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Issue 2912323003: Fix a bug in RtcEventLogSource (Closed)

Created:
3 years, 6 months ago by hlundin-webrtc
Modified:
3 years, 6 months ago
Reviewers:
ivoc
CC:
webrtc-reviews_webrtc.org, danilchap, AleBzk, peah-webrtc, zhuangzesen_agora.io, stefan-webrtc, tlegrand-webrtc, tterriberry_mozilla.com, audio-team_agora.io, hlundin-webrtc, kwiberg-webrtc, minyue-webrtc, mflodman, perkj_webrtc
Target Ref:
refs/heads/master
Project:
webrtc
Visibility:
Public.

Description

Fix a bug in RtcEventLogSource A recent change (https://codereview.webrtc.org/2855143002/) introduced a bug in RtcEventLogSource::NextPacket(). The rtp_packet_index_ must be incremented when a valid packet is found and delivered. Otherwise, the same packet will be delivered over and over again. The recent change also altered the way that audio packets are sifted out. Now, the RTP header is always parsed before discarding any non-audio packets. This means that RtpHeaderParser::Parse is always called, also with video packets, which sometimes contain padding. When header-only dumps (such as RtcEventLogs) are created, the payload is stripped, and the payload length is equal to the RTP header length. However, if the original packet was padded, the RTP header will carry information about this padding length, and the parser will check that the pyaload length is at least the header + padding. This is not the case for header-only dumps, and the parser will return an error. In this CL, we ignore that error when a header-only packet has padding length larger than 0. BUG=webrtc:7538 Review-Url: https://codereview.webrtc.org/2912323003 Cr-Commit-Position: refs/heads/master@{#18385} Committed: https://chromium.googlesource.com/external/webrtc/+/7a2862a933addface23586ef01ac555543a228c7

Patch Set 1 #

Total comments: 4

Patch Set 2 : Making a local solution in the test code #

Total comments: 1

Patch Set 3 : One more small fix #

Unified diffs Side-by-side diffs Delta from patch set Stats (+9 lines, -4 lines) Patch
M webrtc/modules/audio_coding/neteq/tools/packet.cc View 1 1 chunk +7 lines, -2 lines 0 comments Download
M webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.cc View 1 chunk +1 line, -0 lines 0 comments Download
M webrtc/modules/audio_coding/neteq/tools/rtp_file_source.cc View 1 2 1 chunk +1 line, -2 lines 0 comments Download

Messages

Total messages: 18 (8 generated)
hlundin-webrtc
PTAL at this small bugfix. danilchap: webrtc/modules/rtp_rtcp/source/rtp_utility.cc ivoc: webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.cc Thanks!
3 years, 6 months ago (2017-05-31 15:23:46 UTC) #3
danilchap
https://codereview.webrtc.org/2912323003/diff/1/webrtc/modules/rtp_rtcp/source/rtp_utility.cc File webrtc/modules/rtp_rtcp/source/rtp_utility.cc (right): https://codereview.webrtc.org/2912323003/diff/1/webrtc/modules/rtp_rtcp/source/rtp_utility.cc#newcode302 webrtc/modules/rtp_rtcp/source/rtp_utility.cc:302: if (header->headerLength != static_cast<size_t>(length) && this change looks dangerous ...
3 years, 6 months ago (2017-05-31 17:30:21 UTC) #4
danilchap
https://codereview.webrtc.org/2912323003/diff/1/webrtc/modules/rtp_rtcp/source/rtp_utility.cc File webrtc/modules/rtp_rtcp/source/rtp_utility.cc (right): https://codereview.webrtc.org/2912323003/diff/1/webrtc/modules/rtp_rtcp/source/rtp_utility.cc#newcode302 webrtc/modules/rtp_rtcp/source/rtp_utility.cc:302: if (header->headerLength != static_cast<size_t>(length) && On 2017/05/31 17:30:21, danilchap ...
3 years, 6 months ago (2017-05-31 18:57:14 UTC) #5
hlundin-webrtc
https://codereview.webrtc.org/2912323003/diff/1/webrtc/modules/rtp_rtcp/source/rtp_utility.cc File webrtc/modules/rtp_rtcp/source/rtp_utility.cc (right): https://codereview.webrtc.org/2912323003/diff/1/webrtc/modules/rtp_rtcp/source/rtp_utility.cc#newcode302 webrtc/modules/rtp_rtcp/source/rtp_utility.cc:302: if (header->headerLength != static_cast<size_t>(length) && On 2017/05/31 18:57:14, danilchap ...
3 years, 6 months ago (2017-05-31 21:02:57 UTC) #6
ivoc
webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.cc lgtm
3 years, 6 months ago (2017-06-01 07:10:39 UTC) #7
danilchap
https://codereview.webrtc.org/2912323003/diff/1/webrtc/modules/rtp_rtcp/source/rtp_utility.cc File webrtc/modules/rtp_rtcp/source/rtp_utility.cc (right): https://codereview.webrtc.org/2912323003/diff/1/webrtc/modules/rtp_rtcp/source/rtp_utility.cc#newcode302 webrtc/modules/rtp_rtcp/source/rtp_utility.cc:302: if (header->headerLength != static_cast<size_t>(length) && On 2017/05/31 21:02:56, hlundin-webrtc ...
3 years, 6 months ago (2017-06-01 09:45:15 UTC) #8
hlundin-webrtc
This seems a bit hairy. Since the problem I'm trying to solve is for test ...
3 years, 6 months ago (2017-06-01 10:12:17 UTC) #11
ivoc
https://codereview.webrtc.org/2912323003/diff/20001/webrtc/modules/audio_coding/neteq/tools/packet.cc File webrtc/modules/audio_coding/neteq/tools/packet.cc (right): https://codereview.webrtc.org/2912323003/diff/20001/webrtc/modules/audio_coding/neteq/tools/packet.cc#newcode138 webrtc/modules/audio_coding/neteq/tools/packet.cc:138: if (!valid_header && !header_only_with_padding) { It's a bit unfortunate ...
3 years, 6 months ago (2017-06-01 13:02:25 UTC) #12
commit-bot: I haz the power
CQ is trying da patch. Follow status at: https://chromium-cq-status.appspot.com/v2/patch-status/codereview.webrtc.org/2912323003/40001
3 years, 6 months ago (2017-06-01 13:31:42 UTC) #15
commit-bot: I haz the power
3 years, 6 months ago (2017-06-01 14:41:16 UTC) #18
Message was sent while issue was closed.
Committed patchset #3 (id:40001) as
https://chromium.googlesource.com/external/webrtc/+/7a2862a933addface23586ef0...

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