Update smoothed bitrate.

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/common_audio/smoothing_filter.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:
@@ skipping 25 matching lines @@
     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;
+    int update_uplink_bandwidth_interval_ms = 200;
 
    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
     static const int kDefaultComplexity = 9;
 #endif
   };
 
   using AudioNetworkAdaptorCreator =
       std::function<std::unique_ptr<AudioNetworkAdaptor>(const std::string&,
                                                          const Clock*)>;
   AudioEncoderOpus(
       const Config& config,
-      AudioNetworkAdaptorCreator&& audio_network_adaptor_creator = nullptr);
+      AudioNetworkAdaptorCreator&& audio_network_adaptor_creator = nullptr,
+      std::unique_ptr<SmoothingFilter> bitrate_smoother = nullptr);
 
   explicit AudioEncoderOpus(const CodecInst& codec_inst);
 
   ~AudioEncoderOpus() override;
 
   int SampleRateHz() const override;
   size_t NumChannels() const override;
   size_t Num10MsFramesInNextPacket() const override;
   size_t Max10MsFramesInAPacket() const override;
   int GetTargetBitrate() const override;
 
   void Reset() override;
   bool SetFec(bool enable) override;
 
   // Set Opus DTX. Once enabled, Opus stops transmission, when it detects voice
   // being inactive. During that, it still sends 2 packets (one for content, one
   // for signaling) about every 400 ms.
   bool SetDtx(bool enable) override;
   bool GetDtx() const override;
 
   bool SetApplication(Application application) override;
   void SetMaxPlaybackRate(int frequency_hz) override;
   bool EnableAudioNetworkAdaptor(const std::string& config_string,
                                  const Clock* clock) override;
   void DisableAudioNetworkAdaptor() override;
-  void OnReceivedUplinkBandwidth(int uplink_bandwidth_bps) override;
   void OnReceivedUplinkPacketLossFraction(
       float uplink_packet_loss_fraction) override;
-  void OnReceivedTargetAudioBitrate(int target_audio_bitrate_bps) override;
+  void OnReceivedTargetAudioBitrate(
+      int target_audio_bitrate_bps,
+      rtc::Optional<int64_t> probing_interval_ms) override;
   void OnReceivedRtt(int rtt_ms) override;
   void OnReceivedOverhead(size_t overhead_bytes_per_packet) override;
   void SetReceiverFrameLengthRange(int min_frame_length_ms,
                                    int max_frame_length_ms) override;
   rtc::ArrayView<const int> supported_frame_lengths_ms() const {
     return config_.supported_frame_lengths_ms;
   }
 
   // Getters for testing.
   float packet_loss_rate() const { return packet_loss_rate_; }
@@ skipping 20 matching lines @@
 
   // TODO(minyue): remove "override" when we can deprecate
   // |AudioEncoder::SetTargetBitrate|.
   void SetTargetBitrate(int target_bps) override;
 
   void ApplyAudioNetworkAdaptor();
   std::unique_ptr<AudioNetworkAdaptor> DefaultAudioNetworkAdaptorCreator(
       const std::string& config_string,
       const Clock* clock) const;
 
+  void MayUpdateUplinkBandwidth();
+
   Config config_;
   float 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::Optional<size_t> overhead_bytes_per_packet_;
+  const std::unique_ptr<SmoothingFilter> bitrate_smoother_;
+  rtc::Optional<int64_t> last_smoothed_bandwith_update_;

[minyue-webrtc, 2016/12/22 14:51:26]

bitrate_smoother_last_update_time_

[michaelt, 2016/12/22 15:10:10]

Done.

 
   RTC_DISALLOW_COPY_AND_ASSIGN(AudioEncoderOpus);
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_AUDIO_ENCODER_OPUS_H_