I used a command like this to update the paths:
perl -pi -e "s/webrtc\/base/webrtc\/rtc_base/g" `find webrtc/rtc_base -name "*.cc" -o -name "*.h"`
The only manual edit is to add an include of webrtc/rtc_base/checks.h in
webrtc/modules/audio_device/android/opensles_common.h, which likely
was needed due to changed include paths due to 'git cl format'.
BUG=webrtc:7634
NOTRY=True
NOPRESUBMIT=True
Review-Url: https://codereview.webrtc.org/2969653002
Cr-Commit-Position: refs/heads/master@{#18871}
const int16_t* data() const;
int16_t* mutable_data();
- data() returns a zeroed static buffer on muted frames (to avoid unnecessary zeroing of the member buffer) and directly returns AudioFrame::data_ on unmuted frames.
- mutable_data(), lazily zeroes AudioFrame::data_ if the frame is currently muted, sets muted=false, and returns AudioFrame::data_.
These accessors serve to "force" callers to be aware of the mute state field, i.e. lazy zeroing is not the primary motivation.
This change only optimizes handling of muted frames where it is somewhat trivial to do so. Other improvements requiring more significant structural changes will come later.
BUG=webrtc:7343
TBR=henrika
Review-Url: https://codereview.webrtc.org/2750783004
Cr-Commit-Position: refs/heads/master@{#18543}
(Or, in less flattering terms, fixing a performance issue introduced
a few months ago by me).
In GN release mode (is_debug = false), the version of the mixer code
before this CL generated code that multiplied each sample (tens of
thousands/second for each input stream) with a floating point number.
This number is almost always exactly 1.0f. The only situation when it's
not 1 is when an audio steam is added or removed.
For one input stream early return leads to a 30% improvement of audio
mixing time profiled on x86-64 under a release build (is_debug = false,
enable_profiling, enable_full_stack_frames_for_profiling) with 16kHz and no
APM limiter. There can be up to 3 streams.
BUG=chromium:687502
Review-Url: https://codereview.webrtc.org/2659423002
Cr-Commit-Position: refs/heads/master@{#16396}
This CL is in preparation to move the AudioFrame into webrtc/api. The
AudioFrame is a POD type used for representing 10ms of audio. It
appears as a parameter and return value of interfaces being migrated
to webrtc/api, in particular AudioMixer.
Here, methods operator+=, operator>>=, Mute are
moved into a new target webrtc/audio/utility/audio_frame_operations,
and dependencies are changed to use
the new versions. The old AudioFrame methods are marked deprecated.
The audio frame utilities in webrtc/modules/utility:audio_frame_operations
are also moved to the new location.
TBR=kjellander@webrtc.org
BUG=webrtc:6548
NOPRESUBMIT=True
Review-Url: https://codereview.webrtc.org/2424173003
Cr-Commit-Position: refs/heads/master@{#15413}
There's no longer any need to make the two arguments have the same
signedness, so we can drop the "u" suffix on literal integer
arguments.
NOPRESUBMIT=true
BUG=webrtc:6645
Review-Url: https://codereview.webrtc.org/2535593002
Cr-Commit-Position: refs/heads/master@{#15280}
The AudioMixer is now split in a mixer and audio source interface part, which has moved to webrtc/api, and a default implementation part, which lies in webrtc/modules.
This change makes it possible to create other mixer implementations and is a first step to facilitate passing down a mixer from outside of WebRTC.
It will also create less build dependencies when the new mixer has replaced the old one.
NOTRY=True
TBR=henrik.lundin@webrtc.org
BUG=webrtc:6346
Review-Url: https://codereview.webrtc.org/2411313003
Cr-Commit-Position: refs/heads/master@{#14705}
MixerAudioSource is moved to AudioMixerImpl::Source. Structures and methods of the MixerAudioSource interface have been renamed. The RemixFrame method has added checks and is moved to audio_frame_manipulator.h
BUG=webrtc:6346
Review-Url: https://codereview.webrtc.org/2396803004
Cr-Commit-Position: refs/heads/master@{#14600}
Reason for revert:
breaks chromium FYI
Original issue's description:
> Made MixerAudioSource a pure interface.
>
> This required quite a few small changes in the mixing algorithm
> structure, the mixer interface and the mixer unit tests.
>
> BUG=webrtc:6346
>
> Committed: https://crrev.com/2ae5fdff86b784545cbd724de54bb5ffedde1adf
> Cr-Commit-Position: refs/heads/master@{#14567}
TBR=ivoc@webrtc.org,solenberg@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6346
Review-Url: https://codereview.webrtc.org/2394253003
Cr-Commit-Position: refs/heads/master@{#14568}
This required quite a few small changes in the mixing algorithm
structure, the mixer interface and the mixer unit tests.
BUG=webrtc:6346
Review-Url: https://codereview.webrtc.org/2396483002
Cr-Commit-Position: refs/heads/master@{#14567}
The folder structure is now as was agreed on in the 'Slim and Modular
WebRTC' effort. Also added some dependencies that were previously in
another part of the tree.
NOTRY=True
Review-Url: https://codereview.webrtc.org/2238803002
Cr-Commit-Position: refs/heads/master@{#13742}
