webrtc/modules/audio_coding/neteq/mock/mock_decoder_database.h
Danil Chapovalov b602123a5a Replace rtc::Optional with absl::optional in modules/audio_coding
This is a no-op change because rtc::Optional is an alias to absl::optional

This CL generated by running script with parameter 'modules/audio_coding'

find $@ -type f \( -name \*.h -o -name \*.cc \) \
-exec sed -i 's|rtc::Optional|absl::optional|g' {} \+ \
-exec sed -i 's|rtc::nullopt|absl::nullopt|g' {} \+ \
-exec sed -i 's|#include "api/optional.h"|#include "absl/types/optional.h"|' {} \+

find $@ -type f -name BUILD.gn \
-exec sed -r -i 's|"[\./api]*:optional"|"//third_party/abseil-cpp/absl/types:optional"|' {} \+;

git cl format

Bug: webrtc:9078
Change-Id: Ic980ee605148fdb160666d4aa03cc87175e48fe8
Reviewed-on: https://webrtc-review.googlesource.com/84130
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23659}
2018-06-19 12:46:20 +00:00

61 lines
2.1 KiB
C++

/*
* 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.
*/
#ifndef MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_DECODER_DATABASE_H_
#define MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_DECODER_DATABASE_H_
#include <string>
#include "modules/audio_coding/neteq/decoder_database.h"
#include "test/gmock.h"
namespace webrtc {
class MockDecoderDatabase : public DecoderDatabase {
public:
explicit MockDecoderDatabase(
rtc::scoped_refptr<AudioDecoderFactory> factory = nullptr)
: DecoderDatabase(factory, absl::nullopt) {}
virtual ~MockDecoderDatabase() { Die(); }
MOCK_METHOD0(Die, void());
MOCK_CONST_METHOD0(Empty,
bool());
MOCK_CONST_METHOD0(Size,
int());
MOCK_METHOD0(Reset,
void());
MOCK_METHOD3(RegisterPayload,
int(uint8_t rtp_payload_type, NetEqDecoder codec_type,
const std::string& name));
MOCK_METHOD2(RegisterPayload,
int(int rtp_payload_type, const SdpAudioFormat& audio_format));
MOCK_METHOD4(InsertExternal,
int(uint8_t rtp_payload_type,
NetEqDecoder codec_type,
const std::string& codec_name,
AudioDecoder* decoder));
MOCK_METHOD1(Remove,
int(uint8_t rtp_payload_type));
MOCK_METHOD0(RemoveAll, void());
MOCK_CONST_METHOD1(GetDecoderInfo,
const DecoderInfo*(uint8_t rtp_payload_type));
MOCK_METHOD2(SetActiveDecoder,
int(uint8_t rtp_payload_type, bool* new_decoder));
MOCK_CONST_METHOD0(GetActiveDecoder,
AudioDecoder*());
MOCK_METHOD1(SetActiveCngDecoder,
int(uint8_t rtp_payload_type));
MOCK_CONST_METHOD0(GetActiveCngDecoder,
ComfortNoiseDecoder*());
};
} // namespace webrtc
#endif // MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_DECODER_DATABASE_H_