This splits things out of rtc_base and makes dependencies explicit.
Bug: webrtc:6828
Change-Id: Ib813c7bd9e4de7ab015acb917bc09ee7204ba7bd
Reviewed-on: https://webrtc-review.googlesource.com/31940
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21245}
Avoids hitting a DCHECK in AudioRecordJni::OnCacheDirectBufferAddress()
when first init attempt has failed and we try again.
Bug: b/69434512
Change-Id: I4396ba22981d9258d6d72188bad66104255f19cf
Reviewed-on: https://webrtc-review.googlesource.com/31842
Reviewed-by: Alex Glaznev <glaznev@webrtc.org>
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21218}
Usage should now be removed and this change can be relanded.
It was reverted here: https://webrtc-review.googlesource.com/c/src/+/27200
NOTRY=TRUE
TBR=solenberg
Bug: webrtc:7306
Change-Id: I5191263e6cfd48952b59ff8f9af2e59c3e9eadef
Reviewed-on: https://webrtc-review.googlesource.com/29682
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21080}
This reverts commit 94f39301067b9fbf820100cbd4018aad3a32cc52.
Reason for revert: Broke internal builds
Original change's description:
> Removes deprecated ADM APIs (reland).
>
> Usage should now be removed and this change can be relanded.
> It was reverted here: https://webrtc-review.googlesource.com/c/src/+/25320
>
> TBR=solenberg
>
> Bug: webrtc:7306
> Change-Id: I1afea773eff51bf5ec80711f0d7753ac0b7be77b
> Reviewed-on: https://webrtc-review.googlesource.com/27000
> Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
> Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20936}
TBR=henrika@webrtc.org
Change-Id: If91ff815fa69f7c36b0531e295f553a8c4a95590
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:7306
Reviewed-on: https://webrtc-review.googlesource.com/27221
Reviewed-by: Lu Liu <lliuu@webrtc.org>
Commit-Queue: Lu Liu <lliuu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20944}
Usage should now be removed and this change can be relanded.
It was reverted here: https://webrtc-review.googlesource.com/c/src/+/25320
TBR=solenberg
Bug: webrtc:7306
Change-Id: I1afea773eff51bf5ec80711f0d7753ac0b7be77b
Reviewed-on: https://webrtc-review.googlesource.com/27000
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20936}
This type of check should instead be performed by the application/client.
If the app does not have mic permissions, construction of the AudioRecord
object will fail and the user will receive an error callback anyhow.
Bug: b/69434512
Change-Id: If1d7eff488f7c693697e048a567c17ed0c51f040
Reviewed-on: https://webrtc-review.googlesource.com/25261
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20839}
Final stage since these APIs are no longer used in Chrome.
Bug: webrtc:7306
Change-Id: Ia116671bc888daa75c4105ad1ebeb21833f5d090
Reviewed-on: https://webrtc-review.googlesource.com/25220
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20836}
Specifically, I'm moving
safe_compare.h
safe_conversions.h
safe_minmax.h
They shouldn't be part of the API, and moving them to an appropriate
subdirectory of rtc_base/ is a good way to keep track of that.
BUG=webrtc:8445
Change-Id: I458531aeb30bcf4291c4bec3bf22a2fffbf054ff
Reviewed-on: https://webrtc-review.googlesource.com/20860
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20829}
Reverted the logic introduced in https://codereview.webrtc.org/2933953003
This is because the audio device buffer behavior changed with https://codereview.webrtc.org/3009193002, and the RequestPlayoutData method now returns number of samples in each channel, which creates mismatch the reverted CL.
Bug: webrtc:8548
Change-Id: Id4711ca48437ddd3483327c2a4c7827d09e5b770
Reviewed-on: https://webrtc-review.googlesource.com/24122
Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
Commit-Queue: Lu Liu <lliuu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20737}
int32_t SetLoudspeakerStatus(bool enable)
int32_t GetLoudspeakerStatus(bool* enabled) const
These APIs are only implemented on iOS and they do not belong in the
native audio layer since the client can achieve the same functionality
by using the shared audio session in sdk/objc/Framework/Headers/WebRTC/RTCAudioSession.h.
It also gives the client a better flexibility in how the audio routing is done.
Bug: webrtc:7306
Change-Id: I853e2f57e0f5ae0a0f9fc4729ce961d81f92588b
Reviewed-on: https://webrtc-review.googlesource.com/23740
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20721}
Conditional visibility is complex to maintain and it is not well
supported by other build systems.
This CL removes it and falls back on the more relaxed visibility value
("*" in this case).
It is not a problem because the targets that are using conditional
visibility are all marked as "testonly" and this is probably enough to
keep the build graph clean.
Bug: None
Change-Id: I2d2b5ac9a02d08c2863950116db455976ee1459c
No-Try: True
Reviewed-on: https://webrtc-review.googlesource.com/14902
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20658}
In order to create a clean CL to switch to RTC_ prefixed LOG macros
this CL runs `git cl format --full` on the files with LOG macros in
the following directories:
- modules/audio_device
- modules/media_file
- modules/video_capture
This CL has been automatically generated with:
for m in PLOG \
LOG_TAG \
LOG_GLEM \
LOG_GLE_EX \
LOG_GLE \
LAST_SYSTEM_ERROR \
LOG_ERRNO_EX \
LOG_ERRNO \
LOG_ERR_EX \
LOG_ERR \
LOG_V \
LOG_F \
LOG_T_F \
LOG_E \
LOG_T \
LOG_CHECK_LEVEL_V \
LOG_CHECK_LEVEL \
LOG
do
for d in media_file video_capture audio_device; do
cd modules/$d
git grep -l $m | grep -E "\.(cc|h|m|mm)$" | xargs sed -i "1 s/$/ /"
cd ../..
done
done
git cl format --full
Bug: webrtc:8452
Change-Id: I2858b6928e6bd79957f2e5e0b07028eb68a304b2
Reviewed-on: https://webrtc-review.googlesource.com/21322
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20613}
In https://chromium-review.googlesource.com/c/chromium/src/+/750645
Chromium started to use an ErrorProne plugin to discourage synchronized
public methods (an encourage the usage of synchronized blocks).
In order to unblock the Chromium Roll we can suppress these warnings
and decide if we want to align with Chromium on this check or ask
them to make it optional.
More details in the bug.
TBR=magjed@webrtc.org
Bug: webrtc:8491
Change-Id: Ie77a324e54aab44a4f59853959549f1d21f884a0
No-Try: True
Reviewed-on: https://webrtc-review.googlesource.com/20060
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20569}
Second attempt to land https://webrtc-review.googlesource.com/c/src/+/15481.
This time with an extra (dummy) interface to ensure that we don't
break downstream clients.
Improves native Android audio implementations.
Bug: webrtc:8453
Change-Id: I659a3013ae523a2588e4c41ca44b7d0d2d65efb7
Reviewed-on: https://webrtc-review.googlesource.com/16425
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20462}
Summary:
Adds AudioTrackStartErrorCode to separate different types of error
codes in combination with StartPlayout.
Harmonizes WebRtcAudioRecord and WebRtcAudioTrack implementations
to ensure that init/start/stop is performed identically.
Adds thread checking in WebRtcAudio track.
Bug: webrtc:8453
Change-Id: Ic913e888ff9493c9cc748a7b4dae43eb6b37fa85
Reviewed-on: https://webrtc-review.googlesource.com/15481
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Alex Glaznev <glaznev@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20448}
Fixes the warning below:
WebRtcAudioTrack.java:364: warning: [StaticAccessedFromInstance] Static method getMaxVolume
should not be accessed from an object instance; instead use AudioTrack.getMaxVolume
+ "max gain: " + audioTrack.getMaxVolume());
Bug: NONE
Change-Id: I6247584b65ac972a6a3739fba718387873964f9f
Reviewed-on: https://webrtc-review.googlesource.com/14180
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20371}
AddRef() now returns void, and Release() returns an enum
RefCountReleaseStatus, to indicate whether or not this Release
call implied deletion.
Bug: webrtc:8270
Change-Id: If2fb77f26118b61751b51c856af187c72112c630
Reviewed-on: https://webrtc-review.googlesource.com/3320
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20366}
