Let Opus increase complexity for low bitrates

webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h

 /*
  *  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.
  */
 
 #ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_AUDIO_ENCODER_OPUS_H_
 #define WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_AUDIO_ENCODER_OPUS_H_
 
 #include <functional>
+#include <memory>
+#include <string>
 #include <vector>
 
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/base/optional.h"
 #include "webrtc/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h"
 #include "webrtc/modules/audio_coding/codecs/opus/opus_interface.h"
 #include "webrtc/modules/audio_coding/codecs/audio_encoder.h"
 
 namespace webrtc {
 
 struct CodecInst;
 
 class AudioEncoderOpus final : public AudioEncoder {
  public:
   enum ApplicationMode {
     kVoip = 0,
     kAudio = 1,
   };
 
   struct Config {
     Config();
     Config(const Config&);
     ~Config();
     Config& operator=(const Config&);
 
     bool IsOk() const;
     int GetBitrateBps() const;
+    // Returns empty if the current bitrate falls within the hysteresis window,
+    // defined by complexity_threshold_bps +/- complexity_threshold_window_bps.
+    // Otherwise, returns the current complexity depending on whether the
+    // current bitrate is above or below complexity_threshold_bps.
+    rtc::Optional<int> GetNewComplexity() const;
 
     int frame_size_ms = 20;
     size_t num_channels = 1;
     int payload_type = 120;
     ApplicationMode application = kVoip;
     rtc::Optional<int> bitrate_bps;  // Unset means to use default value.
     bool fec_enabled = false;
     int max_playback_rate_hz = 48000;
     int complexity = kDefaultComplexity;
+    // This value may change in the struct's constructor.
+    int low_rate_complexity = kDefaultComplexity;
+    // low_rate_complexity is used when the bitrate is below this threshold.
+    int complexity_threshold_bps = 12500;
+    int complexity_threshold_window_bps = 1500;
     bool dtx_enabled = false;
     std::vector<int> supported_frame_lengths_ms;
     const Clock* clock = nullptr;
 
    private:
 #if defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS) || defined(WEBRTC_ARCH_ARM)
     // If we are on Android, iOS and/or ARM, use a lower complexity setting as
     // default, to save encoder complexity.
     static const int kDefaultComplexity = 5;
 #else
@@ skipping 72 matching lines @@
       const std::string& config_string,
       const Clock* clock) const;
 
   Config config_;
   double packet_loss_rate_;
   std::vector<int16_t> input_buffer_;
   OpusEncInst* inst_;
   uint32_t first_timestamp_in_buffer_;
   size_t num_channels_to_encode_;
   int next_frame_length_ms_;
+  int complexity_;
   std::unique_ptr<PacketLossFractionSmoother> packet_loss_fraction_smoother_;
   AudioNetworkAdaptorCreator audio_network_adaptor_creator_;
   std::unique_ptr<AudioNetworkAdaptor> audio_network_adaptor_;
 
   RTC_DISALLOW_COPY_AND_ASSIGN(AudioEncoderOpus);
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_AUDIO_ENCODER_OPUS_H_