Opus implementation of the AudioEncoderFactoryTemplate API

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/api/audio_codecs/audio_encoder.h"
 #include "webrtc/api/audio_codecs/audio_format.h"
+#include "webrtc/api/audio_codecs/opus/audio_encoder_opus_config.h"
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/base/optional.h"
 #include "webrtc/base/protobuf_utils.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"
 
 namespace webrtc {
 
 class RtcEventLog;
 
 struct CodecInst;
 
 class AudioEncoderOpus final : public AudioEncoder {
  public:
-  enum ApplicationMode {
-    kVoip = 0,
-    kAudio = 1,
-  };
+  static void AppendSupportedEncoders(std::vector<AudioCodecSpec>* specs);
+  static AudioCodecInfo QueryAudioEncoder(const AudioEncoderOpusConfig& config);
+  static std::unique_ptr<AudioEncoder> MakeAudioEncoder(
+      const AudioEncoderOpusConfig&,
+      int payload_type);
 
-  struct Config {
-    Config();
-    Config(const Config&);
-    ~Config();
-    Config& operator=(const Config&);
+  // NOTE: This alias will soon go away. See
+  // https://bugs.chromium.org/p/webrtc/issues/detail?id=7847
+  using Config = AudioEncoderOpusConfig;
 
-    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;
+  // NOTE: This function will soon go away. See
+  // https://bugs.chromium.org/p/webrtc/issues/detail?id=7847
+  static Config CreateConfig(int payload_type, const SdpAudioFormat& format);
 
-    static constexpr int kDefaultFrameSizeMs = 20;
-    int frame_size_ms = kDefaultFrameSizeMs;
-    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;
-    bool cbr_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;
-    int uplink_bandwidth_update_interval_ms = 200;
+  static AudioEncoderOpusConfig CreateConfig(const CodecInst& codec_inst);
+  static rtc::Optional<AudioEncoderOpusConfig> SdpToConfig(
+      const SdpAudioFormat& format);
 
-   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
-  };
-
-  static Config CreateConfig(int payload_type, const SdpAudioFormat& format);
-  static Config CreateConfig(const CodecInst& codec_inst);
+  // 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.
+  static rtc::Optional<int> GetNewComplexity(
+      const AudioEncoderOpusConfig& config);
 
   using AudioNetworkAdaptorCreator =
       std::function<std::unique_ptr<AudioNetworkAdaptor>(const std::string&,
                                                          RtcEventLog*)>;
+
+  // NOTE: This constructor will soon go away. See
+  // https://bugs.chromium.org/p/webrtc/issues/detail?id=7847
+  AudioEncoderOpus(const AudioEncoderOpusConfig& config);
+
   AudioEncoderOpus(
-      const Config& config,
+      const AudioEncoderOpusConfig& config,
+      int payload_type,
       AudioNetworkAdaptorCreator&& audio_network_adaptor_creator = nullptr,
       std::unique_ptr<SmoothingFilter> bitrate_smoother = nullptr);
 
   explicit AudioEncoderOpus(const CodecInst& codec_inst);
   AudioEncoderOpus(int payload_type, const SdpAudioFormat& format);
   ~AudioEncoderOpus() override;
 
   // Static interface for use by BuiltinAudioEncoderFactory.
   static constexpr const char* GetPayloadName() { return "opus"; }
   static rtc::Optional<AudioCodecInfo> QueryAudioEncoder(
       const SdpAudioFormat& format);
 
   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.
+  // 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,
                                  RtcEventLog* event_log) override;
   void DisableAudioNetworkAdaptor() override;
   void OnReceivedUplinkPacketLossFraction(
       float uplink_packet_loss_fraction) override;
   void OnReceivedUplinkRecoverablePacketLossFraction(
       float uplink_recoverable_packet_loss_fraction) override;
   void OnReceivedUplinkBandwidth(
       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_; }
-  ApplicationMode application() const { return config_.application; }
+  AudioEncoderOpusConfig::ApplicationMode application() const {
+    return config_.application;
+  }
   bool fec_enabled() const { return config_.fec_enabled; }
   size_t num_channels_to_encode() const { return num_channels_to_encode_; }
   int next_frame_length_ms() const { return next_frame_length_ms_; }
 
  protected:
   EncodedInfo EncodeImpl(uint32_t rtp_timestamp,
                          rtc::ArrayView<const int16_t> audio,
                          rtc::Buffer* encoded) override;
 
  private:
   class PacketLossFractionSmoother;
 
   size_t Num10msFramesPerPacket() const;
   size_t SamplesPer10msFrame() const;
   size_t SufficientOutputBufferSize() const;
-  bool RecreateEncoderInstance(const Config& config);
+  bool RecreateEncoderInstance(const AudioEncoderOpusConfig& config);
   void SetFrameLength(int frame_length_ms);
   void SetNumChannelsToEncode(size_t num_channels_to_encode);
   void SetProjectedPacketLossRate(float fraction);
 
   // TODO(minyue): remove "override" when we can deprecate
   // |AudioEncoder::SetTargetBitrate|.
   void SetTargetBitrate(int target_bps) override;
 
   void ApplyAudioNetworkAdaptor();
   std::unique_ptr<AudioNetworkAdaptor> DefaultAudioNetworkAdaptorCreator(
       const ProtoString& config_string,
       RtcEventLog* event_log) const;
 
   void MaybeUpdateUplinkBandwidth();
 
-  Config config_;
+  AudioEncoderOpusConfig config_;
+  const int payload_type_;
   const bool send_side_bwe_with_overhead_;
   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> bitrate_smoother_last_update_time_;
 
   RTC_DISALLOW_COPY_AND_ASSIGN(AudioEncoderOpus);
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_AUDIO_ENCODER_OPUS_H_