Move RtcpStatistics from common_types.h to a new header file
New location is modules/rtp_rtcp/include/rtcp_statistics.h. Bug: webrtc:5876 Change-Id: I85f55b58658588228ed77175226b3479352fd5de Reviewed-on: https://webrtc-review.googlesource.com/c/111961 Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25799}
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@ -25,6 +25,7 @@
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#include "api/transport/bitrate_settings.h"
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#include "call/rtp_config.h"
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#include "logging/rtc_event_log/rtc_event_log.h"
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#include "modules/rtp_rtcp/include/rtcp_statistics.h"
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#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
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namespace rtc {
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@ -26,6 +26,7 @@
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#include "api/video/video_timing.h"
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#include "api/video_codecs/sdp_video_format.h"
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#include "call/rtp_config.h"
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#include "modules/rtp_rtcp/include/rtcp_statistics.h"
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#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
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namespace webrtc {
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@ -27,6 +27,7 @@
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#include "api/video/video_stream_encoder_settings.h"
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#include "api/video_codecs/video_encoder_config.h"
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#include "call/rtp_config.h"
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#include "modules/rtp_rtcp/include/rtcp_statistics.h"
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#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
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namespace webrtc {
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@ -39,29 +39,6 @@ enum FrameType {
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kVideoFrameDelta = 4,
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};
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// Statistics for an RTCP channel
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struct RtcpStatistics {
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RtcpStatistics()
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: fraction_lost(0),
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packets_lost(0),
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extended_highest_sequence_number(0),
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jitter(0) {}
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uint8_t fraction_lost;
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int32_t packets_lost; // Defined as a 24 bit signed integer in RTCP
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uint32_t extended_highest_sequence_number;
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uint32_t jitter;
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};
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class RtcpStatisticsCallback {
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public:
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virtual ~RtcpStatisticsCallback() {}
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virtual void StatisticsUpdated(const RtcpStatistics& statistics,
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uint32_t ssrc) = 0;
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virtual void CNameChanged(const char* cname, uint32_t ssrc) = 0;
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};
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// Statistics for RTCP packet types.
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struct RtcpPacketTypeCounter {
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RtcpPacketTypeCounter()
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@ -2103,6 +2103,7 @@ if (rtc_include_tests) {
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"../../logging:mocks",
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"../../logging:rtc_event_audio",
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"../../logging:rtc_event_log_api",
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"../../modules/rtp_rtcp:rtp_rtcp_format",
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"../../rtc_base:checks",
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"../../rtc_base:protobuf_utils",
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"../../rtc_base:rtc_base",
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@ -22,12 +22,12 @@
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#include "api/audio/audio_frame.h"
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#include "api/audio_codecs/builtin_audio_decoder_factory.h"
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#include "common_types.h" // NOLINT(build/include)
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#include "modules/audio_coding/codecs/pcm16b/pcm16b.h"
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#include "modules/audio_coding/neteq/tools/audio_loop.h"
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#include "modules/audio_coding/neteq/tools/neteq_packet_source_input.h"
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#include "modules/audio_coding/neteq/tools/neteq_test.h"
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#include "modules/audio_coding/neteq/tools/rtp_file_source.h"
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#include "modules/rtp_rtcp/include/rtcp_statistics.h"
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#include "rtc_base/ignore_wundef.h"
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#include "rtc_base/messagedigest.h"
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#include "rtc_base/numerics/safe_conversions.h"
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@ -11,6 +11,7 @@ import("../../webrtc.gni")
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rtc_source_set("rtp_rtcp_format") {
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visibility = [ "*" ]
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public = [
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"include/rtcp_statistics.h",
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"include/rtp_cvo.h",
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"include/rtp_header_extension_map.h",
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"include/rtp_rtcp_defines.h",
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@ -18,6 +18,7 @@
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#include "call/rtp_packet_sink_interface.h"
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#include "modules/include/module.h"
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#include "modules/include/module_common_types.h"
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#include "modules/rtp_rtcp/include/rtcp_statistics.h"
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#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
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#include "modules/rtp_rtcp/source/rtcp_packet/report_block.h"
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#include "rtc_base/deprecation.h"
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36
modules/rtp_rtcp/include/rtcp_statistics.h
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36
modules/rtp_rtcp/include/rtcp_statistics.h
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@ -0,0 +1,36 @@
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/*
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* Copyright (c) 2018 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 MODULES_RTP_RTCP_INCLUDE_RTCP_STATISTICS_H_
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#define MODULES_RTP_RTCP_INCLUDE_RTCP_STATISTICS_H_
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#include <stdint.h>
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namespace webrtc {
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// Statistics for an RTCP channel
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struct RtcpStatistics {
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uint8_t fraction_lost = 0;
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int32_t packets_lost = 0; // Defined as a 24 bit signed integer in RTCP
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uint32_t extended_highest_sequence_number = 0;
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uint32_t jitter = 0;
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};
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class RtcpStatisticsCallback {
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public:
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virtual ~RtcpStatisticsCallback() {}
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virtual void StatisticsUpdated(const RtcpStatistics& statistics,
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uint32_t ssrc) = 0;
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virtual void CNameChanged(const char* cname, uint32_t ssrc) = 0;
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};
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} // namespace webrtc
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#endif // MODULES_RTP_RTCP_INCLUDE_RTCP_STATISTICS_H_
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@ -18,9 +18,9 @@
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#include "absl/types/optional.h"
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#include "api/video/video_bitrate_allocation.h"
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#include "common_types.h" // NOLINT(build/include)
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#include "modules/include/module.h"
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#include "modules/rtp_rtcp/include/flexfec_sender.h"
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#include "modules/rtp_rtcp/include/receive_statistics.h"
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#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
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#include "rtc_base/constructormagic.h"
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#include "rtc_base/deprecation.h"
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@ -17,6 +17,7 @@
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#include <string>
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#include <vector>
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#include "modules/rtp_rtcp/include/rtcp_statistics.h"
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#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
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#include "modules/rtp_rtcp/source/rtcp_nack_stats.h"
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#include "modules/rtp_rtcp/source/rtcp_packet/dlrr.h"
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@ -14,6 +14,7 @@
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#include <map>
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#include <vector>
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#include "modules/rtp_rtcp/include/rtcp_statistics.h"
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#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
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namespace webrtc {
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