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10 Commits

Author SHA1 Message Date
Rasmus Brandt
48912451d4 Delete modules/video_processing 
Reasons:
1) It is not used by `PeerConnection` (only in tests)
2) We have no plans on using it
3) The code is functionally untouched since many years

Bug: b/249972434
Change-Id: I1d30edd34231f25d86e8495ff71f1786ba2b0a1c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/277445
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38260}
 2022-09-30 13:50:49 +00:00
Niels Möller
8581877121 Delete interface class VideoCaptureExternal 
Also delete corresponding and unused create method
VideoCaptureFactory::Create(VideoCaptureExternal...),
the code under modules/video_capture/external, and the
build target modules/video_capture:video_capture.

Bug: None
Change-Id: I5ec6139e9ecf460f93ede847868f7f80dbc019f5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131385
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27506}
 2019-04-09 08:18:20 +00:00
henrika
255750bfb0 Adds support for real audio devices in video_quality_test. 
The old test supported audio but only in combination with a fake ADM.
The new version allows the user to run real video and audio.

Now possible to do:

./out/Debug/video_loopback.exe --audio --use_real_adm

To run the test in loopback using real default audio devices.

By default:

./out/Debug/video_loopback.exe --audio

runs with fake audio devices as before.

Bug: webrtc:9265
Change-Id: Id89924ec0276f929487c71fc6321dcd9cb92693d
Reviewed-on: https://webrtc-review.googlesource.com/96161
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24463}
 2018-08-28 09:00:45 +00:00
Fredrik Solenberg
a8b7c7f4c6 Move remaining traces of VoiceEngine 
- Move files from voice_engine/ to audio/.
- Rename voice_engine/utility.* to remix_resample.* since there are no other
  utilities in those files.
- Move test/mock_voe_channel_proxy.h to audio/.
- Removed voe_channel_id from Audio[Receive|Send]Stream::Config.
- Remove VoiceEngine* from AudioState::Config.
- Fix a few cpplint complaints which showed when moving files.

NOPRESUBMIT=true

Bug: webrtc:4690
Change-Id: Id266c822d956625c358fa5e193e6f4837164aef8
Reviewed-on: https://webrtc-review.googlesource.com/39268
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21657}
 2018-01-17 13:27:47 +00:00
Rasmus Brandt
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
Rasmus Brandt
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
brandtr
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
brandtr
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
Mirko Bonadei
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
Mirko Bonadei
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