Stop using old AudioCodingModule::RegisterReceiveCodec overloads

webrtc/modules/audio_coding/BUILD.gn

 # Copyright (c) 2014 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("../../build/webrtc.gni")
 import("audio_coding.gni")
@@ skipping 21 matching lines @@
 }
 if (!build_with_mozilla && !build_with_chromium) {
   audio_codec_deps += [ ":red" ]
 }
 audio_coding_deps = audio_codec_deps + [
                       "../..:webrtc_common",
                       "../../common_audio",
                       "../../system_wrappers",
                     ]
 
-rtc_static_library("audio_decoder_factory_interface") {
+rtc_static_library("audio_format") {
   sources = [
-    "codecs/audio_decoder_factory.h",
     "codecs/audio_format.cc",
     "codecs/audio_format.h",
   ]
   deps = [
-    "../..:webrtc_common",
+    "//webrtc:webrtc_common",
   ]
 }
 
+rtc_static_library("audio_format_conversion") {
+  sources = [
+    "codecs/audio_format_conversion.cc",
+    "codecs/audio_format_conversion.h",
+  ]
+  deps = [
+    ":audio_format",
+    "//webrtc:webrtc_common",
+    "//webrtc/base:rtc_base_approved",
+  ]
+}
+
+rtc_source_set("audio_decoder_factory_interface") {
+  sources = [
+    "codecs/audio_decoder_factory.h",
+  ]
+  deps = [
+    ":audio_decoder_interface",
+    ":audio_format",
+    "//webrtc/base:rtc_base_approved",
+  ]
+}
+
 rtc_static_library("builtin_audio_decoder_factory") {
   sources = [
     "codecs/builtin_audio_decoder_factory.cc",
     "codecs/builtin_audio_decoder_factory.h",
   ]
   deps = [
            "../..:webrtc_common",
            ":audio_decoder_factory_interface",
          ] + audio_codec_deps
   defines = audio_codec_defines
@@ skipping 864 matching lines @@
     testonly = true
     sources = [
       "test/Channel.cc",
       "test/PCMFile.cc",
       "test/delay_test.cc",
       "test/utility.cc",
     ]
 
     deps = [
       ":audio_coding",
+      ":audio_format_conversion",
       "../../:webrtc_common",
       "../../system_wrappers",
       "../../system_wrappers:system_wrappers_default",
       "../../test:test_support",
       "../rtp_rtcp",
       "//testing/gtest",
       "//third_party/gflags:gflags",
     ]
   }  # delay_test
 
   rtc_executable("insert_packet_with_timing") {
     testonly = true
     sources = [
       "test/Channel.cc",
       "test/PCMFile.cc",
       "test/insert_packet_with_timing.cc",
     ]
 
     if (!build_with_chromium && is_clang) {
       # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
       suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
     }
 
     deps = [
       ":audio_coding",
+      ":audio_format_conversion",
       "../../:webrtc_common",
       "../../system_wrappers",
       "../../system_wrappers:system_wrappers_default",
       "../../test:test_support",
       "../rtp_rtcp",
       "//testing/gtest",
       "//third_party/gflags:gflags",
     ]
   }  # insert_packet_with_timing
 
@@ skipping 640 matching lines @@
       "../../test:test_support_main",
       "//testing/gtest",
     ]
 
     if (!build_with_chromium && is_clang) {
       # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
       suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
     }
   }
 }