audio_coding: remove "main" directory

This is the last piece of the old directory layout of the modules.

Duplicated header files are left in audio_coding/main/include until
downstream code is updated to the new location. They have pragma
warnings added to them and identical header guards as the new headers to avoid breaking things.

BUG=webrtc:5095
TESTED=Passing compile-trybots with --clobber flag:
git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel --bot=linux_gn_rel --bot=win_x64_gn_rel --bot=mac_x64_gn_rel --bot=android_gn_rel -m tryserver.webrtc
NOTRY=True
NOPRESUBMIT=True

Review URL: https://codereview.webrtc.org/1481493004

Cr-Commit-Position: refs/heads/master@{#10803}

webrtc/modules/audio_coding/acm2/audio_coding_module.cc

@@ -0,0 +1,98 @@
+/*
+ *  Copyright (c) 2012 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/modules/audio_coding/include/audio_coding_module.h"
+
+#include "webrtc/base/checks.h"
+#include "webrtc/common_types.h"
+#include "webrtc/modules/audio_coding/acm2/audio_coding_module_impl.h"
+#include "webrtc/modules/audio_coding/acm2/rent_a_codec.h"
+#include "webrtc/system_wrappers/include/clock.h"
+#include "webrtc/system_wrappers/include/trace.h"
+
+namespace webrtc {
+
+// Create module
+AudioCodingModule* AudioCodingModule::Create(int id) {
+  Config config;
+  config.id = id;
+  config.clock = Clock::GetRealTimeClock();
+  return Create(config);
+}
+
+AudioCodingModule* AudioCodingModule::Create(int id, Clock* clock) {
+  Config config;
+  config.id = id;
+  config.clock = clock;
+  return Create(config);
+}
+
+AudioCodingModule* AudioCodingModule::Create(const Config& config) {
+  return new acm2::AudioCodingModuleImpl(config);
+}
+
+int AudioCodingModule::NumberOfCodecs() {
+  return static_cast<int>(acm2::RentACodec::NumberOfCodecs());
+}
+
+int AudioCodingModule::Codec(int list_id, CodecInst* codec) {
+  auto codec_id = acm2::RentACodec::CodecIdFromIndex(list_id);
+  if (!codec_id)
+    return -1;
+  auto ci = acm2::RentACodec::CodecInstById(*codec_id);
+  if (!ci)
+    return -1;
+  *codec = *ci;
+  return 0;
+}
+
+int AudioCodingModule::Codec(const char* payload_name,
+                             CodecInst* codec,
+                             int sampling_freq_hz,
+                             int channels) {
+  rtc::Optional<CodecInst> ci = acm2::RentACodec::CodecInstByParams(
+      payload_name, sampling_freq_hz, channels);
+  if (ci) {
+    *codec = *ci;
+    return 0;
+  } else {
+    // We couldn't find a matching codec, so set the parameters to unacceptable
+    // values and return.
+    codec->plname[0] = '\0';
+    codec->pltype = -1;
+    codec->pacsize = 0;
+    codec->rate = 0;
+    codec->plfreq = 0;
+    return -1;
+  }
+}
+
+int AudioCodingModule::Codec(const char* payload_name,
+                             int sampling_freq_hz,
+                             int channels) {
+  rtc::Optional<acm2::RentACodec::CodecId> ci =
+      acm2::RentACodec::CodecIdByParams(payload_name, sampling_freq_hz,
+                                        channels);
+  if (!ci)
+    return -1;
+  rtc::Optional<int> i = acm2::RentACodec::CodecIndexFromId(*ci);
+  return i ? *i : -1;
+}
+
+// Checks the validity of the parameters of the given codec
+bool AudioCodingModule::IsCodecValid(const CodecInst& codec) {
+  bool valid = acm2::RentACodec::IsCodecValid(codec);
+  if (!valid)
+    WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, -1,
+                 "Invalid codec setting");
+  return valid;
+}
+
+}  // namespace webrtc