This CL contains three changes as a preparation for adding audio send streams
to the send-side BWE:
1. Audio packets are passed through the pacer with high priority. This
is needed to be able to set transport sequence numbers on the packets.
2. A feedback observer is passed to the audio stream's rtcp receiver so
that the BWE can get notified of any BWE feedback being received on the
audio feedback channel.
3. Support for the transport sequence number header extension is added
to audio send streams.
BUG=webrtc:5263,webrtc:5307
R=mflodman@webrtc.org, solenberg@webrtc.org
Review URL: https://codereview.webrtc.org/1479023002 .
Cr-Commit-Position: refs/heads/master@{#10909}
- Flatten logic and make the relevant calls on VoE::Channel from AudioSendStream::SendTelephoneEvent().
- Store current payload type for telephone events in WVoMC, instead of setting it on the Channel. This should be refactored to be an AudioSendStream::Config parameter when we redo WVoMC::SetSendCodecs().
BUG=webrtc:4690
R=pthatcher@webrtc.org, tina.legrand@webrtc.org
Review URL: https://codereview.webrtc.org/1491743004 .
Cr-Commit-Position: refs/heads/master@{#10895}
* Move PlatformThread to rtc::.
* Remove ::CreateThread factory method.
* Make non-scoped_ptr from a lot of invocations.
* Make Start/Stop void.
* Remove rtc::Thread priorities, which were unused and would collide.
* Add ::IsRunning() to PlatformThread.
BUG=
R=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1476453002 .
Cr-Commit-Position: refs/heads/master@{#10812}
This is the last piece of the old directory layout of the modules.
Duplicated header files are left in audio_coding/main/include until
downstream code is updated to the new location. They have pragma
warnings added to them and identical header guards as the new headers to avoid breaking things.
BUG=webrtc:5095
TESTED=Passing compile-trybots with --clobber flag:
git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel --bot=linux_gn_rel --bot=win_x64_gn_rel --bot=mac_x64_gn_rel --bot=android_gn_rel -m tryserver.webrtc
NOTRY=True
NOPRESUBMIT=True
Review URL: https://codereview.webrtc.org/1481493004
Cr-Commit-Position: refs/heads/master@{#10803}
This will allow Audio[Send|Receive]Stream to bypass the VoE interfaces in many cases and talk directly to the channel.
BUG=webrtc:4690
Review URL: https://codereview.webrtc.org/1459083007
Cr-Commit-Position: refs/heads/master@{#10788}
Also removes all virtual methods. Permits using a thread from
rtc_base_approved (namely event tracing).
BUG=webrtc:5158
R=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1469013002
Cr-Commit-Position: refs/heads/master@{#10760}
This will make it possible to remove the build_with_libjingle=1 and key=''
GYP_DEFINES the bots are using (https://codereview.chromium.org/1450313002/).
It will also pave the road for enabling more WebRTC native tests on iOS.
BUG=webrtc:4755,webrtc:3185,webrtc:5165
TESTED=git cl try -c --bot=android_compile_rel --bot=linux_compile_rel --bot=win_compile_rel --bot=mac_compile_rel --bot=ios_rel --bot=linux_gn_rel --bot=win_x64_gn_rel --bot=mac_x64_gn_rel --bot=android_gn_rel -m tryserver.webrtc
Local compilation with:
GYP_DEFINES='OS=ios target_arch=arm' webrtc/build/gyp_webrtc
ninja -C out/Release-iphoneos
GYP_DEFINES='OS=ios target_arch=arm chromium_ios_signing=0' webrtc/build/gyp_webrtc
ninja -C out/Release-iphoneos
GYP_DEFINES='OS=ios target_arch=arm64' webrtc/build/gyp_webrtc
ninja -C out/Release-iphoneos
GYP_DEFINES='OS=ios target_arch=ia32' webrtc/build/gyp_webrtc
ninja -C out/Release-iphonesimulator
GYP_DEFINES='OS=ios target_arch=x64' webrtc/build/gyp_webrtc
ninja -C out/Release-iphonesimulator
R=henrika@webrtc.org
Review URL: https://codereview.webrtc.org/1457053003 .
Cr-Commit-Position: refs/heads/master@{#10711}
Opus may have an internal error that causes this. Here we make a workaround by adding some small disturbance to the input signals to break a long sequence of zeros.
BUG=webrtc:5127
Review URL: https://codereview.webrtc.org/1415173005
Cr-Commit-Position: refs/heads/master@{#10565}
Trace checks in the ctor and dtor of VoETestManager are removed, since they can fail if there are more than one VoETestManager (or VoE) used in a test.
BUG=
Review URL: https://codereview.webrtc.org/1407883007
Cr-Commit-Position: refs/heads/master@{#10507}
This changes the following module directories:
* webrtc/modules/audio_conference_mixer/interface
* webrtc/modules/interface
* webrtc/modules/media_file/interface
* webrtc/modules/rtp_rtcp/interface
* webrtc/modules/utility/interface
To avoid breaking downstream, I followed this recipe:
1. Copy the interface dir to a new sibling directory: include
2. Update the header guards in the include directory to match the style guide.
3. Update the header guards in the interface directory to match the ones in include. This is required to avoid getting redefinitions in the not-yet-updated downstream code.
4. Add a pragma warning in the header files in the interface dir. Example:
#pragma message("WARNING: webrtc/modules/interface is DEPRECATED; "
"use webrtc/modules/include")
5. Search for all source references to webrtc/modules/interface and update them to webrtc/modules/include (*.c*,*.h,*.mm,*.S)
6. Update all GYP+GN files. This required manual inspection since many subdirectories of webrtc/modules referenced the interface dir using ../interface etc(*.gyp*,*.gn*)
BUG=5095
TESTED=Passing compile-trybots with --clobber flag:
git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel -m tryserver.webrtc
R=stefan@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1417683006 .
Cr-Commit-Position: refs/heads/master@{#10500}
The end goal is to remove AcmReceiver::SetInitialDelay. This change is
in preparation for that goal. It turns out that
AcmReceiver::SetInitialDelay was only invoked through the following
call chain, where each method in the chain is never referenced from
anywhere else (except from tests in some cases):
ViEChannel::SetReceiverBufferingMode
-> ViESyncModule::SetTargetBufferingDelay
-> VoEVideoSync::SetInitialPlayoutDelay
-> Channel::SetInitialPlayoutDelay
-> AudioCodingModule::SetInitialPlayoutDelay
-> AcmReceiver::SetInitialDelay
The start of the chain, ViEChannel::SetReceiverBufferingMode was never
referenced.
This change deletes all the methods above except
AcmReceiver::SetInitialDelay itself, which will be handled in a
follow-up change.
BUG=webrtc:3520
Review URL: https://codereview.webrtc.org/1421013006
Cr-Commit-Position: refs/heads/master@{#10471}
Also updated the RTPPayloadRegistry::RestoreOriginalPacket signature to not take the first arg as a **, since it isn't modified.
Review URL: https://codereview.webrtc.org/1394573004
Cr-Commit-Position: refs/heads/master@{#10276}
This allows us to pass packet meta data, such as transport sequence
number, to libjingle and further down to the socket implementation. A
similar struct already exist in libjingle, see rtc::PacketOptions in asyncpacketsocket.h.
BUG=4173
Review URL: https://codereview.webrtc.org/1376673004
Cr-Commit-Position: refs/heads/master@{#10144}
Removes ShouldIgnoreTrace from WebRtcVoiceEngine and removes the spammy
log instances instead. Also removes trace-style logging from getters
(::GetLocalSSRC() for instance would print what SSRC it got, spamming
the log).
BUG=
R=henrika@webrtc.org
Review URL: https://codereview.webrtc.org/1347353004 .
Cr-Commit-Position: refs/heads/master@{#10028}
The functions were essentially no-op. Also removing forward declaration
of ACMDTMFDetection, which was not used.
BUG=3520
Review URL: https://codereview.webrtc.org/1356543003
Cr-Commit-Position: refs/heads/master@{#9982}
Starts by removing channel/engine id from ViEChannel which propagates
down to the RTP/RTCP module as well as the transport class.
IncomingVideoStream::RenderFrame() is untouched for now but receives a
fake id instead of the previous channel id. Added a TODO to remove it
later but the RenderFrame call is implemented in a lot of
platform-dependent files and should probably remove the "manager" aspect
of renderers, so preferring to do it separately
BUG=webrtc:1695
R=henrika@webrtc.org, mflodman@webrtc.org
Review URL: https://codereview.webrtc.org/1335353005 .
Cr-Commit-Position: refs/heads/master@{#9978}
WebRtcPassthroughRender has been dead since webrtcvideoengine.cc was
removed, FakeExternalTransport has probably been unused for a long time.
BUG=webrtc:1695
R=henrika@webrtc.org
Review URL: https://codereview.webrtc.org/1343393003 .
Cr-Commit-Position: refs/heads/master@{#9968}
We must remove dependency on Chromium, i.e. we can't use Chromium's base/logging.h. That means we need to define these macros in WebRTC also when doing Chromium builds. And this causes redefinition.
Alternative solutions:
* Check if we already have defined e.g. CHECK, and don't define them in that case. This makes us depend on include order in Chromium, which is not acceptable.
* Don't allow using the macros in WebRTC headers. Error prone since if someone adds it there by mistake it may compile fine, but later break if a header in added or order is changed in Chromium. That will be confusing and hard to enforce.
* Ensure that headers that are included by an embedder don't include our macros. This would require some heavy refactoring to be maintainable and enforcable.
* Changes in Chromium for this is obviously not an option.
BUG=chromium:468375
NOTRY=true
Review URL: https://codereview.webrtc.org/1335923002
Cr-Commit-Position: refs/heads/master@{#9964}
We must remove dependency on Chromium, i.e. we can't use Chromium's base/logging.h. That means we need to define these macros in WebRTC also when doing Chromium builds. And this causes redefinition.
* DISALLOW_ASSIGN -> RTC_DISALLOW_ASSIGN
* DISALLOW_COPY_AND_ASSIGN -> RTC_DISALLOW_COPY_AND_ASSIGN
* DISALLOW_IMPLICIT_CONSTRUCTORS -> RTC_DISALLOW_IMPLICIT_CONSTRUCTORS
Related CL: https://codereview.webrtc.org/1335923002/
BUG=chromium:468375
NOTRY=true
Review URL: https://codereview.webrtc.org/1345433002
Cr-Commit-Position: refs/heads/master@{#9953}
Helps differentiate between different instances when debugging.
Review URL: https://codereview.webrtc.org/1337003003
Cr-Commit-Position: refs/heads/master@{#9927}
An option was added to voe_cmd_test to make a RtcEventLog dump.
BUG=webrtc:4741
Review URL: https://codereview.webrtc.org/1267683002
Cr-Commit-Position: refs/heads/master@{#9901}
