Reimplement the builtin audio codec factories using the new stuff in api/

webrtc/api/audio_codecs/builtin_audio_encoder_factory.cc

 /*
  *  Copyright (c) 2017 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.
  */
 
 #include "webrtc/api/audio_codecs/builtin_audio_encoder_factory.h"
 
-#include "webrtc/modules/audio_coding/codecs/builtin_audio_encoder_factory_internal.h"
+#include <memory>
+#include <vector>
+
+#include "webrtc/api/audio_codecs/L16/audio_encoder_L16.h"
+#include "webrtc/api/audio_codecs/audio_encoder_factory_template.h"
+#include "webrtc/api/audio_codecs/g711/audio_encoder_g711.h"
+#if WEBRTC_USE_BUILTIN_G722
+#include "webrtc/api/audio_codecs/g722/audio_encoder_g722.h"  // nogncheck
+#endif
+#if WEBRTC_USE_BUILTIN_ILBC
+#include "webrtc/api/audio_codecs/ilbc/audio_encoder_ilbc.h"  // nogncheck
+#endif
+#if WEBRTC_USE_BUILTIN_ISAC_FIX
+#include "webrtc/api/audio_codecs/isac/audio_encoder_isac_fix.h"  // nogncheck
+#elif WEBRTC_USE_BUILTIN_ISAC_FLOAT
+#include "webrtc/api/audio_codecs/isac/audio_encoder_isac_float.h"  // nogncheck
+#endif
+#if WEBRTC_USE_BUILTIN_OPUS
+#include "webrtc/api/audio_codecs/opus/audio_encoder_opus.h"  // nogncheck
+#endif
 
 namespace webrtc {
 
+namespace {
+
+// Modify an audio encoder to not advertise support for anything.
+template <typename T>
+struct NotAdvertised {
+  using Config = typename T::Config;
+  static rtc::Optional<Config> SdpToConfig(const SdpAudioFormat& audio_format) {
+    return T::SdpToConfig(audio_format);
+  }
+  static void AppendSupportedEncoders(std::vector<AudioCodecSpec>* specs) {
+    // Don't advertise support for anything.
+  }
+  static AudioCodecInfo QueryAudioEncoder(const Config& config) {
+    return T::QueryAudioEncoder(config);
+  }
+  static std::unique_ptr<AudioEncoder> MakeAudioEncoder(const Config& config,
+                                                        int payload_type) {
+    return T::MakeAudioEncoder(config, payload_type);
+  }
+};
+
+}  // namespace
+
 rtc::scoped_refptr<AudioEncoderFactory> CreateBuiltinAudioEncoderFactory() {
-  return CreateBuiltinAudioEncoderFactoryInternal();
+  return CreateAudioEncoderFactory<
+
+#if WEBRTC_USE_BUILTIN_OPUS
+      AudioEncoderOpus,
+#endif
+
+#if WEBRTC_USE_BUILTIN_ISAC_FIX
+      AudioEncoderIsacFix,
+#elif WEBRTC_USE_BUILTIN_ISAC_FLOAT
+      AudioEncoderIsacFloat,
+#endif
+
+#if WEBRTC_USE_BUILTIN_G722
+      AudioEncoderG722,
+#endif
+
+#if WEBRTC_USE_BUILTIN_ILBC
+      AudioEncoderIlbc,
+#endif
+
+      AudioEncoderG711, NotAdvertised<AudioEncoderL16>>();
 }
 
 }  // namespace webrtc