Refactor handling of AudioOptions.

talk/media/webrtc/fakewebrtcvoiceengine.h

 /*
  * libjingle
  * Copyright 2010 Google Inc.
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions are met:
  *
  *  1. Redistributions of source code must retain the above copyright notice,
  *     this list of conditions and the following disclaimer.
  *  2. Redistributions in binary form must reproduce the above copyright notice,
@@ skipping 28 matching lines @@
 #include "webrtc/base/basictypes.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/base/gunit.h"
 #include "webrtc/base/stringutils.h"
 #include "webrtc/config.h"
 #include "webrtc/modules/audio_coding/acm2/rent_a_codec.h"
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
 
 namespace cricket {
 
-static const char kFakeDefaultDeviceName[] = "Fake Default";
-static const int kFakeDefaultDeviceId = -1;
-static const char kFakeDeviceName[] = "Fake Device";
-#ifdef WIN32
-static const int kFakeDeviceId = 0;
-#else
-static const int kFakeDeviceId = 1;
-#endif
-
 static const int kOpusBandwidthNb = 4000;
 static const int kOpusBandwidthMb = 6000;
 static const int kOpusBandwidthWb = 8000;
 static const int kOpusBandwidthSwb = 12000;
 static const int kOpusBandwidthFb = 20000;
 
 #define WEBRTC_CHECK_CHANNEL(channel) \
   if (channels_.find(channel) == channels_.end()) return -1;
 
 class FakeAudioProcessing : public webrtc::AudioProcessing {
@@ skipping 463 matching lines @@
     WEBRTC_CHECK_CHANNEL(channel);
     if (_stricmp(channels_[channel]->send_codec.plname, "opus") != 0) {
       // Return -1 if current send codec is not Opus.
       return -1;
     }
     channels_[channel]->opus_dtx = enable_dtx;
     return 0;
   }
 
   // webrtc::VoEHardware
-  WEBRTC_FUNC(GetNumOfRecordingDevices, (int& num)) {
-    return GetNumDevices(num);
-  }
-  WEBRTC_FUNC(GetNumOfPlayoutDevices, (int& num)) {
-    return GetNumDevices(num);
-  }
-  WEBRTC_FUNC(GetRecordingDeviceName, (int i, char* name, char* guid)) {
-    return GetDeviceName(i, name, guid);
-  }
-  WEBRTC_FUNC(GetPlayoutDeviceName, (int i, char* name, char* guid)) {
-    return GetDeviceName(i, name, guid);
-  }
+  WEBRTC_STUB(GetNumOfRecordingDevices, (int& num));
+  WEBRTC_STUB(GetNumOfPlayoutDevices, (int& num));
+  WEBRTC_STUB(GetRecordingDeviceName, (int i, char* name, char* guid));
+  WEBRTC_STUB(GetPlayoutDeviceName, (int i, char* name, char* guid));
   WEBRTC_STUB(SetRecordingDevice, (int, webrtc::StereoChannel));
   WEBRTC_STUB(SetPlayoutDevice, (int));
   WEBRTC_STUB(SetAudioDeviceLayer, (webrtc::AudioLayers));
   WEBRTC_STUB(GetAudioDeviceLayer, (webrtc::AudioLayers&));
   WEBRTC_FUNC(SetRecordingSampleRate, (unsigned int samples_per_sec)) {
     recording_sample_rate_ = samples_per_sec;
     return 0;
   }
   WEBRTC_FUNC_CONST(RecordingSampleRate, (unsigned int* samples_per_sec)) {
     *samples_per_sec = recording_sample_rate_;
@@ skipping 238 matching lines @@
     ASSERT(ch != channels_.end());
     return ch->second->neteq_capacity;
   }
   bool GetNetEqFastAccelerate() const {
     auto ch = channels_.find(last_channel_);
     ASSERT(ch != channels_.end());
     return ch->second->neteq_fast_accelerate;
   }
 
  private:
-  int GetNumDevices(int& num) {
-#ifdef WIN32
-    num = 1;
-#else
-    // On non-Windows platforms VE adds a special entry for the default device,
-    // so if there is one physical device then there are two entries in the
-    // list.
-    num = 2;
-#endif
-    return 0;
-  }
-
-  int GetDeviceName(int i, char* name, char* guid) {
-    const char *s;
-#ifdef WIN32
-    if (0 == i) {
-      s = kFakeDeviceName;
-    } else {
-      return -1;
-    }
-#else
-    // See comment above.
-    if (0 == i) {
-      s = kFakeDefaultDeviceName;
-    } else if (1 == i) {
-      s = kFakeDeviceName;
-    } else {
-      return -1;
-    }
-#endif
-    strcpy(name, s);
-    guid[0] = '\0';
-    return 0;
-  }
-
   bool inited_;
   int last_channel_;
   std::map<int, Channel*> channels_;
   bool fail_create_channel_;
   int num_set_send_codecs_;  // how many times we call SetSendCodec().
   bool ec_enabled_;
   bool ec_metrics_enabled_;
   bool cng_enabled_;
   bool ns_enabled_;
   bool agc_enabled_;
   bool highpass_filter_enabled_;
   bool stereo_swapping_enabled_;
   bool typing_detection_enabled_;
   webrtc::EcModes ec_mode_;
   webrtc::AecmModes aecm_mode_;
   webrtc::NsModes ns_mode_;
   webrtc::AgcModes agc_mode_;
   webrtc::AgcConfig agc_config_;
   webrtc::VoiceEngineObserver* observer_;
   int playout_fail_channel_;
   int send_fail_channel_;
   int recording_sample_rate_;
   int playout_sample_rate_;
   FakeAudioProcessing audio_processing_;
 };
 
 }  // namespace cricket
 
 #endif  // TALK_SESSION_PHONE_FAKEWEBRTCVOICEENGINE_H_