Reason for revert:
There is a risk of ending up in a bad state due to race conditions with this patch. Tests in downstream clients have shown that it can
happen that an output stream is opened up in MUSIC mode when it should not.
Reverting since the new functionality added here is not worth the
risk of breaking existing clients.
Original issue's description:
> Android audio playout now supports non-call media streams.
>
> The default (preferred) stream type for output audio is STREAM_VOICE_CALL since the WebRTC stack is mainly intended for VoIP calls. But if the user wants to run in another mode than COMM mode, we now accept it and change the stream type to STREAM_MUSIC instead. It can e.g. be suitable for applications that does not record audio or if a call shall be casted to a Chromecast device.
>
> The solution is somewhat experimental.
>
> NOTRY=TRUE
>
> BUG=webrtc:4767
>
> Committed: https://crrev.com/872f614111f436d15e29516ce19c3b63d25b8639
> Cr-Commit-Position: refs/heads/master@{#14613}
TBR=henrik.lundin@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:4767
Review-Url: https://codereview.webrtc.org/2420583002
Cr-Commit-Position: refs/heads/master@{#14626}
The default (preferred) stream type for output audio is STREAM_VOICE_CALL since the WebRTC stack is mainly intended for VoIP calls. But if the user wants to run in another mode than COMM mode, we now accept it and change the stream type to STREAM_MUSIC instead. It can e.g. be suitable for applications that does not record audio or if a call shall be casted to a Chromecast device.
The solution is somewhat experimental.
NOTRY=TRUE
BUG=webrtc:4767
Review-Url: https://codereview.webrtc.org/2411263003
Cr-Commit-Position: refs/heads/master@{#14613}
NOTE: the new code is disabled by default in the WebRtcAudioManager to ensure that
OpenSL ES is not accidentally activated in existing clients. There are still some
unresolved issues to sort out before it can be utilized.
Enables possibility to use OpenSL ES based audio in both directions for WebRTC.
All unit tests and demo clients have been tested with the new implementation but
the new support is behind a flag (see above).
More testing is needed before it can be used in the field and additional support for
hardware effects is still missing.
BUG=webrtc:5925
R=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/2119633004 .
Cr-Commit-Position: refs/heads/master@{#14290}
A new API is added which enables detection of support of pro-audio on
Android. This is part of a larger change and the new API is not used yet.
Most likely it will only be used for logging purposes.
BUG=webrtc:5925
Review-Url: https://codereview.webrtc.org/2015483002
Cr-Commit-Position: refs/heads/master@{#12890}
This changes the following module directories:
* webrtc/modules/audio_conference_mixer/interface
* webrtc/modules/interface
* webrtc/modules/media_file/interface
* webrtc/modules/rtp_rtcp/interface
* webrtc/modules/utility/interface
To avoid breaking downstream, I followed this recipe:
1. Copy the interface dir to a new sibling directory: include
2. Update the header guards in the include directory to match the style guide.
3. Update the header guards in the interface directory to match the ones in include. This is required to avoid getting redefinitions in the not-yet-updated downstream code.
4. Add a pragma warning in the header files in the interface dir. Example:
#pragma message("WARNING: webrtc/modules/interface is DEPRECATED; "
"use webrtc/modules/include")
5. Search for all source references to webrtc/modules/interface and update them to webrtc/modules/include (*.c*,*.h,*.mm,*.S)
6. Update all GYP+GN files. This required manual inspection since many subdirectories of webrtc/modules referenced the interface dir using ../interface etc(*.gyp*,*.gn*)
BUG=5095
TESTED=Passing compile-trybots with --clobber flag:
git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel -m tryserver.webrtc
R=stefan@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1417683006 .
Cr-Commit-Position: refs/heads/master@{#10500}
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 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}
