This CL removes all external access to the integer sample data in the
AudioBuffer class. It also removes the API in AudioBuffer that provides this.
The purpose of this is to pave the way for removing the sample
duplicating and implicit conversions between integer and floating point
sample formats which is done inside the AudioBuffer.
Bug: webrtc:10882
Change-Id: I1438b691bcef98278aef8e3c63624c367c2d12e9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149162
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28912}
This silences a warning that appeared with iOS 13, and is more efficient
in general.
Bug: webrtc:10866
Change-Id: I23db6b78af36e59b1d825d3f0cccc6008f9b626a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149808
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28911}
This is a reland of 7c6f74ab0344e9c6201de711d54026e9990b8e6c
Compared to the previous commit, new bits are added to log calls of
AddIceCandidate, and the gathering and reception of IPv6 candidates.
Original change's description:
> Set the usage pattern bits for adding remote ICE candidates from SDP.
>
> Currently these bits are only set when a remote ICE candidate is
> successfully added via addIceCandidate. For non-trickled sessions in
> which the remote candidates are added via the remote description, these
> bits are lost. This also happens for trickled sessions, though a rare
> case, when addIceCandidate does not succeed because the peer connection
> is not ready to add any remote candidate.
>
> Bug: webrtc:10868
> Change-Id: Ib2f199f9ffc936060473934d25ba397ef31131a3
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148880
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Qingsi Wang <qingsi@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28844}
Bug: webrtc:10868
Change-Id: Ifac0593dcfb64d88619fd24b4ab61c14a0810beb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149024
Commit-Queue: Qingsi Wang <qingsi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28904}
That method is going away in favor in construction time setting.
Bug: webrtc:10774
Change-Id: I2aba5a2537e5846a3c9438a5b376b230e84c5f32
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149826
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28901}
These tests are now run as part of PacingControllerUnittest instead.
Bug: webrtc:10809
Change-Id: If59e622e8a66565be678106d9341aa6eee78c299
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149803
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28900}
This CL changes the way that values are converted
between fixed and floating point to
-Avoid the former asymmetric conversion causing
nonlinear distortions.
-Reduce the complexity.
In contrast to the initial CL, the DCHECKS on the incoming sample
range was changed to limiting.
Bug: webrtc:6594
Change-Id: I8218dfd5c45388ad5aac61be453d2f28725a2475
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132783
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Original-Commit-Position: refs/heads/master@{#28867}
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149169
Cr-Commit-Position: refs/heads/master@{#28897}
This change fixes a bug where the initial delay could be set incorrectly.
Bug: webrtc:10896
Change-Id: I66b2234b69c46639488f4561e973384001230861
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149820
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28894}
For now there are a lot of logging from signaing phase and from WebRTC
internal components during the call. So this CL will add log entries
about starting or ending important phase of the test to easier determine
when what happend.
Bug: webrtc:10138
Change-Id: I4bf30d687be6ba830daff4c1d6f2e72afd5eb43d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149064
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28891}
The Pacer now just handles interaction with Module/ProcessThread and
forwarding packets to PacketRouter.
All other logic is moved to PacedSendingController, including tests.
PacedSender unittest are now just some basic sanity tests.
Bug: webrtc:10809
Change-Id: I69223cd9d8300997375b03706d2e99c88e46241c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149041
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28886}
This patch is a follow up to https://webrtc-review.googlesource.com/c/src/+/143177.
That patch modified the updating of CSRCS until "publishing" the frame
to the renderer, however the update was added to just after
calling renderer->OnFrame(video_frame).
This patch reverses the calls of renderer->OnFrame(video_frame)
and source_tracker_.OnFrameDelivered(video_frame.packet_infos())
so that the CSRCS are available when the frame is available.
This fixes the the flakes described in webrtc:10827 that has a
test that checks the CSRCs directly after a frame is available.
Note: an optimal/correct solution would be to update the renderer
and the source tracker in the same critical section so that they
would be available at the same time.
Bug: webrtc:10827
Change-Id: Ibf6efa832d8f2f2bcce0a9b0b946188bb67d48b1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149171
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Chen Xing <chxg@google.com>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28885}
Media transport (or, equivalently, datagram transport) may only be
created for data channels. In this case, it's not appropriate to
consider ICE not-yet-connected or failed due to the media transport's
state. If the media transport disconnects or fails, it will signal data
channels separately.
Bug: webrtc:9719
Change-Id: Ieb7cb307116e479d01616559d8bafdfc650a78c5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149420
Reviewed-by: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28884}
If a framerate reduction (input fps - restricted fps) is less than the
configured diff, shorten interval to next qp check.
Bug: none
Change-Id: Ia0b9e0638e5ba75cdc20a1bb45bfcb7d858c5f89
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149040
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28880}
This change adds the following helper functions to convert between "integer milliseconds"-style timestamps and durations, and "UQ32.32 and Q32.32"-style NTP timestamps and durations:
- Int64MsToQ32x32
- UInt64MsToUQ32x32
- Q32x32ToInt64Ms
- UQ32x32ToUInt64Ms
The Q-format NTP timestamps and durations are used by some RTP/RTCP packets.
Bug: webrtc:10739
Change-Id: I89123d2dba7370f26e239d722a4975bf5ac6e668
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148444
Commit-Queue: Chen Xing <chxg@google.com>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28878}
This only fixes the name string you get when you query the threads, the
functionality is not changes.
Bug: None
Change-Id: I29408cf38e0e41faa127a70a010d37a980bb24ce
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149167
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28875}
The public API to obtain the selected candidate pair is changed to
GetSelectedCandidatePair in the ICE transport, and the returned pair
has address-sanitized candidates.
Bug: chromium:993878
Change-Id: I44f9d2385a84f9e22447108be2e57ef9e62671eb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149080
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Qingsi Wang <qingsi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28869}
This CL changes the way that values are converted
between fixed and floating point to
-Avoid the former asymmetric conversion causing
nonlinear distortions.
-Reduce the complexity.
Bug: webrtc:6594
Change-Id: I64d0cc31c5d16f397686a59a062cfbc4b336d94d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132783
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28867}
This CL moves/removes all code from the AudioBuffer that:
-Is not directly handling audio data (e.g., keytaps, VAD descisions).
-Is caching aggregated versions of the rest of the audio data.
-Is not used (or only used in testing)
Bug: webrtc:10882
Change-Id: I737deb3f692748eff30f46ad806b2c6f6292802c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149072
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28866}
The "IsCurrent" check seems to have been missing from the class, but may help with
tracking down issue 10880. I also replaced the 'infinite' wait in SendTask with a
couple of timeouts, arbitrarily chosen 30 seconds for 'abandon wait' and 10
seconds for 'warning' log.
Change-Id: Ia40a68658dd007c60771135718511f7e4110c0b0
Bug: webrtc:10880
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149068
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28865}
Add possibility to configure min bitrate based on resolution.
Only adapt up if bw estimate is above the min bitrate for next higher resolution.
Bug: none
Change-Id: Ie38faae07d23336675ec33697ace6f6fed322efa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148598
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28863}
