This CL contains major modifications of the audio output parts for WebRTC on iOS:
- general code cleanup
- improves thread handling (added thread checks, remove critical section, atomic ops etc.)
- reduces loopback latency of iPhone 6 from ~90ms to ~60ms ;-)
- improves selection of audio parameters on iOS
- reduces complexity by removing complex and redundant delay estimates
- now instead uses fixed delay estimates if for some reason the SW EAC must be used
- adds AudioFineBuffer to compensate for differences in native output buffer size and
the 10ms size used by WebRTC. Same class as is used today on Android and we have unit tests for
this class (the old code was buggy and we have several issue reports of crashes related to it)
Similar improvements will be done for the recording sid as well in a separate CL.
I will also add support for 48kHz in an upcoming CL since that will improve Opus performance.
BUG=webrtc:4796,webrtc:4817,webrtc:4954, webrtc:4212
TEST=AppRTC demo and iOS modules_unittests using --gtest_filter=AudioDevice*
R=pbos@webrtc.org, tkchin@webrtc.org
Review URL: https://codereview.webrtc.org/1254883002 .
Cr-Commit-Position: refs/heads/master@{#9875}
AudioDeviceTemplate doesn't initialize `output_` and `input_` if the
initialization of `audio_manager_` succeeds. Similarly, it doesn't
terminate `input_` and `audio_manager_` if the termination of `output_`
succeeds. This CL fixes this.
BUG=
Review URL: https://codereview.webrtc.org/1296693003
Cr-Commit-Position: refs/heads/master@{#9760}
On GetCapabilities() failure, caps.cDestinations is left uninitialized.
Without a protection the following code runs in a random loop
in the worst case up to 0xFFFFFFFF times.
for (destId = 0; destId < caps.cDestinations; destId++)
{
GetDestinationLineInfo(mixId, destId, destLine);
BUG=webrtc:4882
Review URL: https://codereview.webrtc.org/1269563002
Cr-Commit-Position: refs/heads/master@{#9663}
These are mostly trivial changes and are separated out just to reduce the
diff on that change to the minimum possible.
Note explanatory comments on patch set 1.
BUG=none
TEST=none
Review URL: https://codereview.webrtc.org/1235643003
Cr-Commit-Position: refs/heads/master@{#9617}
In #9243 we added some thread_checker. But it shouldn't be added into PlayoutDevices() and RecordingDevices(), since these two will be invoked from RecThread and PlayoutThread too, other than the main thread.
BUG=webrtc:4852
TEST=voe_cmd_test
R=henrika@webrtc.org
Review URL: https://codereview.webrtc.org/1249573002 .
Cr-Commit-Position: refs/heads/master@{#9605}
This includes changes like:
* Attempt to break lines at better positions
* Use "override" in more places, don't use "virtual" with it
* Use {} where the body is more than one line
* Make declaration and definition arg names match
* Eliminate unused code
* EXPECT_EQ(expected, actual) (but use (actual, expected) for e.g. _GT)
* Correct #include order
* Use anonymous namespaces in preference to "static" for file-scoping
* Eliminate unnecessary casts
* Update reference code in comments of ARM assembly sources to match actual current C code
* Fix indenting to be more style-guide compliant
* Use arraysize() in more places
* Use bool instead of int for "boolean" values (0/1)
* Shorten and simplify code
* Spaces around operators
* 80 column limit
* Use const more consistently
* Space goes after '*' in type name, not before
* Remove unnecessary return values
* Use "(var == const)", not "(const == var)"
* Spelling
* Prefer true, typed constants to "enum hack" constants
* Avoid "virtual" on non-overridden functions
* ASSERT(x == y) -> ASSERT_EQ(y, x)
BUG=none
R=andrew@webrtc.org, asapersson@webrtc.org, henrika@webrtc.org, juberti@webrtc.org, kjellander@webrtc.org, kwiberg@webrtc.org
Review URL: https://codereview.webrtc.org/1172163004
Cr-Commit-Position: refs/heads/master@{#9420}
This makes a variety of small changes to synchronize bits of code using different types, remove useless code or casts, and add explicit casts in some places previously doing implicit ones. For example:
* Change a few type declarations to better match how the majority of code uses those objects.
* Eliminate "< 0" check for unsigned values.
* Replace "(float)sin(x)", where |x| is also a float, with "sinf(x)", and similar.
* Add casts to uint32_t in many places timestamps were used and the existing code stored signed values into the unsigned objects.
* Remove downcasts when the results would be passed to a larger type, e.g. calling "foo((int16_t)x)" with an int |x| when foo() takes an int instead of an int16_t.
* Similarly, add casts when passing a larger type to a function taking a smaller one.
* Add casts to int16_t when doing something like "int16_t = int16_t + int16_t" as the "+" operation would implicitly upconvert to int, and similar.
* Use "false" instead of "0" for setting a bool.
* Shift a few temp types when doing a multi-stage calculation involving typecasts, so as to put the most logical/semantically correct type possible into the temps. For example, when doing "int foo = int + int; size_t bar = (size_t)foo + size_t;", we might change |foo| to a size_t and move the cast if it makes more sense for |foo| to be represented as a size_t.
BUG=none
R=andrew@webrtc.org, asapersson@webrtc.org, henrika@webrtc.org, juberti@webrtc.org, kwiberg@webrtc.org
TBR=andrew, asapersson, henrika
Review URL: https://codereview.webrtc.org/1168753002
Cr-Commit-Position: refs/heads/master@{#9419}
In my previous cl, https://webrtc-codereview.appspot.com/52479004/, there is 'UnLock()' left when we switched to scoped lock, which will cause TSan warning sometimes.
===========================================================
WARNING: ThreadSanitizer: unlock of an unlocked mutex (or by a wrong thread) (pid=9981)
#0 pthread_mutex_unlock <null> (libjingle_peerconnection_unittest+0x00000046836f)
#1 webrtc::CriticalSectionPosix::Leave() webrtc/system_wrappers/source/critical_section_posix.cc:39:10 (libjingle_peerconnection_unittest+0x000000bc368d)
#2 ~CriticalSectionScoped webrtc/system_wrappers/interface/critical_section_wrapper.h:46:48 (libjingle_peerconnection_unittest+0x000000a61fcb)
#3 webrtc::AudioDeviceLinuxPulse::RecThreadProcess() webrtc/modules/audio_device/linux/audio_device_pulse_linux.cc:3003 (libjingle_peerconnection_unittest+0x000000a61fcb)
===========================================================
BUG=3056
TEST=bots
R=tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/56439004
Cr-Commit-Position: refs/heads/master@{#9282}
Add pylintrc file based on
https://code.google.com/p/chromium/codesearch#chromium/src/tools/perf/pylintrc
bit tightened up quite a bit (the one in depot_tools is far
more relaxed).
Remove a few excluded directories from pylint check and fixed/
suppressed all warnings generated.
Add GN format check + formatted all GN files using 'gn format'.
Cleanup redundant rules in tools/PRESUBMIT.py
TESTED=Ran 'git cl presubmit -vv', fixed the PyLint violations.
Ran it again with a modification in webrtc/build/webrtc.gni, formatted
all the GN files and ran it again.
R=henrika@webrtc.org, phoglund@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/50069004
Cr-Commit-Position: refs/heads/master@{#9274}
BUG=3056, 1320
TEST=AutoTest
Mainly add threadchecker and remove unnecessary lock.
And some more styling working.
- audio_device_pulse_linux.cc: wrap lines longer than 80 chars. And add '.' to some comments around. Not do it to all places.
- audio_mixer_manager_pulse_linux.cc: Here I adopt some chromium practice. We use to do many things to the failure of pulse operation, which causes most of the data race issue. In chromium, if we failed to call any pulse function, we just fail it w/o use the previous results. Here I did same. Please check if it's good.
R=bjornv@webrtc.org, henrika@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/52479004
Cr-Commit-Position: refs/heads/master@{#9243}
Avoids existing crash and ensures that error message is passed up to Libjingle. Will lead to the following logcat output:
E/libjingle(31404): Error(channel.cc:1514): Failed to SetSend 2 on voice channel
BUG=b/21273153
R=magjed@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/54459004
Cr-Commit-Position: refs/heads/master@{#9236}
The class doesn't do anything in almost all cases except for grabbing and releasing locks + allocate memory. There are a couple of methods there such as WaitForKey and GetTimeInMs that are used, but those methods aren't specific to audio and we have implementations of these elsewhere. The third method, StringCompare isn't used anywhere (and also isn't specific to audio).
BUG=
R=henrika@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/50009004
Cr-Commit-Position: refs/heads/master@{#9220}
BUG=4573,2982,2175,3590
TEST=modules_unittests --gtest_filter=AudioDevice*, AppRTCDemo and WebRTCDemo
Summary:
- Removes dependency of the 'enable_android_opensl' compiler flag.
Instead, OpenSL ES is always supported, and will enabled for devices that
supports low-latency output.
- WebRTC no longer supports OpenSL ES for the input/recording side.
- Removes old code and demos using OpenSL ES for audio input.
- Improves accuracy of total delay estimates (better AEC performance).
- Reduces roundtrip audio latency; especially when OpenSL can be used.
Performance verified on: Nexus 5, 6, 7 and 9. Samsung Galaxy S4 and S6.
Android One device.
R=magjed@webrtc.org, phoglund@webrtc.org, tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/51759004
Cr-Commit-Position: refs/heads/master@{#9208}
Also prevents that we try to restore audio mode when it has not been changed.
TBR=glaznev
BUG=NONE
TEST=AppRTCDemo and verify that volume control switches from "Ringtone to Phone" mode when call starts and switches back to Ringtone mode when call ends.
Review URL: https://webrtc-codereview.appspot.com/46879004
Cr-Commit-Position: refs/heads/master@{#8975}
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}
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}
Also adds a framework for an AudioManager to be used by both sides (playout and recording).
This initial implementation only does very simple tasks like setting up the correct audio
mode (needed for correct volume behavior). Note that this CL is mainly about modifying
the volume. The added AudioManager is only a place holder for future work. I could have
done the same parts in the WebRtcAudioTrack class but feel that it is better to move these
parts to an AudioManager already at this stage.
The AudioManager supports Init() where actual audio changes are done (set audio mode etc.)
but it can also be used a simple "construct-and-store-audio-parameters" unit, which is the
case here. Hence, the AM now serves as the center for getting audio parameters and then inject
these into playout and recording sides. Previously, both sides acquired their own parameters
and that is more error prone.
BUG=NONE
TEST=AudioDeviceTest
R=perkj@webrtc.org, phoglund@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/45829004
Cr-Commit-Position: refs/heads/master@{#8875}
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
