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Created:
5 years, 3 months ago by stefan-webrtc Modified:
5 years, 3 months ago CC:
webrtc-reviews_webrtc.org, yujie_mao (webrtc), tterriberry_mozilla.com, andresp, perkj_webrtc, mflodman Base URL:
https://chromium.googlesource.com/external/webrtc.git@master Target Ref:
refs/pending/heads/master Project:
webrtc Visibility:
Public. |
DescriptionFix BWE bug where audio has timestamps in us.
The BWE expects arrival timestamps in ms, while the audio path delivered
them in us, causing the BWE to break down under the combined audio/video
BWE experiment. This was introduced in r9892 (https://chromium.googlesource.com/external/webrtc/+/68786d20400f1f3744ad83549325665c18ea9e5b).
BUG=webrtc:4758
R=mflodman@webrtc.org, sprang@webrtc.org
Committed: https://crrev.com/8bffba71072577ef8e81f48b615f231880de25a0
Cr-Commit-Position: refs/heads/master@{#10032}
Patch Set 1 #
Total comments: 4
Patch Set 2 : Unittest added. #Patch Set 3 : . #
Messages
Total messages: 28 (9 generated)
stefan@webrtc.org changed reviewers: + pbos@webrtc.org
PTAL
stefan@webrtc.org changed reviewers: + sprang@webrtc.org
The CQ bit was checked by stefan@webrtc.org to run a CQ dry run
Dry run: CQ is trying da patch. Follow status at https://chromium-cq-status.appspot.com/patch-status/1360913004/1 View timeline at https://chromium-cq-status.appspot.com/patch-timeline/1360913004/1
https://codereview.webrtc.org/1360913004/diff/1/webrtc/video/audio_receive_st... File webrtc/video/audio_receive_stream.cc (right): https://codereview.webrtc.org/1360913004/diff/1/webrtc/video/audio_receive_st... webrtc/video/audio_receive_stream.cc:105: arrival_time_ms = (packet_time.timestamp + 500) / 1000; I can't seem to find a unit tests at all for this method. Should we add small one, at least for this?
The CQ bit was unchecked by commit-bot@chromium.org
Dry run: Try jobs failed on following builders: win_x64_gn_dbg on tryserver.webrtc (JOB_FAILED, http://build.chromium.org/p/tryserver.webrtc/builders/win_x64_gn_dbg/builds/4127)
On 2015/09/23 08:46:26, språng wrote: > https://codereview.webrtc.org/1360913004/diff/1/webrtc/video/audio_receive_st... > File webrtc/video/audio_receive_stream.cc (right): > > https://codereview.webrtc.org/1360913004/diff/1/webrtc/video/audio_receive_st... > webrtc/video/audio_receive_stream.cc:105: arrival_time_ms = > (packet_time.timestamp + 500) / 1000; > I can't seem to find a unit tests at all for this method. Should we add small > one, at least for this? I was looking for one as well, and couldn't find it. I'll write a simple one which mocks the remote bitrate estimator.
mflodman@webrtc.org changed reviewers: + mflodman@webrtc.org
https://codereview.webrtc.org/1360913004/diff/1/webrtc/video/audio_receive_st... File webrtc/video/audio_receive_stream.cc (right): https://codereview.webrtc.org/1360913004/diff/1/webrtc/video/audio_receive_st... webrtc/video/audio_receive_stream.cc:105: arrival_time_ms = (packet_time.timestamp + 500) / 1000; Not sure I understood correctly, but shouldn't AudioReceiveStream::DeliverRtp and VideoReceiveStream::DeliverRtp assume the same units in PacketTime? I.e. I think this should be done earlier than here.
https://codereview.webrtc.org/1360913004/diff/1/webrtc/video/audio_receive_st... File webrtc/video/audio_receive_stream.cc (right): https://codereview.webrtc.org/1360913004/diff/1/webrtc/video/audio_receive_st... webrtc/video/audio_receive_stream.cc:105: arrival_time_ms = (packet_time.timestamp + 500) / 1000; On 2015/09/23 12:11:41, mflodman wrote: > Not sure I understood correctly, but shouldn't AudioReceiveStream::DeliverRtp > and VideoReceiveStream::DeliverRtp assume the same units in PacketTime? > > I.e. I think this should be done earlier than here. Or IncomingPacket accept a PacketTime?
The CQ bit was checked by stefan@webrtc.org to run a CQ dry run
Dry run: CQ is trying da patch. Follow status at https://chromium-cq-status.appspot.com/patch-status/1360913004/20001 View timeline at https://chromium-cq-status.appspot.com/patch-timeline/1360913004/20001
The CQ bit was unchecked by commit-bot@chromium.org
Dry run: Try jobs failed on following builders: android_gn_dbg on tryserver.webrtc (JOB_FAILED, http://build.chromium.org/p/tryserver.webrtc/builders/android_gn_dbg/builds/5818)
https://codereview.webrtc.org/1360913004/diff/1/webrtc/video/audio_receive_st... File webrtc/video/audio_receive_stream.cc (right): https://codereview.webrtc.org/1360913004/diff/1/webrtc/video/audio_receive_st... webrtc/video/audio_receive_stream.cc:105: arrival_time_ms = (packet_time.timestamp + 500) / 1000; On 2015/09/23 12:18:54, pbos-webrtc wrote: > On 2015/09/23 12:11:41, mflodman wrote: > > Not sure I understood correctly, but shouldn't AudioReceiveStream::DeliverRtp > > and VideoReceiveStream::DeliverRtp assume the same units in PacketTime? > > > > I.e. I think this should be done earlier than here. > > Or IncomingPacket accept a PacketTime? The same conversion is done for video in vie_receiver.cc. I could pass in PacketTime instead, but I don't think it's worth it as PacketTime doesn't make as much sense on the send-side, and we are moving all of it there.
My mistake, ignore my previous comment. LGTM
Erik, could you take a look at the unittest I added?
The CQ bit was checked by stefan@webrtc.org to run a CQ dry run
Dry run: CQ is trying da patch. Follow status at https://chromium-cq-status.appspot.com/patch-status/1360913004/40001 View timeline at https://chromium-cq-status.appspot.com/patch-timeline/1360913004/40001
lgtm
The CQ bit was unchecked by commit-bot@chromium.org
Dry run: Try jobs failed on following builders: win_drmemory_light on tryserver.webrtc (JOB_FAILED, http://build.chromium.org/p/tryserver.webrtc/builders/win_drmemory_light/buil...)
Message was sent while issue was closed.
Committed patchset #3 (id:40001) manually as 8bffba71072577ef8e81f48b615f231880de25a0 (presubmit successful).
Message was sent while issue was closed.
Patchset 3 (id:??) landed as https://crrev.com/8bffba71072577ef8e81f48b615f231880de25a0 Cr-Commit-Position: refs/heads/master@{#10032} |