Restructuring of the existing code no functional change.
MozReview-Commit-ID: FARNItrajfD
--HG--
extra : rebase_source : 0887c6546bb43538e32b99950e2febca77a32bfb
This wires up the disabling of a track with actually stopping the device if we
allow it.
This is possible for:
- Camera (enabled by default, controlled by pref
"media.getusermedia.camera.off_while_disabled.enabled")
- Microphone (disabled by default, controlled by pref
"media.getusermedia.microphone.off_while_disabled.enabled")
Screen-, app-, or windowsharing is not supported at this time.
On disabling, there's a delay before the device is ordered to stop. This is
now defaulting to 3 seconds but can be overriden by prefs
"media.getusermedia.camera.off_while_disabled.delay_ms" and
"media.getusermedia.microphone.off_while_disabled.delay_ms".
The delay is in place to prevent misuse by malicious sites. If a track is
re-enabled before the delay has passed, the device will not be touched until
another disable followed by the full delay happens.
MozReview-Commit-ID: D4nZWzrYZGm
--HG--
extra : rebase_source : 6a54fa450bd435ed65de2a30b66d25f4a5e8241e
This is the larger change for this bug. In order to turn off a device on
disabling we want to Stop() it without ending the attached track.
To allow this, this patch breaks out track-creation from Start() to SetTrack()
and moves track-ending logic from Stop() to Deallocate().
It is a programming error to Start() or Stop() a MediaEngineSource that hasn't
seen a SetTrack().
MozReview-Commit-ID: 3KzmuDjCAH0
--HG--
extra : rebase_source : 361d9b9c2a818ce51fa90d88950d5992c51407c6
The scope of flattening this hierarchy quickly grows large, so this patch does
a couple more things:
- Creates a pure interface MediaEngineSourceInterface and a base class
MediaEngineSource with common defaults and refcount support (no state!)
- Breaks out some of the helper classes to dedicated files, e.g.,
AllocationHandle, MediaEnginePrefs.
- Clarifies the threading model (written on one thread *and* under lock,
read under either)
- Fixes style, indentation, include-sorting in the affected files
- Adds comments, especially for clarifying what responsibilities methods have,
and thread usage of class members
- Changes Monitors to Mutexes since we only use them as Mutexes anyhow
- Makes MediaEngineRemoteVideoSource no longer a shared source since we now
support scaling in this source and CamerasChild can act as a broker of frames.
This greatly simplifies it. The only shared source is now
MediaEngineWebRTCMicrophoneSource, so the sharing specific common methods have
been moved to that source.
MozReview-Commit-ID: KeVZQo6gLm2
--HG--
rename : dom/media/webrtc/MediaEngine.h => dom/media/webrtc/MediaEnginePrefs.h
extra : rebase_source : c785a5feb896312912170475d6b8d997e712e48f
This SourceMediaStream is an existing hack to fit AudioCapture into
MediaManager's gUM flow. With the invariant that NotifyPull must produce data
we need to produce silence here.
Best of all would be if this SourceMediaStream was not needed at all, but let's
look at that another time.
MozReview-Commit-ID: J3EeIut1fgy
--HG--
extra : rebase_source : 8b84d8cf0f8d359d3b1c054146b28d494ed12053
This also is long, but simple.
First, we switch to floats everywhere. This allows to work with any rate, is
more flexible with channel layout, and is a stable API (see audio_processing.h
in webrtc.org).
Then, 10ms worth of audio (already at the graph rate) are poped from the
lock-free queue (fed on the other end by the MSG mixer), and does the following:
- Down mixing to stereo (if needed)
- De-interleaving into planar buffer
- Prepare input and output config
- Actually make the API call
- Free the data
Now, first, we should use a ring buffer, and not have to free any data. Then we
also should not use a lock-free queue, and synchronously process the
reverse-stream, but this is enough code already.
Then, the actual mic data processing:
- Pop a packet from the packetizer (that gives us 10ms worth of audio, note that
we switch from int16_t to float, i.e. we don't do this conversion anymore).
- We convert to planar buffers, deinterleaving
- Prepare input and output config
- Allocate a SharedBuffer of the right size
- Process the data with the processing algorithm selected in UpdateSingleSource
- Append to the a MediaSegment, and append to the right MediaStreamTrack for the
correct SourceMediaStream (the data is already planar and all well).
MozReview-Commit-ID: 2IjgHP0GAmw
--HG--
extra : rebase_source : 1e08c4a781db8778e0532f9ef1a8e369513a2c66
extra : source : 0107b3feb84bbe0e643f505ec58e303dfd94e1a7
This part is about setting on/off audio processing feature. It's long, but
it's mostly mechanichal changes, from the old API to the new one.
This also covers reseting the processing in case of device changes (with
macros).
MozReview-Commit-ID: EI2TxHRicEr
--HG--
extra : rebase_source : 5c389e00019633d371d74cdd2d881dab4d353848
extra : source : 2c7a56648de9125ae1893d54ec011b6cbb181d86
This needs the next patches to build fine, but is split out for the review.
A side effect of this patch is to break non-duplex, making the whole
init/cleanup phase much simpler.
MozReview-Commit-ID: Caqc8v7CWwZ
--HG--
extra : rebase_source : 604551cc937ee60064a263ebb5fb1550fa9a9e9f
extra : source : a781b123b252b464f805674144cc01d9dd69c391
This also is long, but simple.
First, we switch to floats everywhere. This allows to work with any rate, is
more flexible with channel layout, and is a stable API (see audio_processing.h
in webrtc.org).
Then, 10ms worth of audio (already at the graph rate) are poped from the
lock-free queue (fed on the other end by the MSG mixer), and does the following:
- Down mixing to stereo (if needed)
- De-interleaving into planar buffer
- Prepare input and output config
- Actually make the API call
- Free the data
Now, first, we should use a ring buffer, and not have to free any data. Then we
also should not use a lock-free queue, and synchronously process the
reverse-stream, but this is enough code already.
Then, the actual mic data processing:
- Pop a packet from the packetizer (that gives us 10ms worth of audio, note that
we switch from int16_t to float, i.e. we don't do this conversion anymore).
- We convert to planar buffers, deinterleaving
- Prepare input and output config
- Allocate a SharedBuffer of the right size
- Process the data with the processing algorithm selected in UpdateSingleSource
- Append to the a MediaSegment, and append to the right MediaStreamTrack for the
correct SourceMediaStream (the data is already planar and all well).
MozReview-Commit-ID: 2IjgHP0GAmw
--HG--
extra : rebase_source : d2245037e8ee7145af7eef528dcee50817b69d83
extra : histedit_source : 79443c35b82d3bc8833d140dd5afc882b85b4c12
This part is about setting on/off audio processing feature. It's long, but
it's mostly mechanichal changes, from the old API to the new one.
This also covers reseting the processing in case of device changes (with
macros).
MozReview-Commit-ID: EI2TxHRicEr
--HG--
extra : rebase_source : 7044c2d1695cdf0d6a69b4faa19349e3261ef204
extra : histedit_source : f5ac61e7b90ab4d5280623095c443529fb36cde5%2C5c969f1833bdc425842f945a5a8a4702ca13cd56
This needs the next patches to build fine, but is split out for the review.
A side effect of this patch is to break non-duplex, making the whole
init/cleanup phase much simpler.
MozReview-Commit-ID: Caqc8v7CWwZ
--HG--
extra : rebase_source : 6e7d501ef99f3ea5d755a610238b8f260194bba0
extra : histedit_source : 298c7e95a2bd40e8f9ce014e06faad159fca513e
The VoEHardware interface has been removed upstream. This replaces it with
calls to the AudioDeviceModule through VoEBase. This code is only used on the
non-full duplex code path.
MozReview-Commit-ID: m5ftmXm3CS
--HG--
extra : rebase_source : ee99440fa6616bc52d42c01b6b264c95e8b2ad89
