Commit Graph

666 Commits

Author SHA1 Message Date
34cdd2d402 Remove AudioDeviceObserver and make ADM not inherit from the Module interface.
(Re-upload of https://codereview.webrtc.org/3020493002/)

Bug: webrtc:4690, webrtc:7306
Change-Id: I67fb9ebca1296aabc08eae8a292a5c69832dc35e
Reviewed-on: https://webrtc-review.googlesource.com/5360
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20083}
2017-10-02 15:01:20 +00:00
b0a0207838 Added RTCMediaStreamTrackStats.jitterBufferDelay for audio
Description of this stat can be found here:
https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats-jitterbufferdelay

Bug: webrtc:8281
Change-Id: Ib2e8174f3449e68ad419ae2d58d5565fc9854023
Reviewed-on: https://webrtc-review.googlesource.com/3381
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20069}
2017-10-02 10:47:00 +00:00
440216fcf3 Split LogRtpHeader and LogRtcpPacket into separate versions for incoming and outgoing packets.
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}
2017-10-02 08:44:20 +00:00
310273459d Revert "Reland of Add full stack tests for MediaCodec encoder (moved from Rietveld)."
This reverts commit 2666cf7eba4bdd697d59d0451a8f74a05d4d207e.

Reason for revert: On Lollipop Nexus 4, the 240p tests fail too.

Original change's description:
> Reland of Add full stack tests for MediaCodec encoder (moved from Rietveld).
> 
> * Add audio_ prefix to CallTest::{en,de}coder_factory_.
> * Let VideoQualityTest only instantiate encoders using encoder factories.
> * Add HW encoder factories to VideoQualityTest.
> * Add full stack tests:
>   - sqcif7 at  30 kbps: libvpx.
>   - 240p10 at 100 kbps: MediaCodec, libvpx, and MediaCodec+libvpx.
> 
> BUG=webrtc:8219
> TBR=asapersson@webrtc.org,kjellander@webrtc.org,stefan@webrtc.org,sprang@webrtc.org
> 
> Change-Id: I464409ac0d5276defa78c1bf66034c6cca717d74
> Reviewed-on: https://webrtc-review.googlesource.com/4740
> Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20041}

TBR=kjellander@webrtc.org,brandtr@webrtc.org,asapersson@webrtc.org,sprang@webrtc.org,stefan@webrtc.org

Change-Id: If558b7fb86740658e50a6897d1eeeb72103a54ec
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8219
Reviewed-on: https://webrtc-review.googlesource.com/4900
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20044}
2017-09-29 13:48:29 +00:00
2666cf7eba Reland of Add full stack tests for MediaCodec encoder (moved from Rietveld).
* Add audio_ prefix to CallTest::{en,de}coder_factory_.
* Let VideoQualityTest only instantiate encoders using encoder factories.
* Add HW encoder factories to VideoQualityTest.
* Add full stack tests:
  - sqcif7 at  30 kbps: libvpx.
  - 240p10 at 100 kbps: MediaCodec, libvpx, and MediaCodec+libvpx.

BUG=webrtc:8219
TBR=asapersson@webrtc.org,kjellander@webrtc.org,stefan@webrtc.org,sprang@webrtc.org

Change-Id: I464409ac0d5276defa78c1bf66034c6cca717d74
Reviewed-on: https://webrtc-review.googlesource.com/4740
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20041}
2017-09-29 12:54:17 +00:00
5aea38c8be Disabling CallPerfTest.{CaptureNtpTimeWithNetworkDelay,CaptureNtpTimeWithNetworkJitter}.
Tests disabled for Mac only.

Tests fail in this way on perf bot Mac 10.11:

[ RUN      ] CallPerfTest.CaptureNtpTimeWithNetworkDelay
../../call/call_perf_tests.cc:407: Failure
Value of: std::abs(time_offset_ms) < threshold_ms_
  Actual: false
Expected: true



TBR=stefan@webrtc.org

Bug: webrtc:8291
Change-Id: I8d173fcff21f096827f31ddd670b6647796bff4b
Reviewed-on: https://webrtc-review.googlesource.com/4041
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19995}
2017-09-27 11:45:46 +00:00
06319b7830 Disable RampUpTest.UpDownUpTransportSequenceNumberPacketLoss on Mac.
Test is flaky.

Failures look like this:

../../call/rampup_tests.cc:379: Failure
Value of: Wait()
  Actual: false
Expected: true

TBR=stefan@webrtc.org
NOTRY=True

Bug: webrtc:7919
Change-Id: I99d468e2af49baf2bd6f6c6aee2c18f99c24bac7
Reviewed-on: https://webrtc-review.googlesource.com/3980
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19991}
2017-09-27 10:01:36 +00:00
1405afe780 Disable RampUpTest.UpDownUpTransportSequenceNumberPacketLoss on Linux due to flakiness.
BUG=webrtc:7919

Review-Url: https://codereview.webrtc.org/3013853002
Cr-Commit-Position: refs/heads/master@{#19978}
2017-09-26 19:11:38 +00:00
3b3622fafc Delete member VideoReceiveStream::Config::Rtp::ulpfec.
Replaced with scalars ulpfec_payload_type and red_payload_type.

In particular, ulpfec.red_rtx_payload_type, which duplicated info in
rtx_associated_payload_types, is deleted. This is a followup to cl
https://codereview.webrtc.org/3012963002.

BUG=webrtc:7135

Review-Url: https://codereview.webrtc.org/3019453002
Cr-Commit-Position: refs/heads/master@{#19965}
2017-09-26 09:49:21 +00:00
2c30120fac Revert of Add full stack tests for MediaCodec. (patchset #10 id:180001 of https://codereview.webrtc.org/3005253002/ )
Reason for revert:
Breaks KitKat/Lollipop perf bots.

Original issue's description:
> Add full stack tests for MediaCodec encoder.
>
> * Add audio_ prefix to CallTest::{en,de}coder_factory_.
> * Let VideoQualityTest only instantiate encoders using encoder factories.
> * Add HW encoder factories to VideoQualityTest.
> * Add full stack tests:
>   - sqcif7 at  30 kbps: MediaCodec and libvpx.
>   - 240p10 at 100 kbps: MediaCodec, libvpx, and MediaCodec+libvpx.
>
> BUG=webrtc:8219
>
> Review-Url: https://codereview.webrtc.org/3005253002
> Cr-Commit-Position: refs/heads/master@{#19923}
> Committed: https://webrtc.googlesource.com/src/+/2cefac6c1685abfcd7b90fdef8e926f1c2b79bfa

TBR=sprang@webrtc.org,asapersson@webrtc.org,kjellander@webrtc.org,stefan@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:8219

Review-Url: https://codereview.webrtc.org/3016593002
Cr-Commit-Position: refs/heads/master@{#19926}
2017-09-22 11:30:08 +00:00
2cefac6c16 Add full stack tests for MediaCodec encoder.
* Add audio_ prefix to CallTest::{en,de}coder_factory_.
* Let VideoQualityTest only instantiate encoders using encoder factories.
* Add HW encoder factories to VideoQualityTest.
* Add full stack tests:
  - sqcif7 at  30 kbps: MediaCodec and libvpx.
  - 240p10 at 100 kbps: MediaCodec, libvpx, and MediaCodec+libvpx.

BUG=webrtc:8219

Review-Url: https://codereview.webrtc.org/3005253002
Cr-Commit-Position: refs/heads/master@{#19923}
2017-09-22 07:46:25 +00:00
99a81b613d Remove #include of rtc_stream_config.h from rtc_event_log.h
StreamConfig is not integral to RTC-event logging in general, but rather to specific events. Therefore, the dependency on it should not be exported through rtc_event_log.h.

BUG=webrtc:8111
TBR=stefan@webrtc.org

Change-Id: I1ece0830cd05fd12220c8c717490e15942bacec9
Reviewed-on: https://webrtc-review.googlesource.com/1238
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19911}
2017-09-21 09:05:54 +00:00
9a2e906b0c Added RTCMediaStreamTrackStats.concealmentEvents
The number of concealment events. This counter increases every time a concealed sample is
synthesized after a non-concealed sample. That is, multiple consecutive concealed samples
will increase the concealedSamples count multiple times but is a single concealment event.

Bug: webrtc:8246
Change-Id: I7ef404edab765218b1f11e3128072c5391e83049
Reviewed-on: https://webrtc-review.googlesource.com/1221
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19881}
2017-09-18 08:58:06 +00:00
7120742701 Adding NOLINT for typedefs.h and common_types.h
Now that we have moved WebRTC from src/webrtc to src/, common_types.h
and typedefs.h are triggering a cpplint error.

The cpplint complaint is:
Include the directory when naming .h files  [build/include] [4]

This CL disables the error but we have to remove these two headers
from the root directory.

NOPRESUBMIT=true

Bug: webrtc:5876
Change-Id: I08e1b69aadcc4b28ab83bf25e3819d135d41d333
Reviewed-on: https://webrtc-review.googlesource.com/1577
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@google.com>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19859}
2017-09-15 13:03:51 +00:00
92ea95e34a Fixing WebRTC after moving from src/webrtc to src/
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}
2017-09-15 05:02:56 +00:00
bb547203bf Moving src/webrtc into src/.
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