_OPUS_CONFIG_H_