Moving functionality to get max payload length from default RTP module
to the payload router.
I'll make a follow up CL changing asserts to DCHECK in rtp_rtcp_impl.cc.
BUG=769
TEST=New unittest and existing sender mtu test
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36119004
Cr-Commit-Position: refs/heads/master@{#8345}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8345 4adac7df-926f-26a2-2b94-8c16560cd09d
This change switches from the old codec wrapper ACMISAC to the new
AudioEncoderIsac wrapped in an ACMGenericCodecWrapper.
This is also the CL where the old codec for producing redundancy (RED)
is inactivated. All RED payloads are now produces through the
AudioEncoderCopyRed or AudioEncoderIsacRed classes.
BUG=4228
TEST=Please, try the iSAC codec extensively.
COAUTHOR=kwiberg@webrtc.orgR=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/33249005
Cr-Commit-Position: refs/heads/master@{#8342}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8342 4adac7df-926f-26a2-2b94-8c16560cd09d
ViECapturer is always calling DeliverFrame with an empty CSRC vector, so
this is basically a dead path already. I added a DCHECK in ViEEncoder to
verify this is true.
BUG=769
TEST=Manually verified in Chromium.
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/39059004
Cr-Commit-Position: refs/heads/master@{#8335}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8335 4adac7df-926f-26a2-2b94-8c16560cd09d
This is a big refactoring of the existing C++/JNI/Java support for audio recording in native WebRTC:
- Removes unused code and old WEBRTC logging macros
- Now uses optimal sample rate and buffer size in Java AudioRecord (used hard-coded sample rate before)
- Makes code more inline with the implementation in Chrome
- Adds helper methods for JNI handling to improve readability
- Changes the threading model (high-prio audio thread now lives in Java-land and C++ only works as proxy)
- Adds basic thread checks
- Removes all locks in C++ land
- Removes all locks in Java
- Improves construction/destruction
- Additional cleanup
Tested using AppRTCDemo and WebRTCDemo APKs on N6, N5, N7, Samsung Galaxy S4 and
Samsung Galaxy S4 mini (which uses 44.1kHz as native sample rate).
BUG=NONE
R=magjed@webrtc.org, perkj@webrtc.org, pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/33969004
Cr-Commit-Position: refs/heads/master@{#8325}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8325 4adac7df-926f-26a2-2b94-8c16560cd09d
This CL performs the following renames of targets to
make GYP and GN more unified and make the targets that
have the same name as the module and include the external
render/capture implementation (the internal one is only
used by WebRTC tests).
This makes it natural to declare dependencies in GN
without having to specify the target.
Summary of the renames:
GYP:
video_render_module_impl -> video_render (new target)
video_capture_module_impl -> video_capture (new target)
GN:
video_capture -> video_capture_module (now identical to the GYP target)
video_capture_impl -> video_capture
video_render -> video_render_module (now identical to the GYP target)
video_render_impl -> video_render
BUG=456815
R=andresp@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/35099004
Cr-Commit-Position: refs/heads/master@{#8323}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8323 4adac7df-926f-26a2-2b94-8c16560cd09d
Now the ChannelBuffer has 2 separate arrays, one for the full-band data and one for the splitted one. The corresponding accessors are added to the ChannelBuffer.
This is done to avoid having to refresh the bands pointers in AudioBuffer. It will also allow us to have a general accessor like data()[band][channel][sample].
All the files using the ChannelBuffer needed to be re-factored.
Tested with modules_unittests, common_audio_unittests, audioproc, audioproc_f, voe_cmd_test.
R=andrew@webrtc.org, kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36999004
Cr-Commit-Position: refs/heads/master@{#8318}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8318 4adac7df-926f-26a2-2b94-8c16560cd09d
This makes it possible to build more flexible simulations, and makes it easier to implement bi-directional simulations. This also removes support for generating baseline files and comparing against a baseline, which hasn't turned out to be particuarly useful.
BUG=4173
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/35069004
Cr-Commit-Position: refs/heads/master@{#8311}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8311 4adac7df-926f-26a2-2b94-8c16560cd09d
This makes it possible to build more flexible simulations, and makes it easier to implement bi-directional simulations. This also removes support for generating baseline files and comparing against a baseline, which hasn't turned out to be particuarly useful.
BUG=4173
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/34989004
Cr-Commit-Position: refs/heads/master@{#8297}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8297 4adac7df-926f-26a2-2b94-8c16560cd09d
We apparently hit an obscure problem on mac where seemingly an unaligned mutex causes memory corruption.
The effect was that the |modules_| list became corrupt and we crashed. At this point I'm not exactly
sure what the alignment requirements are but for now, I've fixed up the layout in a way that doesn't cause these same issues.
I'm also changing auto->proper type at the request of drive by reviewers from my previous cl in the same file.
TBR=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/38989004
Cr-Commit-Position: refs/heads/master@{#8286}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8286 4adac7df-926f-26a2-2b94-8c16560cd09d
Previously only mic level calculated by the legacy agc was logged in aecdebug dumps.
Now we log it for any agc.
In addition, it is now possible to turn on and off debug recording in the test tool voe_cmd_test.
BUG=4274
TESTED=verified using voe_cmd_test
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/39839004
Cr-Commit-Position: refs/heads/master@{#8274}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8274 4adac7df-926f-26a2-2b94-8c16560cd09d
This change switches from the old codec wrappers ACMPCMU and ACMPCMA
to the new AudioEncoderPcmU and AudioEncoderPcmA wrapped in an
ACMGenericCodecWrapper. RED and CNG is also switched to using their
AudioEncoder implementations (AudioEncoderCopyRed and AudioEncoderCng,
respectively), when RED and/or CNG is combined with PCM u/A.
This is the first in a series of changes that will switch all codecs
to use the new AudioEncoder interface.
BUG=4228
COAUTHOR=kwiberg@webrtc.orgR=minyue@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/33209004
Cr-Commit-Position: refs/heads/master@{#8268}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8268 4adac7df-926f-26a2-2b94-8c16560cd09d
sending RTP module for the specified simulcast layer a frame belongs to.
This CL also removes the corresponding functionality from the RTP RTCP
module and fixes lint warnings in the files touched.
BUG=769
TEST=New unittest and manual tests
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/39629004
Cr-Commit-Position: refs/heads/master@{#8267}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8267 4adac7df-926f-26a2-2b94-8c16560cd09d
The new format instantiates the RemoteBitrateEstimator at the send-side and feeds back all packet arrival timestamps and sequence numbers to the sender, where inter-arrival deltas are calculated.
Next step will be to make feedback packets part of regular packets and send them over the network. This also requires bi-directional simulations.
BUG=4173
R=mflodman@webrtc.org, sprang@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/37109004
Cr-Commit-Position: refs/heads/master@{#8264}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8264 4adac7df-926f-26a2-2b94-8c16560cd09d
* Add a new WakeUp method that gives a module a chance to be called back right away on the worker thread.
* Wrote unit tests for the class.
* Significantly reduce the amount of locking.
- ProcessThreadImpl itself does a lot less locking.
- Reimplemented the way we keep track of when to make calls to Process.
This reduces the amount of calls to TimeUntilNextProcess and since most implementations of that function grab a lock, this means less locking.
* Renamed ProcessThread::CreateProcessThread to ProcessThread::Create.
* Added thread checks for Start/Stop. Threading model of other functions is now documented.
* We now log an error if an implementation of TimeUntilNextProcess returns a negative value (some implementations do, but the method should only return a positive nr of ms).
* Removed the DestroyProcessThread method and instead force callers to use scoped_ptr<> to maintain object lifetime.
BUG=2822
R=henrika@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/35999004
Cr-Commit-Position: refs/heads/master@{#8261}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8261 4adac7df-926f-26a2-2b94-8c16560cd09d
This test will replace AcmOpusTest when ACMOpus is removed. The old
AcmOpusTest also contains tests for setting and updating the
"application" setting in Opus. However, in the new AudioEncoderOpus
class, the application is trivially set in the Config struct at
construction, wherefore a test is no longer needed.
BUG=3926
R=minyue@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/37929004
Cr-Commit-Position: refs/heads/master@{#8244}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8244 4adac7df-926f-26a2-2b94-8c16560cd09d
This is the same change as already made for Windows:
https://webrtc-codereview.appspot.com/37069004/
* Remove "dead" and "alive" variables.
* Remove critical section
* Remove implementation of SetNotAlive()
* Always set thread name
* Add thread checks for correct usage.
* Changed AudioDeviceMac to create/start/stop/delete thread objects for playout and recording, inside the respective start and stop method. The reason for this is because the AudioDeviceMac instance is currently being created on one thread and the above Start/Stop methods are being called on a different thread. So, my change makes creation, start/stop, deletion of the thread objects always happen on the same thread.
I'm making CurrentThreadId() in rtc_base_approved more visible so that it can be used from there instead of inside webrtc. Down the line we will have more thread concepts in rtc_base_approved, so I put a TODO for myself to move this functionality to there once we do.
R=henrika@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/40599004
Cr-Commit-Position: refs/heads/master@{#8235}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8235 4adac7df-926f-26a2-2b94-8c16560cd09d
