| Commit message (Collapse) | Author | Age | Files | Lines |
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It works like this on the browser side:
A new object UserInputMonitor is created on BrowserMainLoop and passed to AudioInputRendererHost to pass to AudioInputController.
AudioInputController::DoRecord calls UserInputMonitor::AddKeyStrokeListener --> UserInputMonitor listens to system key events (only implemented on Linux) --> AudioInputController::OnKeyPressed is called and sets key_pressed_ --> When AudioInputController::OnData called, it writes key_pressed_ to shared memory along with the audio data buffer.
On the renderer side a new param "key_pressed" is added through the code path of passing the flag to the webrtc voice engine.
This CL includes all these changes except the implementation of UserInputMonitor for Windows and Mac. The impl of UserInputMonitor is mostly copied from remoting/host/local_input_monitor_linux.cc
BUG=
Committed: https://src.chromium.org/viewvc/chrome?view=rev&revision=217768
Review URL: https://chromiumcodereview.appspot.com/21183002
git-svn-id: svn://svn.chromium.org/chrome/trunk/src@217844 0039d316-1c4b-4281-b951-d872f2087c98
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> Adding key press detection in the browser process.
> It works like this on the browser side:
> A new object KeyPressMonitor is created on BrowserMainLoop and passed to AudioInputRendererHost to pass to AudioInputController.
> AudioInputController::DoRecord calls KeyPressMonitor::AddKeyPressListener --> KeyPressMonitor listens to system key events through UserInputMonitor(only implemented on Linux) --> AudioInputController::OnKeyPressed is called and sets key_pressed_ --> When AudioInputController::OnData called, it writes key_pressed_ to shared memory along with the audio data buffer.
> On the renderer side a new param "key_pressed" is added through the code path of passing the flag to the webrtc voice engine.
> This CL includes all these changes except the implementation of UserInputMonitor for Windows and Mac. The impl of UserInputMonitor is mostly copied from remoting/host/local_input_monitor_linux.cc
>
>
> BUG=
>
> Review URL: https://chromiumcodereview.appspot.com/21183002
TBR=jiayl@chromium.org
Review URL: https://codereview.chromium.org/22871007
git-svn-id: svn://svn.chromium.org/chrome/trunk/src@217774 0039d316-1c4b-4281-b951-d872f2087c98
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It works like this on the browser side:
A new object KeyPressMonitor is created on BrowserMainLoop and passed to AudioInputRendererHost to pass to AudioInputController.
AudioInputController::DoRecord calls KeyPressMonitor::AddKeyPressListener --> KeyPressMonitor listens to system key events through UserInputMonitor(only implemented on Linux) --> AudioInputController::OnKeyPressed is called and sets key_pressed_ --> When AudioInputController::OnData called, it writes key_pressed_ to shared memory along with the audio data buffer.
On the renderer side a new param "key_pressed" is added through the code path of passing the flag to the webrtc voice engine.
This CL includes all these changes except the implementation of UserInputMonitor for Windows and Mac. The impl of UserInputMonitor is mostly copied from remoting/host/local_input_monitor_linux.cc
BUG=
Review URL: https://chromiumcodereview.appspot.com/21183002
git-svn-id: svn://svn.chromium.org/chrome/trunk/src@217768 0039d316-1c4b-4281-b951-d872f2087c98
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stream.
Using a string device_id via IPC message from the render client to open the device is not very safe since IPC message is not trusted. Instead, we should just pass an int (session_id) to the browser, and we do the lookup on AudioInputDeviceManager in the browser to re-map the session_id to the device_id.
BUG=179597,216952
Review URL: https://chromiumcodereview.appspot.com/12440027
git-svn-id: svn://svn.chromium.org/chrome/trunk/src@189342 0039d316-1c4b-4281-b951-d872f2087c98
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We add smarts into MediaStreamDependencyFactory::CreateNativeLocalMediaStream()
to handle MediaStreams originating from WebAudio.
Please see companion WebKit patches:
https://bugs.webkit.org/show_bug.cgi?id=101815
https://bugs.webkit.org/show_bug.cgi?id=106053
BUG=none
TEST=manual test
http://www.corp.google.com/~henrika/WebAudio/MediaStreamAudioDestinationNode.html
Review URL: https://chromiumcodereview.appspot.com/11369171
git-svn-id: svn://svn.chromium.org/chrome/trunk/src@177330 0039d316-1c4b-4281-b951-d872f2087c98
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This capturer contains a source (AudioInputDevice) and a sink (WebRtcAudioDeviceImpl) by default. What it does is to get data from AudioInputDevice via CaptureCallback::Capture() callback,
and forward the data to WebRtcAudioDeviceImpl.
The source can be over written by API:
SetCapturerSource(media::AudioCapturerSource* source)
This capture currently only support one sink (WebRtcAudioDeviceImpl), but this can be extended to multiple sinks for the future.
Design doc: https://docs.google.com/a/google.com/document/d/1FmiXtk1pxFlAw_CWwbfG-EQ4Syi4vpnZm6GWMyJ1UfU/edit
TBR=tommi@chromium.org
BUG=157306
TEST=manual test
Review URL: https://codereview.chromium.org/11366244
git-svn-id: svn://svn.chromium.org/chrome/trunk/src@168242 0039d316-1c4b-4281-b951-d872f2087c98
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WebRtcAudioCapturer.
This CL failed the TabCapture browser-test on the win_rel try-bot, and caused it to fail on Mac & Windows on the build-bots.
BUG=160782
This capturer contains a source (AudioInputDevice) and a sink (WebRtcAudioDeviceImpl) by default. What it does is to get data from AudioInputDevice via CaptureCallback::Capture() callback,
and forward the data to WebRtcAudioDeviceImpl.
The source can be over written by API:
SetCapturerSource(media::AudioCapturerSource* source)
This capture currently only support one sink (WebRtcAudioDeviceImpl), but this can be extended to multiple sinks for the future.
Design doc: https://docs.google.com/a/google.com/document/d/1FmiXtk1pxFlAw_CWwbfG-EQ4Syi4vpnZm6GWMyJ1UfU/edit
BUG=157306
TEST=manual test
Review URL: https://codereview.chromium.org/11231003
TBR=xians@chromium.org
Review URL: https://codereview.chromium.org/11362231
git-svn-id: svn://svn.chromium.org/chrome/trunk/src@167448 0039d316-1c4b-4281-b951-d872f2087c98
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This capturer contains a source (AudioInputDevice) and a sink (WebRtcAudioDeviceImpl) by default. What it does is to get data from AudioInputDevice via CaptureCallback::Capture() callback,
and forward the data to WebRtcAudioDeviceImpl.
The source can be over written by API:
SetCapturerSource(media::AudioCapturerSource* source)
This capture currently only support one sink (WebRtcAudioDeviceImpl), but this can be extended to multiple sinks for the future.
Design doc: https://docs.google.com/a/google.com/document/d/1FmiXtk1pxFlAw_CWwbfG-EQ4Syi4vpnZm6GWMyJ1UfU/edit
BUG=157306
TEST=manual test
Review URL: https://codereview.chromium.org/11231003
git-svn-id: svn://svn.chromium.org/chrome/trunk/src@167387 0039d316-1c4b-4281-b951-d872f2087c98
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