Move RtcEventLogOutput to api/
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}
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4bece9a4f0
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@ -134,6 +134,12 @@ rtc_source_set("libjingle_api_deprecated_headers") {
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]
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]
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}
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}
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rtc_source_set("libjingle_logging_api") {
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sources = [
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"rtceventlogoutput.h",
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]
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}
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rtc_source_set("ortc_api") {
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rtc_source_set("ortc_api") {
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check_includes = false # TODO(deadbeef): Remove (bugs.webrtc.org/6828)
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check_includes = false # TODO(deadbeef): Remove (bugs.webrtc.org/6828)
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sources = [
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sources = [
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@ -8,13 +8,14 @@
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* be found in the AUTHORS file in the root of the source tree.
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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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*/
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#ifndef LOGGING_RTC_EVENT_LOG_OUTPUT_RTC_EVENT_LOG_OUTPUT_H_
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#ifndef API_RTCEVENTLOGOUTPUT_H_
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#define LOGGING_RTC_EVENT_LOG_OUTPUT_RTC_EVENT_LOG_OUTPUT_H_
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#define API_RTCEVENTLOGOUTPUT_H_
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#include <string>
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#include <string>
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namespace webrtc {
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namespace webrtc {
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// NOTE: This class is still under development and may change without notice.
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class RtcEventLogOutput {
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class RtcEventLogOutput {
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public:
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public:
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virtual ~RtcEventLogOutput() = default;
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virtual ~RtcEventLogOutput() = default;
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@ -35,4 +36,4 @@ class RtcEventLogOutput {
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} // namespace webrtc
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} // namespace webrtc
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#endif // LOGGING_RTC_EVENT_LOG_OUTPUT_RTC_EVENT_LOG_OUTPUT_H_
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#endif // API_RTCEVENTLOGOUTPUT_H_
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@ -59,7 +59,6 @@ rtc_source_set("rtc_event_log_api") {
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"rtc_event_log/events/rtc_event_video_receive_stream_config.h",
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"rtc_event_log/events/rtc_event_video_receive_stream_config.h",
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"rtc_event_log/events/rtc_event_video_send_stream_config.cc",
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"rtc_event_log/events/rtc_event_video_send_stream_config.cc",
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"rtc_event_log/events/rtc_event_video_send_stream_config.h",
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"rtc_event_log/events/rtc_event_video_send_stream_config.h",
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"rtc_event_log/output/rtc_event_log_output.h",
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"rtc_event_log/output/rtc_event_log_output_file.cc",
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"rtc_event_log/output/rtc_event_log_output_file.cc",
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"rtc_event_log/output/rtc_event_log_output_file.h",
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"rtc_event_log/output/rtc_event_log_output_file.h",
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"rtc_event_log/rtc_event_log.h",
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"rtc_event_log/rtc_event_log.h",
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@ -71,6 +70,7 @@ rtc_source_set("rtc_event_log_api") {
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deps = [
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deps = [
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"..:webrtc_common",
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"..:webrtc_common",
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"../api:array_view",
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"../api:array_view",
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"../api:libjingle_logging_api",
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"../api:libjingle_peerconnection_api",
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"../api:libjingle_peerconnection_api",
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"../call:video_stream_api",
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"../call:video_stream_api",
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"../modules/audio_coding:audio_network_adaptor_config",
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"../modules/audio_coding:audio_network_adaptor_config",
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@ -16,7 +16,7 @@
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#include <memory>
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#include <memory>
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#include <string>
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#include <string>
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#include "logging/rtc_event_log/output/rtc_event_log_output.h"
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#include "api/rtceventlogoutput.h"
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#include "rtc_base/platform_file.h" // Can't neatly forward PlatformFile.
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#include "rtc_base/platform_file.h" // Can't neatly forward PlatformFile.
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namespace webrtc {
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namespace webrtc {
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@ -14,8 +14,8 @@
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#include <memory>
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#include <memory>
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#include <string>
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#include <string>
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#include "api/rtceventlogoutput.h"
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#include "logging/rtc_event_log/events/rtc_event.h"
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#include "logging/rtc_event_log/events/rtc_event.h"
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#include "logging/rtc_event_log/output/rtc_event_log_output.h"
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namespace webrtc {
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namespace webrtc {
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