AudioEncoderOpus: Default to 32 kbit/s for mono, 64 kbit/s for stereo

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 <vector>
 
 #include "webrtc/base/constructormagic.h"
+#include "webrtc/base/optional.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;
+
     int frame_size_ms = 20;
     size_t num_channels = 1;
     int payload_type = 120;
     ApplicationMode application = kVoip;
-    int bitrate_bps = 64000;
+    rtc::Optional<int> bitrate_bps;  // Unset means to use default value.
     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;
@@ skipping 45 matching lines @@
   double packet_loss_rate_;
   std::vector<int16_t> input_buffer_;
   OpusEncInst* inst_;
   uint32_t first_timestamp_in_buffer_;
   RTC_DISALLOW_COPY_AND_ASSIGN(AudioEncoderOpus);
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_AUDIO_ENCODER_OPUS_H_