
Changed the BUILD.gn file that was lacking some necessary items which caused Chromium to break. Original review: https://webrtc-codereview.appspot.com/52059005/ The revert of the original CL was commit 7a75415419cbd52d798f9226010e9190e1cbad53. BUG=webrtc:4741 R=henrik.lundin@webrtc.org Review URL: https://codereview.webrtc.org/1200833002. Cr-Commit-Position: refs/heads/master@{#9489}
60 lines
2.0 KiB
C++
60 lines
2.0 KiB
C++
/*
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* Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_DUMP_H_
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#define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_DUMP_H_
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#include <string>
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#include "webrtc/base/scoped_ptr.h"
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namespace webrtc {
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// Forward declaration of storage class that is automatically generated from
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// the protobuf file.
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class ACMDumpEventStream;
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class AcmDumpImpl;
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class AcmDump {
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public:
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// The types of debug events that are currently supported for logging.
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enum class DebugEvent { kLogStart, kLogEnd, kAudioPlayout };
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virtual ~AcmDump() {}
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static rtc::scoped_ptr<AcmDump> Create();
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// Starts logging for the specified duration to the specified file.
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// The logging will stop automatically after the specified duration.
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// If the file already exists it will be overwritten.
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// The function will return false on failure.
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virtual void StartLogging(const std::string& file_name, int duration_ms) = 0;
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// Logs an incoming or outgoing RTP packet.
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virtual void LogRtpPacket(bool incoming,
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const uint8_t* packet,
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size_t length) = 0;
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// Logs a debug event, with optional message.
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virtual void LogDebugEvent(DebugEvent event_type,
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const std::string& event_message) = 0;
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virtual void LogDebugEvent(DebugEvent event_type) = 0;
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// Reads an AcmDump file and returns true when reading was successful.
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// The result is stored in the given ACMDumpEventStream object.
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static bool ParseAcmDump(const std::string& file_name,
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ACMDumpEventStream* result);
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};
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} // namespace webrtc
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#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_DUMP_H_
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