This is a follow up to a previous CL that removed direct dependency on
the `cricket::` string globals.
Bug: none
Change-Id: I4d839a36739fc4694ce81b72ee036e83dae580df
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/335420
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41623}
Includes removing the duplicate MockTransformableAudioFrame definition
in test/ in favour of the existing one in api/test/
Bug: webrtc:15802
Change-Id: Ib5f86b8b2095dd4e580cd9ff0038134f8a43cd93
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/336340
Auto-Submit: Tony Herre <herre@google.com>
Commit-Queue: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41622}
Rename RtpVideoSender::SetActiveModules to SetSending to better match
what it does. When a RtpVideoSender::SetSending(true) RTP packets can be
sent on all associcated RTP streams (simulcast streams).
Move registration of RtpRtcpModules to RtpTransportControllerSend to
allow RtpTransportControllerSend to know when there are sending RTP
streams. Purpose is to in later CLs allow RtpTransportControllerSend to
trigger BWE probes.
Bug: webrtc:14928
Change-Id: Ibf6c040b86713cdc4763c4691b7fd794b251eb49
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/335961
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41620}
Instead, always use VideoSendStream::Start.
VideoSendStream::StartPerRtpStream was used for controlling if
individual rtp stream for a RtpEncodingParameter should be able to send RTP packets. It was not used for controlling the actual encoder layers.
With this change RtpEncodingParameter.active still controls actual encoder layers but it does not control if RTP packets can be sent or not.
The cleanup is done to simplify code and in the future allow sending
probe packet on a RtpTransceiver that allows sending, regardless of the
RtpEncodingParameter.active flag.
Bug: webrtc:14928
Change-Id: I896c055ed4de76db58d76f452147c29783f77ae1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/335042
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41619}
This CL does:
- Run IWYU on the relevant elements
- Make connection depend on port_interface, not port
- Make port_allocator depend only on port
- Move some constants from port.h into p2p_constants
This allows a dependency graph without ugly groups.
Bug: webrtc:15796
Change-Id: I0ff0e14eacdfe3b230a8d84902a78eb062d6c8af
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/336320
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41618}
With this CL the migration to absl::optional<T> is complete.
Bug: webrtc:15164
Change-Id: I978d86833fbd39154ed9026206a02ea7496f8c4f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/335640
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@google.com>
Cr-Commit-Position: refs/heads/main@{#41616}
This is a reland of commit 33c7edd58ad0edc71939b9372fff3ab563c1f4a7
taking into account GFD which can be enabled by field trials and somewhat conflicts with DD
Original change's description:
> Enable DD and VLA header extensions by default for Simulcast/SVC
>
> When Simulcast (more than one encoding) or SVC (a scalability mode
> other than the default L1T1) is used, enable the AV1 Dependency
> Descriptor and the video-layer-allocations RTP header extensions by
> default.
>
> The RTP header extensions API can be used to disable them if needed.
>
> BUG=webrtc:15378
>
> Change-Id: I587ac32c9d681461496a136f6950b007e72da86d
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/326100
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Commit-Queue: Philipp Hancke <phancke@microsoft.com>
> Cr-Commit-Position: refs/heads/main@{#41332}
Bug: webrtc:15378
Change-Id: I190edc9435083c0a0a65a6959363f3c41e4a3d1b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/330563
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41615}
With this CL, the only usage of RTCStatsMember within WebRTC is the
actual type alias declaration. It's not referenced anywhere anymore.
This allows us to deleting the type alias, but let's do that in a
standalone CL in case it gets reverted.
Bug: webrtc:15164
Change-Id: I766d07abb62b5ddd524859b8ed749394fc439e52
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/335621
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@google.com>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41612}
This CL splits many of the source files in p2p:rtc_p2p into individual
compile targets.
One target - connection_and_port - was left with multiple source files
because it was too tangled to detangle at once.
Bug: webrtc:15796
Change-Id: I607417e5945306ef64335f40a0ae50f0d15dee6f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/335881
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41611}
Low-Coverage-Reason: EXPERIMENTAL_CODE Code is behind field trial that will only be used for testing.
Bug: webrtc:13322
Change-Id: Ie306be808381b3a20b4e0d58349927bf3524018a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/335840
Reviewed-by: Tomas Lundqvist <tomasl@google.com>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41608}
This is a reland of commit 79ac694d9b70fa9cd7b6a0f00bbee5d7fbbe64de
Original change's description:
> [Stats] Make RTCStatsMember<T> a type alias for absl::optional<T>.
>
> The moment we've all been waiting for.
>
> Step 1: Add type alias (this CL).
> Step 2: Migrate all uses of RTCStatsMember<T> to absl::optional<T>.
> Step 3: Delete type alias.
>
> Bug: webrtc:15164
> Change-Id: I00a7202c0b684fb2c57fcad4f501bccc167f1fa3
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/334680
> Commit-Queue: Henrik Boström <hbos@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Evan Shrubsole <eshr@google.com>
> Cr-Commit-Position: refs/heads/main@{#41593}
# Only unrelated bot failures
NOTRY=True
Bug: webrtc:15164
Change-Id: I0f1991409326679a260cb24907808eaa28385350
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/335960
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@google.com>
Cr-Commit-Position: refs/heads/main@{#41603}
This isn't a problem right now but will be when we move away from
using a free string as the storage type.
Bug: none
Change-Id: I12c2de416f9fce13a8284e7d36fa3c2081453299
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/335880
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Auto-Submit: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41602}
These are no longer needed since the RTP transport runs on the worker
thread now.
Some tests that were too strict on ordering needed change.
Bug: none
Change-Id: I4265cb1a4fd3355208f19aefdbb7abeb45b6cadf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/335700
Reviewed-by: Tomas Lundqvist <tomasl@google.com>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41599}
This reverts commit 79ac694d9b70fa9cd7b6a0f00bbee5d7fbbe64de.
Reason for revert: Breaks downstream (found another use of
ValueOrDefault instead of value_or)...
Original change's description:
> [Stats] Make RTCStatsMember<T> a type alias for absl::optional<T>.
>
> The moment we've all been waiting for.
>
> Step 1: Add type alias (this CL).
> Step 2: Migrate all uses of RTCStatsMember<T> to absl::optional<T>.
> Step 3: Delete type alias.
>
> Bug: webrtc:15164
> Change-Id: I00a7202c0b684fb2c57fcad4f501bccc167f1fa3
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/334680
> Commit-Queue: Henrik Boström <hbos@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Evan Shrubsole <eshr@google.com>
> Cr-Commit-Position: refs/heads/main@{#41593}
Bug: webrtc:15164
Change-Id: Ice3f44057b82a7ba9be000d9a0b714152fd07d2f
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/335701
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@google.com>
Reviewed-by: Christoffer Dewerin <jansson@google.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41595}
The moment we've all been waiting for.
Step 1: Add type alias (this CL).
Step 2: Migrate all uses of RTCStatsMember<T> to absl::optional<T>.
Step 3: Delete type alias.
Bug: webrtc:15164
Change-Id: I00a7202c0b684fb2c57fcad4f501bccc167f1fa3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/334680
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@google.com>
Cr-Commit-Position: refs/heads/main@{#41593}
Instead embed functionality of the rtc::TaskQueue into destructors and describe the potential race.
Bug: webrtc:14169
Change-Id: I01b570b530986a0d07798893057201493a8bef5f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/335141
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41592}
There can only be one!
Bug: webrtc:15164
Change-Id: Ib7265bf2103f24a6dab07737b2caed7f39ba75c3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/334643
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@google.com>
Cr-Commit-Position: refs/heads/main@{#41591}
This removes several references across the code base that depended on
the global string constants.
Bug: none
Change-Id: I007bd4b195c35261039f655f1a8f52e632c3691f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/335320
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41578}
rtc::TaskQueue is not used in any api and thus TaskQueueForTest no longer need to provide implementation for it.
TaskQueueForTest still helpful for code that expects TaskQueueBase.
Bug: webrtc:14169
Change-Id: I64ee2691ddeb0ff3a72cd4dd192730c06260f61f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/335323
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41577}
This is a reland of commit 54be7084e0861a0179a5fccd0b27edf7d7994bbb
Previously reverted due to an importer issue (b/320646178) and later a
dependency on RTCStatsMember<T>::ValueToString().
In this reland, we add Attribute::ToString() but we don't delete the
RTCStatsMember<T> stringifier methods, allowing downstream to migrate
before they are deleted.
Original change's description:
> [Stats] Attribute::ToString(), to replace member ValueToString/ToJson.
>
> Delete RTCStatsMember<T>::ValueToString() and ValueToJson() in favor of
> Attribute::ToString().
>
> The difference between "ToString" and "ToJson" is that the "ToJson"
> version converts 64-bit integers and doubles to floating points with no
> more than ~15 digits of precision as to not exceed JSON's precision
> limitations. So only in edge cases of really large numbers or numbers
> with a silly number of digits will the two methods produce different
> results. Also JSON puts '\"' around map key names, e.g. "{\"foo\":123}"
> as opposed to "{foo:123}".
>
> Going forward we see no reason to maintain two different string
> converted paths that are this similar, so we only implement one
> Attribute::ToString() method which does what "ToJson" did.
>
> In the next CL we can delete RTCStatsMember<T>.
>
> Bug: webrtc:15164
> Change-Id: Iaa8cf3bf14b40dc44664f75989832469603131c5
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/334640
> Commit-Queue: Henrik Boström <hbos@webrtc.org>
> Reviewed-by: Evan Shrubsole <eshr@google.com>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#41544}
Bug: webrtc:15164
Change-Id: I281ccf5b23d8f194b5ce00186a32846c757b46fc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/334860
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41575}
This is a step towards sending audio timestamps from Meet in iOS.
Next step is to enable sending the audio timestamps (in harmony).
After enable absolute-capture-time header extension in harmony, the receiving participants will be able to store E2E audio latency and A/V sync metrics.
Bug: webrtc:13609
Change-Id: I797c1ed0035625ed065307314ac34c932c5abe7e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/334720
Commit-Queue: Olov Brändström <brandstrom@google.com>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41574}
