Makes sense, since it not recording/playing to begin with.
BUG=b/35415663
Review-Url: https://codereview.webrtc.org/2783673002
Cr-Commit-Position: refs/heads/master@{#17423}
The class creates one WavReaderInterface object for each unique audiotrack and builds the set of speaker names.
Validating if the audiotrack lengths and the timing information are compatible (and hence valid) is not implemented yet.
MultiEndCall is designed using dependency injection. This allows to use mock objects with which we can quickly simulate different timings and track lengths without needing actual wav files.
BUG=webrtc:7218
Review-Url: https://codereview.webrtc.org/2761853002
Cr-Commit-Position: refs/heads/master@{#17421}
When screen/window capturers fail to use shared memory, they are
supposed to fall back to XGetImage(). It's slower, but should still
allow to capture desktop content. This wasn't implemented correclty in
XScreenPixelBuffer - it was failing completely when shmget() returns an
error.
BUG=705146
Review-Url: https://codereview.webrtc.org/2775793003
Cr-Commit-Position: refs/heads/master@{#17401}
Decouples encode flags and calculates them the same for both default and
screencast temporal layers.
With this change encoders could start using TemporalReferences for
temporal-layers flags, but they can not be used by asynchronous encoders
(hardware encoders) yet.
Also removes 'timestamp' as a dead parameter to FrameEncoded().
BUG=chromium:702017, webrtc:7349
R=marpan@google.com, sprang@webrtc.org, marpan@webrtc.org
Review-Url: https://codereview.webrtc.org/2769263002 .
Cr-Commit-Position: refs/heads/master@{#17397}
Implementation owned by call, and passed to VideoSendStream and
AudioSendStream.
BUG=webrtc:6847, webrtc:7135
Review-Url: https://codereview.webrtc.org/2685673003
Cr-Commit-Position: refs/heads/master@{#17389}
TWCC-PLR -based FecController doesn’t need smoothing; instead, use "null-smoothing", which just returns the last value (if any).
If we end up using TWCC-PLR, we'll just remove smoothing altogether. Until then, this is the least intrusive way to modify the code while still letting it work correctly for RTCP-PLR.
BUG=webrtc:7058
Review-Url: https://codereview.webrtc.org/2687433004
Cr-Commit-Position: refs/heads/master@{#17375}
Reason for revert:
Makes perf and Chromium FYI bots unhappy.
Original issue's description:
> WebRtcVoiceMediaChannel::AddRecvStream: Don't call SetRecPayloadType
>
> This removes one more place where we were unable to handle codecs not
> in the built-in set.
>
> BUG=webrtc:5805
>
> Review-Url: https://codereview.webrtc.org/2686043006
> Cr-Commit-Position: refs/heads/master@{#17370}
> Committed: 1724cfbdbaTBR=ossu@webrtc.org,solenberg@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5805
Review-Url: https://codereview.webrtc.org/2772043002
Cr-Commit-Position: refs/heads/master@{#17374}
1. Rename FecController to FecControllerPlrBased.
2. Introduce FecControllerRplrBased - a version of FecController that makes its decision based on RPLR instead of PLR.
BUG=webrtc:7058
Review-Url: https://codereview.webrtc.org/2672933003
Cr-Commit-Position: refs/heads/master@{#17373}
Otherwise cpplint will trigger during presubmit for unrelated changes
in these files.
BUG=webrtc:5149
NOTRY=True
Review-Url: https://codereview.webrtc.org/2767393003
Cr-Commit-Position: refs/heads/master@{#17371}
This removes one more place where we were unable to handle codecs not
in the built-in set.
BUG=webrtc:5805
Review-Url: https://codereview.webrtc.org/2686043006
Cr-Commit-Position: refs/heads/master@{#17370}
This is part of a series of CLs. Next CLs:
1. CL for RPLR-based FecController
2. CL for allowing experiment-driven configuration of the above (through both field-trials and protobuf)
BUG=webrtc:7058
Review-Url: https://codereview.webrtc.org/2661043003
Cr-Commit-Position: refs/heads/master@{#17368}
This CL is one in a series. To finish the work, the following CLs will be added:
1. CL for connecting RPLR as well
2. CL for RPLR-based FecController
3. CL for allowing experiment-driven configuration of the above (through both field-trials and protobuf)
BUG=webrtc:7058
Review-Url: https://codereview.webrtc.org/2638083002
Cr-Commit-Position: refs/heads/master@{#17365}
The ConversationalSpeechTest.* unit tests are now part of modules_unittests.
The rtc_test target has been replaced with an rtc_source_set one.
The latter is included as dependency in audio_processing_unittests.
BUG=webrtc:7218
Review-Url: https://codereview.webrtc.org/2769863005
Cr-Commit-Position: refs/heads/master@{#17360}
constexpr function should be preferred than a macro. So this change replaces
FOURCC() macro with a constexpr uint32_t FourCC() function.
BUG=679523, 650926
Review-Url: https://codereview.webrtc.org/2771573002
Cr-Commit-Position: refs/heads/master@{#17351}
Moves towards separating which layers may be referenced instead of
referencing libvpx flags directly. This will make strategies easier to
extract and usable from hardware encoders (RTCVideoEncoder, for
instance).
BUG=chromium:702017, webrtc:7349
R=brandtr@webrtc.org, marpan@webrtc.org. sprang@webrtc.org
Review-Url: https://codereview.webrtc.org/2747123005
Cr-Commit-Position: refs/heads/master@{#17349}
This change adds a DesktopCapturerId namespace, and attaches an int to each
DesktopFrame. ScreenCapturerWinGdi and ScreenCapturerWinDirectx now actively set
this field to differentiate themselves.
BUG=679523, 650926
Review-Url: https://codereview.webrtc.org/2759493002
Cr-Original-Commit-Position: refs/heads/master@{#17329}
Committed: 41e3d9ff3b
Review-Url: https://codereview.webrtc.org/2759493002
Cr-Commit-Position: refs/heads/master@{#17347}
The conversational_speech::Timing class models a list of turns.
Each turn, is identified by a speaker, the audiotrack name, and an offset in milliseconds.
The unit test checks that an instance of Timing is correctly populated and that save/reload leads to the same data.
BUG=webrtc:7218
Review-Url: https://codereview.webrtc.org/2750353002
Cr-Commit-Position: refs/heads/master@{#17346}
Reason for revert:
I suspect that this CL breaks Chromium WebRTC FYI bots. (Thanks kjellander@ for spotting.) The added dep in BUILD.gn would be the problem.
Example:
https://luci-logdog.appspot.com/v/?s=chromium%2Fbb%2Fchromium.webrtc.fyi%2FLinux_Builder%2F15058%2F%2B%2Frecipes%2Fsteps%2Fcompile%2F0%2Fstdout
FAILED: newlib_pnacl/obj/third_party/webrtc/api/libjingle_peerconnection_api/mediaconstraintsinterface.o
/b/c/goma_client/gomacc ../../native_client/toolchain/linux_x86/pnacl_newlib/bin/pnacl-clang++ -MMD -MF newlib_pnacl/obj/third_party/webrtc/api/libjingle_peerconnection_api/mediaconstraintsinterface.o.d -DNACL_TC_REV=5dfe030a71ca66e72c5719ef5034c2ed24706c43 -DV8_DEPRECATION_WARNINGS -DUSE_OPENSSL_CERTS=1 -DNO_TCMALLOC -DFULL_SAFE_BROWSING -DSAFE_BROWSING_CSD -DSAFE_BROWSING_DB_LOCAL -DCHROMIUM_BUILD -DENABLE_MEDIA_ROUTER=1 -DFIELDTRIAL_TESTING_ENABLED -D_FILE_OFFSET_BITS=64 -D_LARGEFILE_SOURCE -D_LARGEFILE64_SOURCE -D__STDC_CONSTANT_MACROS -D__STDC_FORMAT_MACROS -D_FORTIFY_SOURCE=2 -DNDEBUG -DNVALGRIND -DWEBRTC_RESTRICT_LOGGING -DEXPAT_RELATIVE_PATH -DHAVE_SCTP -DENABLE_EXTERNAL_AUTH -DHAVE_WEBRTC_VIDEO -DHAVE_WEBRTC_VOICE -DLOGGING_INSIDE_WEBRTC -DUSE_WEBRTC_DEV_BRANCH -DFEATURE_ENABLE_VOICEMAIL -DEXPAT_RELATIVE_PATH -DGTEST_RELATIVE_PATH -DNO_MAIN_THREAD_WRAPPING -DNO_SOUND_SYSTEM -DWEBRTC_CHROMIUM_BUILD -DWEBRTC_POSIX -I../.. -Inewlib_pnacl/gen -I../../third_party/webrtc_overrides -I../../third_party -fno-strict-aliasing -Wno-builtin-macro-redefined -D__DATE__= -D__TIME__= -D__TIMESTAMP__= -fcolor-diagnostics -Wall -Werror -Wextra -Wno-missing-field-initializers -Wno-unused-parameter -Wno-c++11-narrowing -Wno-covered-switch-default -Wno-deprecated-register -Wno-unneeded-internal-declaration -Wno-inconsistent-missing-override -O2 -fno-ident -fdata-sections -ffunction-sections -g0 -fvisibility=hidden -fvisibility-inlines-hidden -std=gnu++11 -fno-rtti -fno-exceptions -c ../../third_party/webrtc/api/mediaconstraintsinterface.cc -o newlib_pnacl/obj/third_party/webrtc/api/libjingle_peerconnection_api/mediaconstraintsinterface.o
In file included from ../../third_party/webrtc/api/mediaconstraintsinterface.cc:11:
In file included from ../../third_party/webrtc/api/mediaconstraintsinterface.h:27:
In file included from ../../third_party/webrtc/api/peerconnectioninterface.h:77:
In file included from ../../third_party/webrtc/api/dtmfsenderinterface.h:16:
In file included from ../../third_party/webrtc/api/mediastreaminterface.h:33:
In file included from ../../third_party/webrtc/media/base/mediachannel.h:28:
../../third_party/webrtc/base/socket.h:18:10: fatal error: 'sys/socket.h' file not found
#include <sys/socket.h>
^
1 error generated.
Original issue's description:
> Add DesktopCapturerId and attach it to DesktopFrame
>
> This change adds a DesktopCapturerId namespace, and attaches an int to each
> DesktopFrame. ScreenCapturerWinGdi and ScreenCapturerWinDirectx now actively set
> this field to differentiate themselves.
>
> BUG=679523, 650926
>
> Review-Url: https://codereview.webrtc.org/2759493002
> Cr-Commit-Position: refs/heads/master@{#17329}
> Committed: 41e3d9ff3bTBR=sergeyu@chromium.org,zijiehe@chromium.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=679523, 650926
Review-Url: https://codereview.webrtc.org/2767003002
Cr-Commit-Position: refs/heads/master@{#17336}
This change adds a DesktopCapturerId namespace, and attaches an int to each
DesktopFrame. ScreenCapturerWinGdi and ScreenCapturerWinDirectx now actively set
this field to differentiate themselves.
BUG=679523, 650926
Review-Url: https://codereview.webrtc.org/2759493002
Cr-Commit-Position: refs/heads/master@{#17329}
New class ReceiveSideCongestionController, extracted from CongestionController, and responsible for the
OnReceivedPacket processing.
Rest of the CongestionController moved to a new class
SendSideCongestionController.
To avoid breaking applications, CongestionController is redefined
as a union of these two classes, with no intended change in behavior.
With one exception: CongestionController::SetBweBitrates used to call
remote_bitrate_estimator_.SetMinBitrate, but after remote_bitrate_estimator_ was moved to ReceiveSideCongestionController,
it no longer does this.
BUG=webrtc:6847
Review-Url: https://codereview.webrtc.org/2752233002
Cr-Commit-Position: refs/heads/master@{#17321}
The only thing that was holding us back was the indeterministic teardown of voe::Channel(), but it turned out that fixing it wasn't that hard :)
BUG=webrtc:4508
Review-Url: https://codereview.webrtc.org/2755273004
Cr-Commit-Position: refs/heads/master@{#17315}
It depends on RTCP RPSI and SLI messages, which are being deleted.
TBR=stefan@webrtc.org # TODO comments added to common_types.h
BUG=webrtc:7338
Review-Url: https://codereview.webrtc.org/2753783002
Cr-Commit-Position: refs/heads/master@{#17314}
This is one step towards separation of send-side and receive-side
processing.
BUG=webrtc:7135
Review-Url: https://codereview.webrtc.org/2740163002
Cr-Commit-Position: refs/heads/master@{#17306}
