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Move RtcEventLogOutput to api/

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Move RtcEventLogOutput into the API, so that we would be able to change StartRtcEventLog (in PeerConnectionInterface) to use it.

Bug: webrtc:8111
Change-Id: I1d70af792ec584d3f1a8eced1b66c38e4a360642
Reviewed-on: https://webrtc-review.googlesource.com/7220
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20189}
This commit is contained in:
Elad Alon
2017-10-06 13:07:32 +02:00
committed by Commit Bot
parent 4bece9a4f0
commit 8081073677
5 changed files with 13 additions and 6 deletions
Download Patch File Download Diff File Expand all files Collapse all files

2
logging/BUILD.gn
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@ -59,7 +59,6 @@ rtc_source_set("rtc_event_log_api") {
"rtc_event_log/events/rtc_event_video_receive_stream_config.h",
"rtc_event_log/events/rtc_event_video_send_stream_config.cc",
"rtc_event_log/events/rtc_event_video_send_stream_config.h",
"rtc_event_log/output/rtc_event_log_output.h",
"rtc_event_log/output/rtc_event_log_output_file.cc",
"rtc_event_log/output/rtc_event_log_output_file.h",
"rtc_event_log/rtc_event_log.h",
@ -71,6 +70,7 @@ rtc_source_set("rtc_event_log_api") {
deps = [
"..:webrtc_common",
"../api:array_view",
"../api:libjingle_logging_api",
"../api:libjingle_peerconnection_api",
"../call:video_stream_api",
"../modules/audio_coding:audio_network_adaptor_config",

38
logging/rtc_event_log/output/rtc_event_log_output.h
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@ -1,38 +0,0 @@
/*
* Copyright (c) 2017 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 LOGGING_RTC_EVENT_LOG_OUTPUT_RTC_EVENT_LOG_OUTPUT_H_
#define LOGGING_RTC_EVENT_LOG_OUTPUT_RTC_EVENT_LOG_OUTPUT_H_
#include <string>
namespace webrtc {
class RtcEventLogOutput {
public:
virtual ~RtcEventLogOutput() = default;
// An output normally starts out active, though that might not always be
// the case (e.g. failed to open a file for writing).
// Once an output has become inactive (e.g. maximum file size reached), it can
// never become active again.
virtual bool IsActive() const = 0;
// Write encoded events to an output. Returns true if the output was
// successfully written in its entirety. Otherwise, no guarantee is given
// about how much data was written, if any. The output sink becomes inactive
// after the first time |false| is returned. Write() may not be called on
// an inactive output sink.
virtual bool Write(const std::string& output) = 0;
};
} // namespace webrtc
#endif // LOGGING_RTC_EVENT_LOG_OUTPUT_RTC_EVENT_LOG_OUTPUT_H_

2
logging/rtc_event_log/output/rtc_event_log_output_file.h
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@ -16,7 +16,7 @@
#include <memory>
#include <string>
#include "logging/rtc_event_log/output/rtc_event_log_output.h"
#include "api/rtceventlogoutput.h"
#include "rtc_base/platform_file.h" // Can't neatly forward PlatformFile.
namespace webrtc {

2
logging/rtc_event_log/rtc_event_log.h
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@ -14,8 +14,8 @@
#include <memory>
#include <string>
#include "api/rtceventlogoutput.h"
#include "logging/rtc_event_log/events/rtc_event.h"
#include "logging/rtc_event_log/output/rtc_event_log_output.h"
namespace webrtc {