117 Commits

Author SHA1 Message Date
solenberg
43e83d44f0 Revert of Implement AudioReceiveStream::GetStats(). (patchset #19 id:360001 of https://codereview.webrtc.org/1390753002/ )
Reason for revert:
webrtc_perf_tests started failing on Win32 Release, Mac32 Release and Linux64 Release (all running large tests). These were not caught by try bots.

Original issue's description:
> Implement AudioReceiveStream::GetStats().
>
> R=tommi@webrtc.org
> TBR=hta@webrtc.org
> BUG=webrtc:4690
>
> Committed: a457752f4a

TBR=tommi@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:4690

Review URL: https://codereview.webrtc.org/1411083006

Cr-Commit-Position: refs/heads/master@{#10340}
2015-10-20 13:41:06 +00:00
Fredrik Solenberg
a457752f4a Implement AudioReceiveStream::GetStats().
R=tommi@webrtc.org
TBR=hta@webrtc.org
BUG=webrtc:4690

Review URL: https://codereview.webrtc.org/1390753002 .

Cr-Commit-Position: refs/heads/master@{#10338}
2015-10-20 13:01:55 +00:00
solenberg
1ac561447e Remove default receive channel from WVoE; baby step 3.
Get rid of default receive channel.

BUG=webrtc:4690

Review URL: https://codereview.webrtc.org/1385893002

Cr-Commit-Position: refs/heads/master@{#10262}
2015-10-13 10:58:25 +00:00
solenberg
8fb30c328b Remove default receive channel from WVoE; baby step 2.
Rename voe_channel_ to default_send_channel_id_.

BUG=webrtc:4690

Review URL: https://codereview.webrtc.org/1388733002

Cr-Commit-Position: refs/heads/master@{#10261}
2015-10-13 10:07:07 +00:00
solenberg
4bac9c53da Change SetOutputScaling to set a single level, not left/right levels.
BUG=webrtc:4690

Review URL: https://codereview.webrtc.org/1397773002

Cr-Commit-Position: refs/heads/master@{#10234}
2015-10-09 09:32:58 +00:00
solenberg
0b67546d8c Remove default receive channel from WVoE; baby step 1.
Rx AGC config bits copied from https://codereview.webrtc.org/1315903004.

BUG=webrtc:4690

Review URL: https://codereview.webrtc.org/1388723002

Cr-Commit-Position: refs/heads/master@{#10233}
2015-10-09 08:37:17 +00:00
Peter Boström
0c4e06b4c6 Use suffixed {uint,int}{8,16,32,64}_t types.
Removes the use of uint8, etc. in favor of uint8_t.

BUG=webrtc:5024
R=henrik.lundin@webrtc.org, henrikg@webrtc.org, perkj@webrtc.org, solenberg@webrtc.org, stefan@webrtc.org, tina.legrand@webrtc.org

Review URL: https://codereview.webrtc.org/1362503003 .

Cr-Commit-Position: refs/heads/master@{#10196}
2015-10-07 10:23:32 +00:00
solenberg
d97ec30ce4 Remove default receive channel from WVoE; baby step 0.
Cleanup + add thread checker DCHECKs to various method in WebRtcVoiceEngine/MediaChannel.

BUG=webrtc:4690

Review URL: https://codereview.webrtc.org/1386653002

Cr-Commit-Position: refs/heads/master@{#10194}
2015-10-07 08:40:38 +00:00
solenberg
5b14b42e93 Remove unused SignalMediaError and infrastructure.
BUG=webrtc:4690

Review URL: https://codereview.webrtc.org/1362913004

Cr-Commit-Position: refs/heads/master@{#10133}
2015-10-01 11:10:40 +00:00
deadbeef
cbecd358e0 Reland of TransportController refactoring. (patchset #1 id:1 of https://codereview.webrtc.org/1358413003/ )
Reason for revert:
This CL just landed: https://codereview.chromium.org/1323243006/

Which fixes the FYI bots for the original CL, and breaks them for this revert.

Original issue's description:
> Revert of TransportController refactoring. (patchset #6 id:100001 of https://codereview.webrtc.org/1350523003/ )
>
> Reason for revert:
> This CL causes problems with the WebRTC-in-Chromium FYI bots. Presumably it needs to be done in several steps, where removed files are emptied instead of removed in the first step.
>
> Original issue's description:
> > TransportController refactoring.
> >
> > Getting rid of TransportProxy, and in its place adding a
> > TransportController class which will facilitate access to and manage
> > the lifetimes of Transports. These Transports will now be accessed
> > solely from the worker thread, simplifying their implementation.
> >
> > This refactoring also pulls Transport-related code out of BaseSession.
> > Which means that BaseChannels will now rely on the TransportController
> > interface to create channels, rather than BaseSession.
> >
> > Committed: https://crrev.com/47ee2f3b9f33e8938948c482c921d4e13a3acd83
> > Cr-Commit-Position: refs/heads/master@{#10022}
>
> TBR=pthatcher@webrtc.org,deadbeef@webrtc.org
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
>
> Committed: https://crrev.com/a81a42f584baa0d93a4b93da9632415e8922450c
> Cr-Commit-Position: refs/heads/master@{#10024}

TBR=pthatcher@webrtc.org,torbjorng@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true

Review URL: https://codereview.webrtc.org/1361773005

Cr-Commit-Position: refs/heads/master@{#10036}
2015-09-23 18:50:31 +00:00
Fredrik Solenberg
7d173362d0 Remove the [Un]RegisterVoiceProcessor() API.
BUG=webrtc:4690
R=pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/1361633002 .

Cr-Commit-Position: refs/heads/master@{#10027}
2015-09-23 10:23:31 +00:00
Fredrik Solenberg
09677342ae Remove VoEFile from VoeWrapper and the remaining places in libjingle where it was being used.
BUG=webrtc:4690
R=pbos@webrtc.org

Review URL: https://codereview.webrtc.org/1360773002 .

Cr-Commit-Position: refs/heads/master@{#10026}
2015-09-23 10:05:47 +00:00
torbjorng
a81a42f584 Revert of TransportController refactoring. (patchset #6 id:100001 of https://codereview.webrtc.org/1350523003/ )
Reason for revert:
This CL causes problems with the WebRTC-in-Chromium FYI bots. Presumably it needs to be done in several steps, where removed files are emptied instead of removed in the first step.

Original issue's description:
> TransportController refactoring.
>
> Getting rid of TransportProxy, and in its place adding a
> TransportController class which will facilitate access to and manage
> the lifetimes of Transports. These Transports will now be accessed
> solely from the worker thread, simplifying their implementation.
>
> This refactoring also pulls Transport-related code out of BaseSession.
> Which means that BaseChannels will now rely on the TransportController
> interface to create channels, rather than BaseSession.
>
> Committed: https://crrev.com/47ee2f3b9f33e8938948c482c921d4e13a3acd83
> Cr-Commit-Position: refs/heads/master@{#10022}

TBR=pthatcher@webrtc.org,deadbeef@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true

Review URL: https://codereview.webrtc.org/1358413003

Cr-Commit-Position: refs/heads/master@{#10024}
2015-09-23 09:24:27 +00:00
deadbeef
47ee2f3b9f TransportController refactoring.
Getting rid of TransportProxy, and in its place adding a
TransportController class which will facilitate access to and manage
the lifetimes of Transports. These Transports will now be accessed
solely from the worker thread, simplifying their implementation.

This refactoring also pulls Transport-related code out of BaseSession.
Which means that BaseChannels will now rely on the TransportController
interface to create channels, rather than BaseSession.

Review URL: https://codereview.webrtc.org/1350523003

Cr-Commit-Position: refs/heads/master@{#10022}
2015-09-22 22:08:31 +00:00
solenberg
c1a1b353ec Remove the SetLocalMonitor() API.
BUG=webrtc:4690

Review URL: https://codereview.webrtc.org/1344083004

Cr-Commit-Position: refs/heads/master@{#10020}
2015-09-22 20:31:28 +00:00
solenberg
22011c1b54 Remove Channel::SetRingbackTone() and Channel::PlayRingbackTone(), and the code beneath it (within libjingle).
BUG=webrtc:4690
TBR=juberti

Review URL: https://codereview.webrtc.org/1325023005

Cr-Commit-Position: refs/heads/master@{#10011}
2015-09-22 10:12:49 +00:00
Guo-wei Shieh
8902433a43 Revert "TransportController refactoring."
This reverts commit 9af63f473e1d0d6c47a741a046c41642dfc1c178.

Cr-Commit-Position: refs/heads/master@{#9994}
2015-09-18 20:50:31 +00:00
deadbeef
9af63f473e TransportController refactoring.
Getting rid of TransportProxy, and in its place adding a
TransportController class which will facilitate access to and manage
the lifetimes of Transports. These Transports will now be accessed
solely from the worker thread, simplifying their implementation.

This refactoring also pulls Transport-related code out of BaseSession.
Which means that BaseChannels will now rely on the TransportController
interface to create channels, rather than BaseSession.

This CL also adds some unit tests, and does some renaming.
For example, from "CandidateReady" to "CandidateGathered".

Review URL: https://codereview.webrtc.org/1246913005

Cr-Commit-Position: refs/heads/master@{#9993}
2015-09-18 19:56:02 +00:00
Fredrik Solenberg
b071a19019 Full use of NnChannel::SetSendParameters and NnChannel::SetRecvParameters.
SetOptions(), SetMaxBandwidth(), Set[Send|Recv]RtpHeaderExtensions(), Set[Send|Recv]Codecs() are now private.

BUG=webrtc:4690
R=pbos@webrtc.org, pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/1327933002 .

Cr-Commit-Position: refs/heads/master@{#9973}
2015-09-17 14:43:06 +00:00
henrikg
91d6edef35 Add RTC_ prefix to (D)CHECKs and related macros.
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}
2015-09-17 07:24:51 +00:00
Fredrik Solenberg
709ed67c38 Move instantiation of webrtc::Call into a MediaController class so that it can be used for both audio and video media channels.
I'm not super happy with the GetVoE() function added on MediaEngineInterface, but this will eventually be gone, once webrtc::Call owns the shared VoE state (or initially, maps ADM* to an implicitly created VoE).

BUG=webrtc:4690
R=pbos@webrtc.org, pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/1269863005 .

Cr-Commit-Position: refs/heads/master@{#9939}
2015-09-15 10:26:45 +00:00
solenberg
66f43392a3 Remove [Voice|Video]MediaChannel::GetOptions().
BUG=webrtc:4690

Review URL: https://codereview.webrtc.org/1324853003

Cr-Commit-Position: refs/heads/master@{#9904}
2015-09-09 08:36:31 +00:00
solenberg
bb741b3afa Remove GetOutputScaling from VoiceMediaChannel.
BUG=webrtc:4690

Review URL: https://codereview.webrtc.org/1331443003

Cr-Commit-Position: refs/heads/master@{#9870}
2015-09-07 10:56:45 +00:00
stefan
658910cc3c Revert "Speculative revert of "- Move test cases for more natural ordering.""
Did not resolve the build bot issue.

This reverts commit 02d283a6ff5364d94aa88f5f5df4cfd3a5411346.

BUG=
TBR=solenberg@webrtc.org

Review URL: https://codereview.webrtc.org/1324123002

Cr-Commit-Position: refs/heads/master@{#9849}
2015-09-03 12:48:36 +00:00
Stefan Holmer
02d283a6ff Speculative revert of "- Move test cases for more natural ordering."
This reverts commit c20a5dc9305b988ca173cd63e606124b02e6d54c.

BUG=webrtc:4959
R=solenberg@webrtc.org
TBR=solenberg@webrtc.org

Review URL: https://codereview.webrtc.org/1309313008 .

Cr-Commit-Position: refs/heads/master@{#9829}
2015-09-01 13:48:31 +00:00
Fredrik Solenberg
c20a5dc930 - Move test cases for more natural ordering.
- Get rid of the CoInitialize tests for WVoE/WViE.

BUG=webrtc:4690
R=pbos@webrtc.org, tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1319163002 .

Cr-Commit-Position: refs/heads/master@{#9817}
2015-08-31 09:14:05 +00:00
solenberg
4fbae2b791 Add send transports to individual webrtc::Call streams.
BUG=webrtc:4690

Review URL: https://codereview.webrtc.org/1273363005

Cr-Commit-Position: refs/heads/master@{#9807}
2015-08-28 11:07:15 +00:00
pbos
6bb1b6e7fe Control combined_audio_video_bwe with config bool.
Permits setting RTP extensions for AudioReceiveStream without enabling
combined A/V BWE. This prevents spamming the log with "Failed to find
extension id:".

BUG=webrtc:4870
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1256803004

Cr-Commit-Position: refs/heads/master@{#9633}
2015-07-24 14:10:25 +00:00
pbos
8fc7fa798f Base A/V synchronization on sync_labels.
Groups of streams that should be synchronized are signalled through
SDP. These should be used rather than synchronizing the first-added
video stream to the first-added audio stream implicitly.

BUG=webrtc:4667
R=hta@webrtc.org, solenberg@webrtc.org, stefan@webrtc.org
TBR=mflodman@webrtc.org

Review URL: https://codereview.webrtc.org/1181653002

Cr-Commit-Position: refs/heads/master@{#9586}
2015-07-15 15:03:04 +00:00
Henrik Lundin
441f634731 Re-land r9378 "Rename APM Config DelayCorrection to ExtendedFilter"
(This reverts commit 3fbf3f8841b5460503fb646eaedcb063620434a8.)

The original submission was reverted because it broke the Chrome build. This is fixed in patch set 2 of this change by keeping the old MediaConstraintsInterface string kExperimentalEchoCancellation. It will be removed once the Chrome code has been updated.

Original description:
"We use this Config struct for enabling/disabling Extended filter mode in AEC. This change renames it to ExtendedFilter for readability reasons. The corresponding media constraint is also renamed to kExtendedFilterEchoCancellation.

The old Config is kept in parallel with the new during a transition period. This is to avoid problems with API breakages. During this period, if any of the two Configs are enabled, the extended filter mode is engaged in APM. That is, the two Configs are combined with an "OR" operation.

This change also renames experimental_aec in AudioOptions to extended_filter_aec."

BUG=webrtc:4696
R=bjornv@webrtc.org, tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1151573021.

Cr-Commit-Position: refs/heads/master@{#9401}
2015-06-09 14:03:23 +00:00
Henrik Lundin
3fbf3f8841 Revert r9378 "Rename APM Config DelayCorrection to ExtendedFilter"
This reverts commit 5f4b7e2873864c61e2ad6d88679dcd5d321bfd16, since it
broke some of the build bots.

BUG=4696
TBR=bjornv@webrtc.org

Review URL: https://codereview.webrtc.org/1166463006

Cr-Commit-Position: refs/heads/master@{#9380}
2015-06-05 09:04:20 +00:00
Henrik Lundin
5f4b7e2873 Rename APM Config DelayCorrection to ExtendedFilter
We use this Config struct for enabling/disabling Extended filter mode
in AEC. This change renames it to ExtendedFilter for readability
reasons. The corresponding media constraint is also renamed to
kExtendedFilterEchoCancellation.

The old Config is kept in parallel with the new during a transition
period. This is to avoid problems with API breakages. During this
period, if any of the two Configs are enabled, the extended filter
mode is engaged in APM. That is, the two Configs are combined with an
"OR" operation.

This change also renames experimental_aec in AudioOptions to extended_filter_aec.

BUG=4696
R=bjornv@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/54659004

Cr-Commit-Position: refs/heads/master@{#9378}
2015-06-05 07:55:40 +00:00
Henrik Lundin
8e6fd46cc3 Route time-stretching metrics through libjingle
This change connects currentAccelerateRate and currentPreemptiveRate
in webrtc::NetworkStatistics, through corresponding variables in
VoiceReceiverInfo, to googAccelerateRate and googPreemptiveExpandRate.

R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/50179004

Cr-Commit-Position: refs/heads/master@{#9350}
2015-06-02 07:25:03 +00:00
Henrik Lundin
5263b3c1dd Add options for NetEq fast accelerate mode through libjingle
This CL connects RTCConfiguration::audioJitterBufferFastMode in
PeerConnection.java, through libjingle, down to
NetEq::Config::enable_fast_accelerate in native WebRTC.

When enabled, it will allow NetEq to do faster time-compression when
the buffer level is very high.

BUG=4691
R=henrika@webrtc.org, mflodman@webrtc.org, pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/55479004

Cr-Commit-Position: refs/heads/master@{#9344}
2015-06-01 08:29:55 +00:00
Jelena Marusic
c28a896a7b VoE: Initialize WebRtcVoiceMediaChannel with AudioOptions during creation
BUG=4690

Changes:
1. In MediaEngineInterface changed CreateChannel() to CreateChannel(const AudioOptions&). Plan is to eventually remove Get/SetAudioOptions and the cousins SetDelayOffset and SetDevices.
2. In ChannelManager changed CreateVoiceChannel(...) to CreateVoiceChannel(..., const AudioOptions&).
3. In ChannelManager removed SetEngineAudioOptions, because it is not used and we want to eventually remove SetAudioOptions.
4. Updated MediaEngineInterface implementations and unit tests accordingly.
5. In WebRtcVoiceEngine changed access of Set/ClearOptionOverrides to protected. These are only used by WebRtcVoiceMediaChannel (now a friend). Plan is to rethink the logic behind option overrides.
6. Cosmetics: replaced NULL with nullptr in touched code

R=solenberg@google.com, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/56499004

Cr-Commit-Position: refs/heads/master@{#9330}
2015-05-29 13:05:52 +00:00
Tommi
0eefb4d5c3 Detach base/logging.* from base/stream.*.
This is being done in preparation of moving base/logging.* to rtc_base_approved. base/stream.* has libjingle dependencies that webrtc can't use, so logging.* can't depend on streams. It does look like stream.* isn't used much, so cleaning that up as well as cleaning up usage of the actual stream support (now LogStream) in the logging code, is in order, but I'll leave that to another cl.

BUG=
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/54529004

Cr-Commit-Position: refs/heads/master@{#9269}
2015-05-23 07:54:19 +00:00
Fredrik Solenberg
ccb49e79fd Remove Soundclip handling from libjingle.
BUG=
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/51009004

Cr-Commit-Position: refs/heads/master@{#9216}
2015-05-19 09:37:39 +00:00
Minyue
2013aeced2 Propagating RTT from send-only channel to receive-only channel.
This is important for obtaining ntp time at receiver-only channel, which does not have RTT directly.

BUG=3978

TEST=chromium with hangout calls
R=henrika@webrtc.org, mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29989004

Cr-Commit-Position: refs/heads/master@{#9186}
2015-05-13 12:14:36 +00:00
Peter Boström
300eeb68f5 Remove VideoEngine interfaces.
Removes ViE interfaces, _impl.cc files, managers (such as
ViEChannelManager and ViEInputManager) as well as ViESharedData.

Interfaces necessary to implement observers have been moved to a
corresponding header (such as vie_channel.h).

BUG=1695, 4491
R=mflodman@webrtc.org, solenberg@webrtc.org
TBR=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/55379004

Cr-Commit-Position: refs/heads/master@{#9179}
2015-05-12 14:51:08 +00:00
Henrik Lundin
64dad838e6 Reland r9159 "Adding a new constraint to set NetEq buffer capacity ..."
The original change was reverted due to a breakage in the chrome build.
This change includes a fix for this.

TBR=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/49329004

Cr-Commit-Position: refs/heads/master@{#9169}
2015-05-11 10:44:20 +00:00
Henrik Lundin
1f629232d5 Revert r9164 "Adding a new constraint to set NetEq buffer capacity ..."
This reverts commit fd32f35aff8fc28ec084bddc274de284e0422a57.

TBR=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/55369004

Cr-Commit-Position: refs/heads/master@{#9165}
2015-05-10 09:06:20 +00:00
Henrik Lundin
fd32f35aff Reland r9159 "Adding a new constraint to set NetEq buffer capacity ..."
This reverts commit cdb47a4533b7b1e29e803ed6591a68bb1a4f1692.

Contains a tentative fix to the chrome build breakage caused by the
original change.

TBR=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/47139004

Cr-Commit-Position: refs/heads/master@{#9164}
2015-05-10 09:03:00 +00:00
Henrik Lundin
cdb47a4533 Revert r9159 "Adding a new constraint to set NetEq buffer capacity ..."
This reverts commit 208a2294cde839025318f1b3d57559cb0611a4e7.
Breaks the Chrome build.

TBR=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/53399004

Cr-Commit-Position: refs/heads/master@{#9161}
2015-05-08 12:03:46 +00:00
Henrik Lundin
208a2294cd Adding a new constraint to set NetEq buffer capacity from peerconnection
This change makes it possible to set a custom value for the maximum
capacity of the packet buffer in NetEq (the audio jitter buffer). The
default value is 50 packets, but any value can be set with the new
functionality.

R=jmarusic@webrtc.org, mflodman@webrtc.org, pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/50869004

Cr-Commit-Position: refs/heads/master@{#9159}
2015-05-08 10:58:51 +00:00
Fredrik Solenberg
d3ddc1b69e Consistently use DCHECK, not ASSERT or assert in talk/media/webrtc/.
BUG=
R=pbos@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/49929004

Cr-Commit-Position: refs/heads/master@{#9156}
2015-05-07 15:07:36 +00:00
Fredrik Solenberg
aaf8ff2e45 WebRtcVoiceEngine: virtual to override + git cl format.
BUG=
R=kwiberg@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/54369004

Cr-Commit-Position: refs/heads/master@{#9154}
2015-05-07 14:05:57 +00:00
Fredrik Solenberg
6179b89e53 Remove unused API on WebRtcVoiceEngine.
BUG=1695
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/46209004

Cr-Commit-Position: refs/heads/master@{#9153}
2015-05-07 14:01:30 +00:00
Fredrik Solenberg
4b60c73e74 Hook up libjingle WebRtcVoiceEngine to Call API for combined A/V BWE.
BUG=4574,3109
R=pbos@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/49269004

Cr-Commit-Position: refs/heads/master@{#9150}
2015-05-07 12:07:46 +00:00
Karl Wiberg
9478437fde rtc::Buffer improvements
1. Constructors, SetData(), and AppendData() now accept uint8_t*,
     int8_t*, and char*. Previously, they accepted void*, meaning that
     any kind of pointer was accepted. I think requiring an explicit
     cast in cases where the input array isn't already of a byte-sized
     type is a better compromise between convenience and safety.

  2. data() can now return a uint8_t* instead of a char*, which seems
     more appropriate for a byte array, and is harder to mix up with
     zero-terminated C strings. data<int8_t>() is also available so
     that callers that want that type instead won't have to cast, as
     is data<char>() (which remains the default until all existing
     callers have been fixed).

  3. Constructors, SetData(), and AppendData() now accept arrays
     natively, not just decayed to pointers. The advantage of this is
     that callers don't have to pass the size separately.

  4. There are new constructors that allow setting size and capacity
     without initializing the array. Previously, this had to be done
     separately after construction.

  5. Instead of TransferTo(), Buffer now supports swap(), and move
     construction and assignment, and has a Pass() method that works
     just like std::move(). (The Pass method is modeled after
     scoped_ptr::Pass().)

R=jmarusic@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/42989004

Cr-Commit-Position: refs/heads/master@{#9033}
2015-04-20 12:03:00 +00:00
Minyue Li
7100dcd317 Adding "usedtx" as Opus codec parameter.
This is according to https://tools.ietf.org/html/draft-spittka-payload-rtp-opus-03

Specifically,

usedtx: specifies if the decoder prefers the use of DTX. values are 1 and 0. If no value is specified, usedtx is assumed to be 0.

BUG=1014
R=juberti@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/48499004

Cr-Commit-Position: refs/heads/master@{#8872}
2015-03-27 04:06:35 +00:00