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Add test for logging of large compound RTCP packets.

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Bug: chromium:1134107
Change-Id: Ic6ce50d33700c05733747584ce45480660cf64c9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/188583
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32445}
This commit is contained in:
Bjorn Terelius
2020-10-19 19:01:21 +02:00
committed by Commit Bot
parent e0e2705f0d
commit 945b7d8e31
5 changed files with 85 additions and 43 deletions
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6
logging/rtc_event_log/rtc_event_log_parser.h
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@ -670,8 +670,7 @@ class ParsedRtcEventLog {
// NB: The packet must have space for at least IP_PACKET_SIZE bytes.
ParseStatus GetRtcpPacket(const rtclog::Event& event,
PacketDirection* incoming,
uint8_t* packet,
size_t* length) const;
std::vector<uint8_t>* packet) const;
ParseStatusOr<rtclog::StreamConfig> GetVideoReceiveConfig(
const rtclog::Event& event) const;
@ -873,8 +872,7 @@ class ParsedRtcEventLog {
std::vector<LoggedRouteChangeEvent> route_change_events_;
std::vector<LoggedRemoteEstimateEvent> remote_estimate_events_;
uint8_t last_incoming_rtcp_packet_[IP_PACKET_SIZE];
uint8_t last_incoming_rtcp_packet_length_;
std::vector<uint8_t> last_incoming_rtcp_packet_;
int64_t first_timestamp_;
int64_t last_timestamp_;