This is a reland of e6ee8fab7e
Original change's description:
> Deprecate microsecond timestamps in RTC event log.
>
> (Microsecond timestamps are only used in the legacy wire-format,
> and the clocks only have microsecond resolution on some platforms.)
>
> Also convert structs on the parsing side to use a Timestamp instead
> of a uint64_t to represent the log time.
>
> Bug: webrtc:11933
> Change-Id: Ide5a0217d99f13f2e243115b163f13e0525648c7
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219467
> Commit-Queue: Björn Terelius <terelius@webrtc.org>
> Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#34097}
Bug: webrtc:11933
Change-Id: I295be966ee96b50719ceb4690dad7e7ce958dbac
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221361
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34321}
This reverts commit e6ee8fab7e.
Reason for revert: Breaks downstream test
Original change's description:
> Deprecate microsecond timestamps in RTC event log.
>
> (Microsecond timestamps are only used in the legacy wire-format,
> and the clocks only have microsecond resolution on some platforms.)
>
> Also convert structs on the parsing side to use a Timestamp instead
> of a uint64_t to represent the log time.
>
> Bug: webrtc:11933
> Change-Id: Ide5a0217d99f13f2e243115b163f13e0525648c7
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219467
> Commit-Queue: Björn Terelius <terelius@webrtc.org>
> Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#34097}
TBR=terelius@webrtc.org,srte@webrtc.org,crodbro@webrtc.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com
Change-Id: I478c9a4a1664b984891c4fcfc78f0ce9a51fe4c0
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:11933
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219636
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34100}
(Microsecond timestamps are only used in the legacy wire-format,
and the clocks only have microsecond resolution on some platforms.)
Also convert structs on the parsing side to use a Timestamp instead
of a uint64_t to represent the log time.
Bug: webrtc:11933
Change-Id: Ide5a0217d99f13f2e243115b163f13e0525648c7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219467
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34097}
Correspondingly, change the parser so that it provides the frames
grouped by SSRC.
Also fix a small bug that made the audio playout test terminate
too early before verifying correct logging of all events.
Bug: webrtc:8802
Change-Id: I363ef120cf88fe99290998cbc14ab5dbf32e9607
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/181066
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31962}
This CL adds the possibility to log metainformation about
decoded frames in RTC event log, including encoding parsing
and tests. It will be wired up in a followup CL.
Bug: webrtc:8802
Change-Id: Ied598b266513d0f63fce0484d741af1782607e74
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/181061
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31873}
This change does not include receive_timestamps for ACKs, because there is 1 problem.
That problem will be resolved in a separate change.
I am getting receive_timestamp errors that have to do with delta compression with optional fields.
Two failure modes that I noticed:
1) the base event does not have the timestamp: it crashes with length validation
# Check failed: base <= MaxUnsignedValueOfBitWidth(params_.value_width_bits()) (1820716 vs. 131071)
2) all events are null, it crashes with assert that X events were expected, but no events were deserialized.
Bug: webrtc:9719
Change-Id: I5d1bbb95dfd15ca7321667aad5e4d89c085e9c06
Reviewed-on: https://webrtc-review.googlesource.com/c/122360
Commit-Queue: Peter Slatala <psla@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26668}
This is a follow-up to
https://webrtc-review.googlesource.com/c/src/+/106280.
This time the whole code base is covered.
Some files may have not been fixed though, whenever the IWYU tool
was breaking the build.
Bug: webrtc:8311
Change-Id: I2c31f552a87e887d33931d46e87b6208b1e483ef
Reviewed-on: https://webrtc-review.googlesource.com/c/111965
Commit-Queue: Yves Gerey <yvesg@google.com>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25830}
The new event log format makes use of delta encoding to compress
parts of the log.
Bug: webrtc:8111
Change-Id: I7bec839555323a7537dcec831d4ac1d5eb109932
Reviewed-on: https://webrtc-review.googlesource.com/c/109161
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25584}
We're no longer verifying CSRCs or configurations for remb, rtcp mode
and codec since we're planning to drop those fields from the log in an upcoming CL.
Bug: webrtc:8111
Change-Id: I38a7d87b21f8e6d8a791d8e27a0f54c293f3d340
Reviewed-on: https://webrtc-review.googlesource.com/c/106380
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25229}
The new API stores events gathered by event type. For example, it is
possible to ask for a list of all incoming RTCP messages or all audio
playout events.
The new API is experimental and may change over next few weeks. Once
it has stabilized and all unit tests and existing tools have been
ported to the new API, the old one will be removed.
This CL also updates the event_log_visualizer tool to use the new
parser API. This is not a funcional change except for:
- Incoming and outgoing audio level are now drawn in two separate plots.
- Incoming and outgoing timstamps are now drawn in two separate plots.
- RTCP count is no longer split into Video and Audio. It also counts
all RTCP packets rather than only specific message types.
- Slight timing difference in sendside BWE simulation due to only
iterating over transport feedbacks and not over all RTCP packets.
This timing changes are not visible in the plots.
Media type for RTCP messages might not be identified correctly by
rtc_event_log2text anymore. On the other hand, assigning a specific
media type to an RTCP packet was a bit hacky to begin with.
Bug: webrtc:8111
Change-Id: Ib244338c86a2c1a010c668a7aba440482023b512
Reviewed-on: https://webrtc-review.googlesource.com/73140
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23056}
This reverts commit 9e336ec0b8.
Reason for revert: Code can accidentally include the deprecated parser but link with the new one, or vice versa. Reverting to fix naming.
Original change's description:
> Create new API for RtcEventLogParser.
>
> The new API stores events gathered by event type. For example, it is
> possible to ask fo a list of all incoming RTCP messages or all audio
> playout events.
>
> The new API is experimental and may change over next few weeks. Once
> it has stabilized and all unit tests and existing tools have been
> ported to the new API, the old one will be removed.
>
> This CL also updates the event_log_visualizer tool to use the new
> parser API. This is not a funcional change except for:
> - Incoming and outgoing audio level are now drawn in two separate plots.
> - Incoming and outgoing timstamps are now drawn in two separate plots.
> - RTCP count is no longer split into Video and Audio. It also counts
> all RTCP packets rather than only specific message types.
> - Slight timing difference in sendside BWE simulation due to only
> iterating over transport feedbacks and not over all RTCP packets.
> This timing changes are not visible in the plots.
>
>
> Media type for RTCP messages might not be identified correctly by
> rtc_event_log2text anymore. On the other hand, assigning a specific
> media type to an RTCP packet was a bit hacky to begin with.
>
> Bug: webrtc:8111
> Change-Id: I8e7168302beb69b2e163a097a2a142b86dd4a26b
> Reviewed-on: https://webrtc-review.googlesource.com/60865
> Reviewed-by: Minyue Li <minyue@webrtc.org>
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Commit-Queue: Björn Terelius <terelius@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23015}
TBR=terelius@webrtc.org,srte@webrtc.org,minyue@webrtc.org
Change-Id: Ib4bbcf0563423675a3cc1dce59ebf665e0c5dae9
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8111
Reviewed-on: https://webrtc-review.googlesource.com/72500
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23026}
The new API stores events gathered by event type. For example, it is
possible to ask fo a list of all incoming RTCP messages or all audio
playout events.
The new API is experimental and may change over next few weeks. Once
it has stabilized and all unit tests and existing tools have been
ported to the new API, the old one will be removed.
This CL also updates the event_log_visualizer tool to use the new
parser API. This is not a funcional change except for:
- Incoming and outgoing audio level are now drawn in two separate plots.
- Incoming and outgoing timstamps are now drawn in two separate plots.
- RTCP count is no longer split into Video and Audio. It also counts
all RTCP packets rather than only specific message types.
- Slight timing difference in sendside BWE simulation due to only
iterating over transport feedbacks and not over all RTCP packets.
This timing changes are not visible in the plots.
Media type for RTCP messages might not be identified correctly by
rtc_event_log2text anymore. On the other hand, assigning a specific
media type to an RTCP packet was a bit hacky to begin with.
Bug: webrtc:8111
Change-Id: I8e7168302beb69b2e163a097a2a142b86dd4a26b
Reviewed-on: https://webrtc-review.googlesource.com/60865
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23015}
Change LogIncomingRtcpPacket and LogOutgoingRtcpPacket to take ArrayView<uint8_t>.
Split LogSessionAndReadBack into three functions and create class to share state between them.
Split VerifyRtpEvent into one incoming and one outgoing version.
Originally uploaded as https://codereview.webrtc.org/2997973002/
Bug: webrtc:8111
Change-Id: I22bdc35163bef60bc8293679226b19e41e8f49b3
Reviewed-on: https://webrtc-review.googlesource.com/5020
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20063}
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC
from src/webrtc to src/ (in order to preserve an healthy git history).
This CL takes care of fixing header guards, #include paths, etc...
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org
Bug: chromium:611808
Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578
Reviewed-on: https://webrtc-review.googlesource.com/1561
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19846}
In order to eliminate the WebRTC Subtree mirror in Chromium,
WebRTC is moving the content of the src/webrtc directory up
to the src/ directory.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org
Bug: chromium:611808
Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38
Reviewed-on: https://webrtc-review.googlesource.com/1560
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19845}
2017-09-15 04:25:06 +00:00
Renamed from webrtc/logging/rtc_event_log/rtc_event_log_unittest_helper.h (Browse further)
