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}
diff --git a/webrtc/modules/audio_coding/acm2/codec_manager.h b/webrtc/modules/audio_coding/acm2/codec_manager.h
new file mode 100644
index 0000000..61832e4
--- /dev/null
+++ b/webrtc/modules/audio_coding/acm2/codec_manager.h
@@ -0,0 +1,81 @@
+/*
+ * Copyright (c) 2015 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.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_ACM2_CODEC_MANAGER_H_
+#define WEBRTC_MODULES_AUDIO_CODING_ACM2_CODEC_MANAGER_H_
+
+#include <map>
+
+#include "webrtc/base/constructormagic.h"
+#include "webrtc/base/optional.h"
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/base/thread_checker.h"
+#include "webrtc/modules/audio_coding/acm2/rent_a_codec.h"
+#include "webrtc/modules/audio_coding/include/audio_coding_module_typedefs.h"
+#include "webrtc/common_types.h"
+
+namespace webrtc {
+
+class AudioDecoder;
+class AudioEncoder;
+
+namespace acm2 {
+
+class CodecManager final {
+ public:
+ CodecManager();
+ ~CodecManager();
+
+ int RegisterEncoder(const CodecInst& send_codec);
+
+ void RegisterEncoder(AudioEncoder* external_speech_encoder);
+
+ rtc::Optional<CodecInst> GetCodecInst() const;
+
+ bool SetCopyRed(bool enable);
+
+ int SetVAD(bool enable, ACMVADMode mode);
+
+ void VAD(bool* dtx_enabled, bool* vad_enabled, ACMVADMode* mode) const;
+
+ int SetCodecFEC(bool enable_codec_fec);
+
+ // Returns a pointer to AudioDecoder of the given codec. For iSAC, encoding
+ // and decoding have to be performed on a shared codec instance. By calling
+ // this method, we get the codec instance that ACM owns.
+ // If |codec| does not share an instance between encoder and decoder, returns
+ // null.
+ AudioDecoder* GetAudioDecoder(const CodecInst& codec);
+
+ bool red_enabled() const { return codec_stack_params_.use_red; }
+
+ bool codec_fec_enabled() const { return codec_stack_params_.use_codec_fec; }
+
+ AudioEncoder* CurrentEncoder() { return rent_a_codec_.GetEncoderStack(); }
+ const AudioEncoder* CurrentEncoder() const {
+ return rent_a_codec_.GetEncoderStack();
+ }
+
+ bool CurrentEncoderIsOpus() const { return encoder_is_opus_; }
+
+ private:
+ rtc::ThreadChecker thread_checker_;
+ CodecInst send_codec_inst_;
+ RentACodec rent_a_codec_;
+ RentACodec::StackParameters codec_stack_params_;
+
+ bool encoder_is_opus_;
+
+ RTC_DISALLOW_COPY_AND_ASSIGN(CodecManager);
+};
+
+} // namespace acm2
+} // namespace webrtc
+#endif // WEBRTC_MODULES_AUDIO_CODING_ACM2_CODEC_MANAGER_H_