Injectable audio encoders: BuiltinAudioEncoderFactory

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_format.h"
 #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 {
 
 class RtcEventLog;
@@ skipping 14 matching lines @@
     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;
+    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;
     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 = Clock::GetRealTimeClock();
     int uplink_bandwidth_update_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
   };
 
+  static Config CreateConfig(int payload_type, const SdpAudioFormat& format);
+  static Config CreateConfig(const CodecInst& codec_inst);
+
   using AudioNetworkAdaptorCreator =
       std::function<std::unique_ptr<AudioNetworkAdaptor>(const std::string&,
                                                          RtcEventLog*,
                                                          const Clock*)>;
   AudioEncoderOpus(
       const Config& config,
       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;
 
-  ~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;
 
@@ skipping 72 matching lines @@
   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_