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

webrtc/api/audio_codecs/BUILD.gn

 # 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.
 
 import("../../webrtc.gni")
 if (is_android) {
@@ skipping 20 matching lines @@
   ]
 }
 
 rtc_static_library("builtin_audio_decoder_factory") {
   sources = [
     "builtin_audio_decoder_factory.cc",
     "builtin_audio_decoder_factory.h",
   ]
   deps = [
     ":audio_codecs_api",
-    "../../modules/audio_coding:builtin_audio_decoder_factory_internal",
     "../../rtc_base:rtc_base_approved",
+    "L16:audio_decoder_L16",
+    "g711:audio_decoder_g711",
   ]
+  defines = []
+  if (rtc_include_ilbc) {
+    deps += [ "ilbc:audio_decoder_ilbc" ]
+    defines += [ "WEBRTC_USE_BUILTIN_ILBC=1" ]
+  } else {
+    defines += [ "WEBRTC_USE_BUILTIN_ILBC=0" ]
+  }
+  if (rtc_include_opus) {
+    deps += [ "opus:audio_decoder_opus" ]
+    defines += [ "WEBRTC_USE_BUILTIN_OPUS=1" ]
+  } else {
+    defines += [ "WEBRTC_USE_BUILTIN_OPUS=0" ]
+  }
+  if (build_with_mozilla) {
+    defines += [
+      "WEBRTC_USE_BUILTIN_G722=0",
+      "WEBRTC_USE_BUILTIN_ISAC_FIX=0",
+      "WEBRTC_USE_BUILTIN_ISAC_FLOAT=0",
+    ]
+  } else {
+    if (current_cpu == "arm") {
+      deps += [ "isac:audio_decoder_isac_fix" ]
+      defines += [
+        "WEBRTC_USE_BUILTIN_ISAC_FIX=1",
+        "WEBRTC_USE_BUILTIN_ISAC_FLOAT=0",
+      ]
+    } else {
+      deps += [ "isac:audio_decoder_isac_float" ]
+      defines += [
+        "WEBRTC_USE_BUILTIN_ISAC_FIX=0",
+        "WEBRTC_USE_BUILTIN_ISAC_FLOAT=1",
+      ]
+    }
+    deps += [ "g722:audio_decoder_g722" ]
+    defines += [ "WEBRTC_USE_BUILTIN_G722=1" ]
+  }
 }
 
 rtc_static_library("builtin_audio_encoder_factory") {
   sources = [
     "builtin_audio_encoder_factory.cc",
     "builtin_audio_encoder_factory.h",
   ]
   deps = [
     ":audio_codecs_api",
-    "../../modules/audio_coding:builtin_audio_encoder_factory_internal",
     "../../rtc_base:rtc_base_approved",
+    "L16:audio_encoder_L16",
+    "g711:audio_encoder_g711",
   ]
+  defines = []
+  if (rtc_include_ilbc) {
+    deps += [ "ilbc:audio_encoder_ilbc" ]
+    defines += [ "WEBRTC_USE_BUILTIN_ILBC=1" ]
+  } else {
+    defines += [ "WEBRTC_USE_BUILTIN_ILBC=0" ]
+  }
+  if (rtc_include_opus) {
+    deps += [ "opus:audio_encoder_opus" ]
+    defines += [ "WEBRTC_USE_BUILTIN_OPUS=1" ]
+  } else {
+    defines += [ "WEBRTC_USE_BUILTIN_OPUS=0" ]
+  }
+  if (build_with_mozilla) {
+    defines += [
+      "WEBRTC_USE_BUILTIN_G722=0",
+      "WEBRTC_USE_BUILTIN_ISAC_FIX=0",
+      "WEBRTC_USE_BUILTIN_ISAC_FLOAT=0",
+    ]
+  } else {
+    if (current_cpu == "arm") {
+      deps += [ "isac:audio_encoder_isac_fix" ]
+      defines += [
+        "WEBRTC_USE_BUILTIN_ISAC_FIX=1",
+        "WEBRTC_USE_BUILTIN_ISAC_FLOAT=0",
+      ]
+    } else {
+      deps += [ "isac:audio_encoder_isac_float" ]
+      defines += [
+        "WEBRTC_USE_BUILTIN_ISAC_FIX=0",
+        "WEBRTC_USE_BUILTIN_ISAC_FLOAT=1",
+      ]
+    }
+    deps += [ "g722:audio_encoder_g722" ]
+    defines += [ "WEBRTC_USE_BUILTIN_G722=1" ]
+  }
 }