* Move constants into the files/functions that use them
* Declare variables in the narrowest scope possible
* Use correct (expected, actual) order for gtest macros
* Remove unused functions
* Untabify
* 80-column limit
* Avoid C-style casts
* Prefer true typed constants to "enum hack" constants
* Print size_t using the right format macro
* Shorten and simplify code
* Other random cleanup bits and style fixes
BUG=none
TEST=none
R=henrik.lundin@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36179004
Cr-Commit-Position: refs/heads/master@{#8467}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8467 4adac7df-926f-26a2-2b94-8c16560cd09d
ACMGenericCodecWrapper was the only remaining subclass of
ACMGenericCodec, and was the only class that was ever instantiated.
This CL merges the two, essentially keeping the function implementations
from ACMGenericCodecWrapper except where the base class's code was
invoked.
As it turns out, a lot of functions were never used, but in some cases
they were refernced in AudioCodingModuleImpl. In these cases, the
referencing code is commented out and marked FATAL(). This will be
further cleaned up in follow-up CLs.
BUG=4228
COAUTHOR=kwiberg@webrtc.orgR=minyue@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/38209004
Cr-Commit-Position: refs/heads/master@{#8463}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8463 4adac7df-926f-26a2-2b94-8c16560cd09d
This CL is part II in a major refactoring effort. See https://webrtc-codereview.appspot.com/33969004 for part I.
- Removes unused code and old WEBRTC logging macros
- Now uses optimal sample rate and buffer size in Java AudioTrack (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)
- Simplified the delay estimate
- 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
Review URL: https://webrtc-codereview.appspot.com/39169004
Cr-Commit-Position: refs/heads/master@{#8460}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8460 4adac7df-926f-26a2-2b94-8c16560cd09d
They have all been replaced by AudioEncoder subclasses, accessed throgh
ACMGenericCodecWrapper objects. After this change, the only subclass of
ACMGenericCodec is ACMGenericCodecWrapper. (The two will be consolidated
in a future cl.)
This CL also deletes acm_opus_unittest.cc. This test file was already
replaced audio_encoder_opus_unittest.cc in r8244.
BUG=4228
COAUTHOR=kwiberg@webrtc.orgR=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/40729004
Cr-Commit-Position: refs/heads/master@{#8457}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8457 4adac7df-926f-26a2-2b94-8c16560cd09d
This cl just moves the logic form the default module
SetTargetSendBitrates to PayloadRouter. There might be glitch / mismatch
in size between trate the vector and rtp modules. This was the same in
the default module and is quite hard to protect from before we have the
new video API.
I also removed some test form rtp_rtcp_impl_unittest that were affected
by this change. The test tests code that isn't implemented, hence the
DISABLED_, and this will never be implemented in the RTP module, rather
the payload router in the future.
BUG=769
R=pbos@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/42419004
Cr-Commit-Position: refs/heads/master@{#8453}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8453 4adac7df-926f-26a2-2b94-8c16560cd09d
* Don't use ConjugateDotProduct to calculate the norm.
* Only resize Matrix when needed.
This makes the Beamformer run in 93.6% the original time.
The error between the new and original output is really small and is caused by the new norm calculation.
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/37339004
Cr-Commit-Position: refs/heads/master@{#8438}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8438 4adac7df-926f-26a2-2b94-8c16560cd09d
Fetching the current codec for sake of gathering stats, is frequently blocked since it's done by acquiring the same lock as is held while encoding frames. This can mean tens of milliseconds.
To improve this, I'm taking advantage of the fact that the codec information is set on the same thread as is used to query the information. This means that locking isn't needed for querying this information. I'm adding checks to make sure debug builds will crash if this isn't followed.
An alternative to this approach could be to add one more lock that is specifically used for the codec information variable. This would also decouple querying codec information from the encoder itself, but still requires a lock.
This patch depends on making ThreadChecker part of rtc_base_approved:
https://webrtc-codereview.appspot.com/40539004/
BUG=2822
R=mflodman@webrtc.org, pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/37779004
Cr-Commit-Position: refs/heads/master@{#8435}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8435 4adac7df-926f-26a2-2b94-8c16560cd09d
When an "empty frame" was produced by ACMGenericCodecWrapper::Encode,
the timestamp value was not set. This is now fixed, and the first byte
of the bitstream is set to something as well to avoid similar problems.
BUG=chromium:459483
R=jmarusic@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/34269004
Cr-Commit-Position: refs/heads/master@{#8431}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8431 4adac7df-926f-26a2-2b94-8c16560cd09d
The old AcmOpusTest depends on the ACMOpus class, but this class was
obsoleted by AudioEncoderOpus. In this CL, the test code is re-written
to use AudioEncoderOpus and ACMGenericCodecWrapper instead of
ACMOpus. Most of the test functionality is preserved, except for the
packet loss rate tests, which where already transferred to
AudioEncoderOpusTest in r8244.
R=kwiberg@webrtc.org, minyue@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/40029004
Cr-Commit-Position: refs/heads/master@{#8410}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8410 4adac7df-926f-26a2-2b94-8c16560cd09d
Although sample_rate_hz(), num_channels(), and rtp_timestamp_rate_hz()
are simple accessors for almost all implementations of AudioEncoder,
they are virtual and not guaranteed to be just simple accessors. Thus,
it makes more sense to use the normal CamelCase naming scheme.
BUG=4235
R=henrik.lundin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/34239004
Cr-Commit-Position: refs/heads/master@{#8407}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8407 4adac7df-926f-26a2-2b94-8c16560cd09d
Unblocks pending threads (render thread + decoder thread) when
destroying renderers and shutting down decoders.
Speeds up SetLocalDescription significantly (10x or so) under
WebRtcVideoEngine2 but also shutdown times in ~ViEChannel and
~ViEReceiver in general.
BUG=1788
R=mflodman@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/41959004
Cr-Commit-Position: refs/heads/master@{#8387}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8387 4adac7df-926f-26a2-2b94-8c16560cd09d
Modifies WebRtcVideoSendStream to use a default width/height of 16px.
This significantly reduces SetRemoteDescription time under
WebRtcVideoEngine2. Also preventing (expensive) reconfigurations due to
incoming frames when the channel is not sending yet.
Tests have been modified to generate a frame before expecting a certain
encoder size to have been configured.
Also adding tracing to WebRtcVideoSendStream::InputFrame as it can lead
to reconfigurations of the encoder which is expensive and it should show
up in chrome://tracing.
BUG=1788
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/42369004
Cr-Commit-Position: refs/heads/master@{#8381}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8381 4adac7df-926f-26a2-2b94-8c16560cd09d
In some cases it was not possible to set another payload type for CNG
than the default one. This CL fixes this. The problem was also
dependent on whether the comfort noise codec was registered before or
after the speech codec.
A test is implement to expose the bug, registering comfort noise at a
non-default payload type, and both before and after the speech codec.
BUG=4228
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/35199004
Cr-Commit-Position: refs/heads/master@{#8380}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8380 4adac7df-926f-26a2-2b94-8c16560cd09d
