Revert of WebRtcVoiceEngine: Use AudioDecoderFactory to generate recv codecs.

webrtc/media/engine/webrtcvoiceengine.cc

Index: webrtc/media/engine/webrtcvoiceengine.cc
diff --git a/webrtc/media/engine/webrtcvoiceengine.cc b/webrtc/media/engine/webrtcvoiceengine.cc
index 59a2870cf8d7e48737c73593a5fcdb8693afaf15..469098a32274973cb38189eb8094336920ae113c 100644
--- a/webrtc/media/engine/webrtcvoiceengine.cc
+++ b/webrtc/media/engine/webrtcvoiceengine.cc
@@ -14,7 +14,6 @@
 
 #include <algorithm>
 #include <cstdio>
-#include <functional>
 #include <string>
 #include <vector>
 
@@ -33,7 +32,6 @@
 #include "webrtc/media/base/audiosource.h"
 #include "webrtc/media/base/mediaconstants.h"
 #include "webrtc/media/base/streamparams.h"
-#include "webrtc/media/engine/payload_type_mapper.h"
 #include "webrtc/media/engine/webrtcmediaengine.h"
 #include "webrtc/media/engine/webrtcvoe.h"
 #include "webrtc/modules/audio_coding/acm2/rent_a_codec.h"
@@ -250,7 +248,7 @@
  public:
   // TODO(solenberg): Do this filtering once off-line, add a simple AudioCodec
   // list and add a test which verifies VoE supports the listed codecs.
-  static std::vector<AudioCodec> SupportedSendCodecs() {
+  static std::vector<AudioCodec> SupportedCodecs() {
     std::vector<AudioCodec> result;
     // Iterate first over our preferred codecs list, so that the results are
     // added in order of preference.
@@ -513,20 +511,13 @@
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   LOG(LS_INFO) << "WebRtcVoiceEngine::WebRtcVoiceEngine";
   RTC_DCHECK(voe_wrapper);
-  RTC_DCHECK(decoder_factory);
 
   signal_thread_checker_.DetachFromThread();
 
   // Load our audio codec list.
-  LOG(LS_INFO) << "Supported send codecs in order of preference:";
-  send_codecs_ = WebRtcVoiceCodecs::SupportedSendCodecs();
-  for (const AudioCodec& codec : send_codecs_) {
-    LOG(LS_INFO) << ToString(codec);
-  }
-
-  LOG(LS_INFO) << "Supported recv codecs in order of preference:";
-  recv_codecs_ = CollectRecvCodecs();
-  for (const AudioCodec& codec : recv_codecs_) {
+  LOG(LS_INFO) << "Supported codecs in order of preference:";
+  codecs_ = WebRtcVoiceCodecs::SupportedCodecs();
+  for (const AudioCodec& codec : codecs_) {
     LOG(LS_INFO) << ToString(codec);
   }
 
@@ -945,12 +936,12 @@
 
 const std::vector<AudioCodec>& WebRtcVoiceEngine::send_codecs() const {
   RTC_DCHECK(signal_thread_checker_.CalledOnValidThread());
-  return send_codecs_;
+  return codecs_;
 }
 
 const std::vector<AudioCodec>& WebRtcVoiceEngine::recv_codecs() const {
   RTC_DCHECK(signal_thread_checker_.CalledOnValidThread());
-  return recv_codecs_;
+  return codecs_;
 }
 
 RtpCapabilities WebRtcVoiceEngine::GetCapabilities() const {
@@ -1088,61 +1079,6 @@
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   RTC_DCHECK(adm_);
   return adm_;
-}
-
-AudioCodecs WebRtcVoiceEngine::CollectRecvCodecs() const {
-  PayloadTypeMapper mapper;
-  AudioCodecs out;
-  const std::vector<webrtc::SdpAudioFormat>& formats =
-      decoder_factory_->GetSupportedFormats();
-
-  // Only generate CN payload types for these clockrates
-  std::map<int, bool, std::greater<int>> generate_cn = {{ 8000,  false },
-                                                        { 16000, false },
-                                                        { 32000, false }};
-
-  auto map_format = [&mapper, &out] (const webrtc::SdpAudioFormat& format) {
-    rtc::Optional<AudioCodec> opt_codec = mapper.ToAudioCodec(format);
-    if (!opt_codec) {
-      LOG(LS_ERROR) << "Unable to assign payload type to format: " << format;
-      return false;
-    }
-
-    auto& codec = *opt_codec;
-    if (IsCodec(codec, kOpusCodecName)) {
-      // TODO(ossu): Set this specifically for Opus for now, until we have a
-      // better way of dealing with rtcp-fb parameters.
-      codec.AddFeedbackParam(
-          FeedbackParam(kRtcpFbParamTransportCc, kParamValueEmpty));
-    }
-    out.push_back(codec);
-    return true;
-  };
-
-  for (const auto& format : formats) {
-    if (map_format(format)) {
-      // TODO(ossu): We should get more than just a format from the factory, so
-      // we can determine if a format should be used with CN or not. For now,
-      // generate a CN entry for each supported clock rate also used by a format
-      // supported by the factory.
-      auto cn = generate_cn.find(format.clockrate_hz);
-      if (cn != generate_cn.end() /* && format.allow_comfort_noise */) {
-        cn->second = true;
-      }
-    }
-  }
-
-  // Add CN codecs after "proper" audio codecs
-  for (const auto& cn : generate_cn) {
-    if (cn.second) {
-      map_format({kCnCodecName, cn.first, 1});
-    }
-  }
-
-  // Add telephone-event codec last
-  map_format({kDtmfCodecName, 8000, 1});
-
-  return out;
 }
 
 class WebRtcVoiceMediaChannel::WebRtcAudioSendStream