Remove VoECodec from FakeWebRtcVoiceEngine.

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 <stddef.h>
-
-#include <list>
 #include <map>
 #include <vector>
 
 #include "webrtc/base/checks.h"
-#include "webrtc/base/stringutils.h"
-#include "webrtc/base/checks.h"
-#include "webrtc/config.h"
-#include "webrtc/media/base/codec.h"
-#include "webrtc/media/base/rtputils.h"
 #include "webrtc/media/engine/webrtcvoe.h"
-#include "webrtc/modules/audio_coding/acm2/rent_a_codec.h"
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
 
 namespace webrtc {
 namespace voe {
 class TransmitMixer;
 }  // namespace voe
 }  // namespace webrtc
 
 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;
-
 #define WEBRTC_CHECK_CHANNEL(channel) \
   if (channels_.find(channel) == channels_.end()) return -1;
 
 #define WEBRTC_STUB(method, args) \
   int method args override { return 0; }
 
-#define WEBRTC_STUB_CONST(method, args) \
-  int method args const override { return 0; }
-
-#define WEBRTC_BOOL_STUB(method, args) \
-  bool method args override { return true; }
-
-#define WEBRTC_VOID_STUB(method, args) \
-  void method args override {}
-
 #define WEBRTC_FUNC(method, args) int method args override
 
-class FakeWebRtcVoiceEngine
-    : public webrtc::VoEBase, public webrtc::VoECodec {
+class FakeWebRtcVoiceEngine : public webrtc::VoEBase {
  public:
   struct Channel {
     std::vector<webrtc::CodecInst> recv_codecs;
     size_t neteq_capacity = 0;
     bool neteq_fast_accelerate = false;
   };
 
   explicit FakeWebRtcVoiceEngine(webrtc::AudioProcessing* apm,
                                  webrtc::voe::TransmitMixer* transmit_mixer)
       : apm_(apm), transmit_mixer_(transmit_mixer) {
@@ skipping 60 matching lines @@
   WEBRTC_STUB(StartPlayout, (int channel));
   WEBRTC_STUB(StartSend, (int channel));
   WEBRTC_STUB(StopReceive, (int channel));
   WEBRTC_STUB(StopPlayout, (int channel));
   WEBRTC_STUB(StopSend, (int channel));
   WEBRTC_STUB(GetVersion, (char version[1024]));
   WEBRTC_STUB(LastError, ());
   WEBRTC_STUB(AssociateSendChannel, (int channel,
                                      int accociate_send_channel));
 
-  // webrtc::VoECodec
-  WEBRTC_STUB(NumOfCodecs, ());
-  WEBRTC_STUB(GetCodec, (int index, webrtc::CodecInst& codec));
-  WEBRTC_STUB(SetSendCodec, (int channel, const webrtc::CodecInst& codec));
-  WEBRTC_STUB(GetSendCodec, (int channel, webrtc::CodecInst& codec));
-  WEBRTC_STUB(SetBitRate, (int channel, int bitrate_bps));
-  WEBRTC_STUB(GetRecCodec, (int channel, webrtc::CodecInst& codec));
-  WEBRTC_FUNC(SetRecPayloadType, (int channel,
-                                  const webrtc::CodecInst& codec)) {
-    WEBRTC_CHECK_CHANNEL(channel);
-    Channel* ch = channels_[channel];
-    // Check if something else already has this slot.
-    if (codec.pltype != -1) {
-      for (std::vector<webrtc::CodecInst>::iterator it =
-          ch->recv_codecs.begin(); it != ch->recv_codecs.end(); ++it) {
-        if (it->pltype == codec.pltype &&
-            _stricmp(it->plname, codec.plname) != 0) {
-          return -1;
-        }
-      }
-    }
-    // Otherwise try to find this codec and update its payload type.
-    int result = -1;  // not found
-    for (std::vector<webrtc::CodecInst>::iterator it = ch->recv_codecs.begin();
-         it != ch->recv_codecs.end(); ++it) {
-      if (strcmp(it->plname, codec.plname) == 0 &&
-          it->plfreq == codec.plfreq &&
-          it->channels == codec.channels) {
-        it->pltype = codec.pltype;
-        result = 0;
-      }
-    }
-    return result;
-  }
-  WEBRTC_STUB(SetSendCNPayloadType, (int channel, int type,
-                                     webrtc::PayloadFrequencies frequency));
-  WEBRTC_FUNC(GetRecPayloadType, (int channel, webrtc::CodecInst& codec)) {
-    WEBRTC_CHECK_CHANNEL(channel);
-    Channel* ch = channels_[channel];
-    for (std::vector<webrtc::CodecInst>::iterator it = ch->recv_codecs.begin();
-         it != ch->recv_codecs.end(); ++it) {
-      if (strcmp(it->plname, codec.plname) == 0 &&
-          it->plfreq == codec.plfreq &&
-          it->channels == codec.channels &&
-          it->pltype != -1) {
-        codec.pltype = it->pltype;
-        return 0;
-      }
-    }
-    return -1;  // not found
-  }
-  WEBRTC_STUB(SetVADStatus, (int channel, bool enable, webrtc::VadModes mode,
-                             bool disableDTX));
-  WEBRTC_STUB(GetVADStatus, (int channel, bool& enabled,
-                             webrtc::VadModes& mode, bool& disabledDTX));
-  WEBRTC_STUB(SetFECStatus, (int channel, bool enable));
-  WEBRTC_STUB(GetFECStatus, (int channel, bool& enable));
-  WEBRTC_STUB(SetOpusMaxPlaybackRate, (int channel, int frequency_hz));
-  WEBRTC_STUB(SetOpusDtx, (int channel, bool enable_dtx));
-
   size_t GetNetEqCapacity() const {
     auto ch = channels_.find(last_channel_);
     RTC_DCHECK(ch != channels_.end());
     return ch->second->neteq_capacity;
   }
   bool GetNetEqFastAccelerate() const {
     auto ch = channels_.find(last_channel_);
     RTC_CHECK(ch != channels_.end());
     return ch->second->neteq_fast_accelerate;
   }
 
  private:
   bool inited_ = false;
   int last_channel_ = -1;
   std::map<int, Channel*> channels_;
   bool fail_create_channel_ = false;
   webrtc::AudioProcessing* apm_ = nullptr;
   webrtc::voe::TransmitMixer* transmit_mixer_ = nullptr;
 
   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(FakeWebRtcVoiceEngine);
 };
 
 }  // namespace cricket
 
 #endif  // WEBRTC_MEDIA_ENGINE_FAKEWEBRTCVOICEENGINE_H_