Fold AudioEncoderMutable into AudioEncoder

webrtc/modules/audio_coding/codecs/opus/interface/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_INTERFACE_AUDIO_ENCODER_OPUS_H_
 #define WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_INTERFACE_AUDIO_ENCODER_OPUS_H_
 
 #include <vector>
 
+#include "webrtc/base/checks.h"
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/audio_coding/codecs/audio_encoder_mutable_impl.h"
 #include "webrtc/modules/audio_coding/codecs/opus/interface/opus_interface.h"
 #include "webrtc/modules/audio_coding/codecs/audio_encoder.h"
 
 namespace webrtc {
 
-// NOTE: This class has neither ThreadChecker, nor locks. The owner of an
-// AudioEncoderOpus object must ensure that it is not accessed concurrently.
+struct CodecInst;
 
 class AudioEncoderOpus final : public AudioEncoder {
  public:
   enum ApplicationMode {
     kVoip = 0,
     kAudio = 1,
   };
 
   struct Config {
-    Config();
     bool IsOk() const;
-    int frame_size_ms;
-    int num_channels;
-    int payload_type;
-    ApplicationMode application;
-    int bitrate_bps;
-    bool fec_enabled;
-    int max_playback_rate_hz;
-    int complexity;
-    bool dtx_enabled;
+    int frame_size_ms = 20;
+    int num_channels = 1;
+    int payload_type = 120;
+    ApplicationMode application = kVoip;
+    int bitrate_bps = 64000;
+    bool fec_enabled = false;
+    int max_playback_rate_hz = 48000;
+    int complexity = kDefaultComplexity;
+    bool dtx_enabled = false;
+
+   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
   };
 
   explicit AudioEncoderOpus(const Config& config);
+  explicit AudioEncoderOpus(const CodecInst& codec_inst);
   ~AudioEncoderOpus() override;
 
+  size_t MaxEncodedBytes() const override;
   int SampleRateHz() const override;
   int NumChannels() const override;
-  size_t MaxEncodedBytes() const override;
   size_t Num10MsFramesInNextPacket() const override;
   size_t Max10MsFramesInAPacket() const override;
   int GetTargetBitrate() const override;
-  void SetTargetBitrate(int bits_per_second) override;
-  void SetProjectedPacketLossRate(double fraction) override;
-
-  double packet_loss_rate() const { return packet_loss_rate_; }
-  ApplicationMode application() const { return application_; }
-  bool dtx_enabled() const { return dtx_enabled_; }
 
   EncodedInfo EncodeInternal(uint32_t rtp_timestamp,
                              const int16_t* audio,
                              size_t max_encoded_bytes,
                              uint8_t* encoded) override;
 
- private:
-  const size_t num_10ms_frames_per_packet_;
-  const int num_channels_;
-  const int payload_type_;
-  const ApplicationMode application_;
-  int bitrate_bps_;
-  const bool dtx_enabled_;
-  const size_t samples_per_10ms_frame_;
-  std::vector<int16_t> input_buffer_;
-  OpusEncInst* inst_;
-  uint32_t first_timestamp_in_buffer_;
-  double packet_loss_rate_;
-};
-
-struct CodecInst;
-
-class AudioEncoderMutableOpus
-    : public AudioEncoderMutableImpl<AudioEncoderOpus> {
- public:
-  explicit AudioEncoderMutableOpus(const CodecInst& codec_inst);
+  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 SetApplication(Application application) override;
   bool SetMaxPlaybackRate(int frequency_hz) override;
-  AudioEncoderOpus::ApplicationMode application() const {
-    CriticalSectionScoped cs(encoder_lock_.get());
-    return encoder()->application();
-  }
-  double packet_loss_rate() const {
-    CriticalSectionScoped cs(encoder_lock_.get());
-    return encoder()->packet_loss_rate();
-  }
-  bool dtx_enabled() const {
-    CriticalSectionScoped cs(encoder_lock_.get());
-    return encoder()->dtx_enabled();
-  }
+  void SetProjectedPacketLossRate(double fraction) override;
+  void SetTargetBitrate(int target_bps) override;
+
+  // Getters for testing.
+  double packet_loss_rate() const { return packet_loss_rate_; }
+  ApplicationMode application() const { return config_.application; }
+  bool dtx_enabled() const { return config_.dtx_enabled; }
+
+ private:
+  int Num10msFramesPerPacket() const;
+  int SamplesPer10msFrame() const;
+  bool RecreateEncoderInstance(const Config& config);
+
+  Config config_;
+  double packet_loss_rate_;
+  std::vector<int16_t> input_buffer_;
+  OpusEncInst* inst_;
+  uint32_t first_timestamp_in_buffer_;
 };
 
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_INTERFACE_AUDIO_ENCODER_OPUS_H_