Add mock AudioDeviceModule.

webrtc/media/engine/fakewebrtcvoiceengine.h

 /*
  *  Copyright (c) 2010 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_MEDIA_ENGINE_FAKEWEBRTCVOICEENGINE_H_
 #define WEBRTC_MEDIA_ENGINE_FAKEWEBRTCVOICEENGINE_H_
 
 #include <list>
 #include <map>
 #include <vector>
 
 #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/media/base/codec.h"
 #include "webrtc/media/base/rtputils.h"
 #include "webrtc/media/engine/fakewebrtccommon.h"
 #include "webrtc/media/engine/webrtcvoe.h"
 #include "webrtc/modules/audio_coding/acm2/rent_a_codec.h"
-#include "webrtc/modules/audio_device/include/fake_audio_device.h"
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
 
 namespace cricket {
 
 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;
 
@@ skipping 73 matching lines @@
   webrtc::VoiceDetection* voice_detection() const override { return NULL; }
 
   bool experimental_ns_enabled() {
     return experimental_ns_enabled_;
   }
 
  private:
   bool experimental_ns_enabled_;
 };
 
-// TODO(solenberg): Swap this for a proper mock of the ADM.
-class FakeAudioDeviceModule : public webrtc::FakeAudioDeviceModule {
- public:
-  ~FakeAudioDeviceModule() override {
-    RTC_DCHECK_EQ(0, ref_count_);
-  }
-  int32_t AddRef() const override {
-    ref_count_++;
-    return ref_count_;
-  }
-  int32_t Release() const override {
-    RTC_DCHECK_LT(0, ref_count_);
-    ref_count_--;
-    return ref_count_;
-  }
-
- private:
-  mutable int32_t ref_count_ = 0;
-};
-
 class FakeWebRtcVoiceEngine
     : public webrtc::VoEAudioProcessing,
       public webrtc::VoEBase, public webrtc::VoECodec,
       public webrtc::VoEHardware,
       public webrtc::VoENetwork, public webrtc::VoERTP_RTCP,
       public webrtc::VoEVolumeControl {
  public:
   struct Channel {
-    explicit Channel()
-        : external_transport(false),
-          playout(false),
-          volume_scale(1.0),
-          vad(false),
-          codec_fec(false),
-          max_encoding_bandwidth(0),
-          opus_dtx(false),
-          red(false),
-          nack(false),
-          cn8_type(13),
-          cn16_type(105),
-          red_type(117),
-          nack_max_packets(0),
-          send_ssrc(0),
-          associate_send_channel(-1),
-          recv_codecs(),
-          neteq_capacity(-1),
-          neteq_fast_accelerate(false) {
+    Channel() {
       memset(&send_codec, 0, sizeof(send_codec));
     }
-    bool external_transport;
-    bool playout;
-    float volume_scale;
-    bool vad;
-    bool codec_fec;
-    int max_encoding_bandwidth;
-    bool opus_dtx;
-    bool red;
-    bool nack;
-    int cn8_type;
-    int cn16_type;
-    int red_type;
-    int nack_max_packets;
-    uint32_t send_ssrc;
-    int associate_send_channel;
+    bool external_transport = false;
+    bool playout = false;
+    float volume_scale = 1.0f;
+    bool vad = false;
+    bool codec_fec = false;
+    int max_encoding_bandwidth = 0;
+    bool opus_dtx = false;
+    bool red = false;
+    bool nack = false;
+    int cn8_type = 13;
+    int cn16_type = 105;
+    int red_type = 117;
+    int nack_max_packets = 0;
+    uint32_t send_ssrc = 0;
+    int associate_send_channel = -1;
     std::vector<webrtc::CodecInst> recv_codecs;
     webrtc::CodecInst send_codec;
     webrtc::PacketTime last_rtp_packet_time;
     std::list<std::string> packets;
-    int neteq_capacity;
-    bool neteq_fast_accelerate;
+    int neteq_capacity = -1;
+    bool neteq_fast_accelerate = false;
   };
 
-  FakeWebRtcVoiceEngine()
-      : inited_(false),
-        last_channel_(-1),
-        fail_create_channel_(false),
-        num_set_send_codecs_(0),
-        ec_enabled_(false),
-        ec_metrics_enabled_(false),
-        cng_enabled_(false),
-        ns_enabled_(false),
-        agc_enabled_(false),
-        highpass_filter_enabled_(false),
-        stereo_swapping_enabled_(false),
-        typing_detection_enabled_(false),
-        ec_mode_(webrtc::kEcDefault),
-        aecm_mode_(webrtc::kAecmSpeakerphone),
-        ns_mode_(webrtc::kNsDefault),
-        agc_mode_(webrtc::kAgcDefault),
-        observer_(NULL),
-        playout_fail_channel_(-1),
-        recording_sample_rate_(-1),
-        playout_sample_rate_(-1) {
+  FakeWebRtcVoiceEngine() {
     memset(&agc_config_, 0, sizeof(agc_config_));
   }
   ~FakeWebRtcVoiceEngine() override {
     RTC_CHECK(channels_.empty());
   }
 
   bool ec_metrics_enabled() const { return ec_metrics_enabled_; }
 
   bool IsInited() const { return inited_; }
   int GetLastChannel() const { return last_channel_; }
@@ skipping 42 matching lines @@
     if (result) {
       std::string packet = channels_[channel]->packets.front();
       result = (packet == std::string(static_cast<const char*>(data), len));
       channels_[channel]->packets.pop_front();
     }
     return result;
   }
   bool CheckNoPacket(int channel) {
     return channels_[channel]->packets.empty();
   }
-  void TriggerCallbackOnError(int channel_num, int err_code) {
-    RTC_DCHECK(observer_ != NULL);
-    observer_->CallbackOnError(channel_num, err_code);
-  }
   void set_playout_fail_channel(int channel) {
     playout_fail_channel_ = channel;
   }
   void set_fail_create_channel(bool fail_create_channel) {
     fail_create_channel_ = fail_create_channel;
   }
   int AddChannel(const webrtc::Config& config) {
     if (fail_create_channel_) {
       return -1;
     }
@@ skipping 11 matching lines @@
 
   int GetNumSetSendCodecs() const { return num_set_send_codecs_; }
 
   int GetAssociateSendChannel(int channel) {
     return channels_[channel]->associate_send_channel;
   }
 
   WEBRTC_STUB(Release, ());
 
   // webrtc::VoEBase
-  WEBRTC_FUNC(RegisterVoiceEngineObserver, (
-      webrtc::VoiceEngineObserver& observer)) {
-    observer_ = &observer;
-    return 0;
-  }
+  WEBRTC_STUB(RegisterVoiceEngineObserver, (
+      webrtc::VoiceEngineObserver& observer));
   WEBRTC_STUB(DeRegisterVoiceEngineObserver, ());
   WEBRTC_FUNC(Init, (webrtc::AudioDeviceModule* adm,
                      webrtc::AudioProcessing* audioproc)) {
     inited_ = true;
     return 0;
   }
   WEBRTC_FUNC(Terminate, ()) {
     inited_ = false;
     return 0;
   }
   webrtc::AudioProcessing* audio_processing() override {
     return &audio_processing_;
   }
   webrtc::AudioDeviceModule* audio_device_module() override {
-    return &audio_device_module_;
+    return nullptr;
   }
   WEBRTC_FUNC(CreateChannel, ()) {
     webrtc::Config empty_config;
     return AddChannel(empty_config);
   }
   WEBRTC_FUNC(CreateChannel, (const webrtc::Config& config)) {
     return AddChannel(config);
   }
   WEBRTC_FUNC(DeleteChannel, (int channel)) {
     WEBRTC_CHECK_CHANNEL(channel);
@@ skipping 427 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:
-  bool inited_;
-  int last_channel_;
+  bool inited_ = false;
+  int last_channel_ = -1;
   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_;
+  bool fail_create_channel_ = false;
+  int num_set_send_codecs_ = 0;  // how many times we call SetSendCodec().
+  bool ec_enabled_ = false;
+  bool ec_metrics_enabled_ = false;
+  bool cng_enabled_ = false;
+  bool ns_enabled_ = false;
+  bool agc_enabled_ = false;
+  bool highpass_filter_enabled_ = false;
+  bool stereo_swapping_enabled_ = false;
+  bool typing_detection_enabled_ = false;
+  webrtc::EcModes ec_mode_ = webrtc::kEcDefault;
+  webrtc::AecmModes aecm_mode_ = webrtc::kAecmSpeakerphone;
+  webrtc::NsModes ns_mode_ = webrtc::kNsDefault;
+  webrtc::AgcModes agc_mode_ = webrtc::kAgcDefault;
   webrtc::AgcConfig agc_config_;
-  webrtc::VoiceEngineObserver* observer_;
-  int playout_fail_channel_;
-  int recording_sample_rate_;
-  int playout_sample_rate_;
+  int playout_fail_channel_ = -1;
+  int recording_sample_rate_ = -1;
+  int playout_sample_rate_ = -1;
   FakeAudioProcessing audio_processing_;
-  FakeAudioDeviceModule audio_device_module_;
 };
 
 }  // namespace cricket
 
 #endif  // WEBRTC_MEDIA_ENGINE_FAKEWEBRTCVOICEENGINE_H_