A better solution than forcing OPUS_APPLICATION_VOIP when enabling DTX has been found, which is to set OPUS_SIGNAL_VOICE.
This reduces the uncertainty of entering DTX over silence period of audio.
This CL contains the setup of OPUS_SIGNAL_VOICE and decoupling opus application mode with DTX.
BUG=4559
R=henrik.lundin@webrtc.org, henrika@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/46959004
Cr-Commit-Position: refs/heads/master@{#9168}
CodecOwner is introduced here; AudioEncoderMutable was introduced in a
previous commit, but had no users until now. The only remaining task
for ACMGenericCodec was to construct and maintain the stack of speech,
CNG, and RED encoders. This task is now handled by the CodecOwner,
which is owned and used by the CodecManager.
COAUTHOR=henrik.lundin@webrtc.org
BUG=4228
R=jmarusic@webrtc.org, minyue@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/43189004
Cr-Commit-Position: refs/heads/master@{#9152}
Passed building isac_neon and modules_unittests on Android ARM64 and
ARMv7.
Passed modules_unittests with following filters:
--gtest_filter=FiltersTest*
--gtest_filter=LpcMaskingModelTest*
--gtest_filter=TransformTest*
--gtest_filter=FilterBanksTest*
WebRtcIsacfix_CalculateResidualEnergyNeon is not enabled due to Issue
4224.
BUG=4002
R=andrew@webrtc.org, jridges@masque.com, kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/44229004
Patch from Zhongwei Yao <zhongwei.yao@arm.com>.
Cr-Commit-Position: refs/heads/master@{#9092}
This change is needed by ChromeOS as it introduces -fno-omit-frame-pointer
flag (see code.google.com/p/chromium/issues/detail?id=477749). This causes
compile error for MIPS, as some MIPS optimization blocks use maximum possible
number of available registers.
Also, this change contains minor GN build fix for MIPS platform regarding the
pitch_filter_mips.c / pitch_filter_c.c file inclusion.
BUG=477749
R=andrew@webrtc.org, djordje.pesut@imgtec.com, tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/48139004
Patch from Ljubomir Papuga <lpapuga@mips.com>.
Cr-Commit-Position: refs/heads/master@{#9047}
Also change to use virtual_packet_length_bytes in order to print the
actual packet size of the complete packet even when the RTP file only
contains RTP headers.
BUG=2692
R=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/51559004
Cr-Commit-Position: refs/heads/master@{#9025}
With this change, the currently used encoder is held in a scoped_ptr.
iSAC is a special case, since the encoder instance is also a decoder
instance, so it may have to be available also if another send codec is
used. This is accomplished by having a separate scoped_ptr for iSAC.
Remove mirror ID from ACM codec database functions, and remove unused
functions from the database.
COAUTHOR=kwiberg@webrtc.org
BUG=4228
R=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/48729004
Cr-Commit-Position: refs/heads/master@{#8982}
The macro is defined as
#define WEBRTC_SPL_MUL_16_16_RSFT(a, b, c) \
(WEBRTC_SPL_MUL_16_16(a, b) >> (c))
where the latter macro is in C defined as
#define WEBRTC_SPL_MUL_16_16(a, b) \
((int32_t) (((int16_t)(a)) * ((int16_t)(b))))
(For definitions on ARMv7 and MIPS, see common_audio/signal_processing/include/spl_inl_{armv7,mips}.h)
The replacement consists of
- avoiding casts to int16_t if inputs already are int16_t
- adding explicit cast to <type> if result is assigned to <type> (other than int or int32_t)
- minor cleanups like remove of unnecessary parentheses and style changes
- removed commented code lines used during development
- excluded fft.c since there are neon optimizations used and a removal may cause a performance regression
BUG=3348, 3353
TESTED=locally on linux and trybots
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/48799004
Cr-Commit-Position: refs/heads/master@{#8967}
Now that android_webview_build is no longer supported, remove build
conditionals referencing it and also remove the extra level of
indirection used to reference the cpufeatures target.
BUG=chromium:440793
R=henrika@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/44119005
Patch from Richard Coles <torne@chromium.org>.
Cr-Commit-Position: refs/heads/master@{#8963}
This reverts commit cf3c83e76c273309558c86fda915410f65b7a899.
Reverting EventWrapper split did not fix the issue, re-landing.
BUG=chromium:470013
TBR=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/49629004
Cr-Commit-Position: refs/heads/master@{#8946}
The macro is defined as
#define WEBRTC_SPL_LSHIFT_W32(a, b) ((a) << (b))
hence trivial.
The macro name may in fact mislead the user to assume a cast/truncation to int32_t is done.
- Removing usage of it.
- Some style changes.
BUG=3348, 3353
TESTED=locally on linux and trybots
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/46749005
Cr-Commit-Position: refs/heads/master@{#8918}
This reverts commit 9509fbfc301dd5412804ce5731afedc81480f2f8.
This is to debug a Chromium issue that WebRTC hangs if there is > 1 PeerConnection active in the browser on Win XP.
BUG=
TBR=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/43019004
Cr-Commit-Position: refs/heads/master@{#8912}
Pass content_browsertests in Chromium. Performance test result (lower is
better):
C version: 100%
old intrinsics Neon version (with bug): 16.5%
new intrinsics Neon version: 18.0%
asm Neon version: 23.3%
BUG=4002
R=andrew@webrtc.org, jridges@masque.com
Change-Id: Ia0a96ac237216b635fc528f67d39319cdf246281
Review URL: https://webrtc-codereview.appspot.com/46739004
Cr-Commit-Position: refs/heads/master@{#8907}
This change essentially divides AudioCodingModuleImpl into two parts:
one is the code related to managing codecs, now moved into CodecManager,
and the other is what remains in AudioCodingModuleImpl.
This change also removes AudioCodingModuleImpl::InitializeSender. The
function was essentially no-op, since it was always called immediately
after construction.
COAUTHOR=kwiberg@webrtc.org
BUG=4228
R=minyue@webrtc.org, tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/51469004
Cr-Commit-Position: refs/heads/master@{#8893}
We want to crate the illusion that iSAC supports 48000 Hz decoding,
while in fact it outputs 32000 Hz. This is the iSAC fullband mode.
Currently this is (also) handled by higher layers, but in future
changes this will not be the case.
R=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/47809004
Cr-Commit-Position: refs/heads/master@{#8889}
Change internal indexing of registered decoders. It makes sense because payload type is unique, while ACM codec ID may not be. This is a step towards allowing for addition of external decoders.
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/44869004
Cr-Commit-Position: refs/heads/master@{#8867}
I'm splitting the timer functions in EventWrapper into a separate interface.
- Users of the timer functions have different needs than users of a generic event
- Providing a default implementation for EventWrapper that simply uses rtc::Event.
This means that clients of WebRTC that don't use the relatively few classes, typically rendering classes, that depend on the event timer functionality, also don't pull in dependencies on multimedia timers.
R=mflodman@webrtc.org, mflodman
BUG=
Review URL: https://webrtc-codereview.appspot.com/48599004
Cr-Commit-Position: refs/heads/master@{#8833}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8833 4adac7df-926f-26a2-2b94-8c16560cd09d
In SDP, RED audio codec has its own sample rate. Currently, we offer RED/8000 (8 kHz). But the actual send codec can violate this sample rate. The way to solve it is to introduce more RED payload types, e.g., RED/16000, RED/32000.
As a first step towards that, we, in this CL, limit the current RED (RED/8000) to work only with 8 kHz codecs.
BUG=3619
R=henrik.lundin@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/43849004
Cr-Commit-Position: refs/heads/master@{#8830}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8830 4adac7df-926f-26a2-2b94-8c16560cd09d
A codec's packet-loss concealer is called once from NetEq before
decoding the first packet after a packet loss. The purpose is not to
use the PLC output, but to prepare the state of the decoder such that
it may recover faster after the loss. However, this effect is not
achieved by calling iSAC's PLC. Also, there are some problems with the
fixed-point implementation of the PLC (see the associated bug).
BUG=4423
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/42849004
Cr-Commit-Position: refs/heads/master@{#8827}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8827 4adac7df-926f-26a2-2b94-8c16560cd09d
The macro is defined as
#define WEBRTC_SPL_LSHIFT_W32(a, b) ((a) << (b))
It is a trivial operation that need no macro. In fact it may be confusing for to the user, since it can be interpreted as having an implicit cast to int32_t.
BUG=3348,3353
TESTED=locally on linux and trybots
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/44659004
Cr-Commit-Position: refs/heads/master@{#8801}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8801 4adac7df-926f-26a2-2b94-8c16560cd09d
The macro is defined as
#define WEBRTC_SPL_LSHIFT_W32(a, b) ((a) << (b))
It is a trivial operation that need no macro. In fact it may be confusing for to the user, since it can be interpreted as having an implicit cast to int32_t.
Also removes unnecessary casts to int32_t from int16_t.
BUG=3348,3353
TESTED=locally on linux and trybots
R=kwiberg@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/48519004
Cr-Commit-Position: refs/heads/master@{#8800}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8800 4adac7df-926f-26a2-2b94-8c16560cd09d
