Now the templated AudioEncoderFactory can create Opus encoders!
BUG=webrtc:7831
Review-Url: https://codereview.webrtc.org/2930243003
Cr-Commit-Position: refs/heads/master@{#18645}
Now the templated AudioEncoderFactory can create G722 encoders!
BUG=webrtc:7833
Review-Url: https://codereview.webrtc.org/2934833002
Cr-Commit-Position: refs/heads/master@{#18644}
Now the templated AudioDecoderFactory can create G722 decoders!
BUG=webrtc:7839
Review-Url: https://codereview.webrtc.org/2940833002
Cr-Commit-Position: refs/heads/master@{#18643}
In this case it wasn't an issue, because only one result would be found
by remove_if, but might as well fix it just in case.
BUG=None
TBR=pthatcher@webrtc.org
Review-Url: https://codereview.webrtc.org/2945723002
Cr-Commit-Position: refs/heads/master@{#18641}
No real encoder implements the correct API yet, so we're just testing
dummies.
BUG=webrtc:7823
Review-Url: https://codereview.webrtc.org/2935643002
Cr-Commit-Position: refs/heads/master@{#18637}
Adds the VideoEncoderFactory interface and implements it for use with HardwareVideoEncoder. This uses MediaCodecVideoEncoder's initialization code as an example.
BUG=webrtc:7760
Change-Id: I9fbc93ce9ac4ad866750a4386c4f15e800a3073e
Reviewed-on: https://chromium-review.googlesource.com/530063
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18636}
1. To make the files conform to chromium-style guidelines, and stop the compiler from complaing:
1.1. Move constructors out of .h file.
1.2. Move destructors out of .h file.
1.3. Move virtual functions out of .h file.
2. BlockLength() and Create() did not have consistent access modifiers in the various subclasses of RtcpPacket. Change the access level to public throughout.
3. Reorder BlockLength() and Create() where necessary, to reflect the order defined in the parent class (RtcpPacket).
BUG=None
Review-Url: https://codereview.webrtc.org/2937403002
Cr-Commit-Position: refs/heads/master@{#18633}
These methods have the same behavior as their counterparts in std::optional, except that rtc::Optional::value() requires that the value exists whereas std::optional::value() throws an exception.
BUG=webrtc:7843
Review-Url: https://codereview.webrtc.org/2942203002
Cr-Commit-Position: refs/heads/master@{#18631}
There are some functions in packet_router.cc and modules/congestion_controller that could be used by different threads, but they're protected using rtc::ThreadChecker which doesn't allow them to be called by more than one thread even if the calls are synchronised. This CL replaces those with rtc::RaceChecker, which allows serialized access of the functions from multiple threads.
BUG=webrtc:7826
Review-Url: https://codereview.webrtc.org/2940133003
Cr-Commit-Position: refs/heads/master@{#18628}
All setting switches except "Loopback mode" is now in the Settings
screen instead of the main screen. They are also persisted across app
launches.
Bug: webrtc:7748
Change-Id: Iafd84e5e39639770118e2503148d1bf7fb9c3d8d
Reviewed-on: https://chromium-review.googlesource.com/527034
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18626}
Guarded by field trial - similar to high profile encoder.
If high profile is requested, but device do not support it
then fallback to baseline profile.
BUG=b/34816463
Review-Url: https://codereview.webrtc.org/2936313002
Cr-Commit-Position: refs/heads/master@{#18619}
This CL makes the WebRTC more modular and allows the users to build
WebRTC without audio and video(DataChannel only).
The BUILD files in call/, logging/, media/ and pc/ are modified to
support modular WebRTC.
The dependencies on Call and RtcEventLog are removed from the
PeerConnection. Instead of being created internally, they would be
passed in by the PeerConnectionFactory.
Add the CreateModularPeerConnectionFactory function which allow the
users to create a PeerConnectionFactory with the modules they need.
If the users want to build WebRTC without audio and video, they can
pass in null pointers for modules they don't need. (MediaEngine,
VideoEncoderFactory etc.)
BUG=webrtc:7613
Review-Url: https://codereview.webrtc.org/2854123003
Cr-Commit-Position: refs/heads/master@{#18617}
This eliminates a thread hop in PeerConnectionFactory initialization,
and will allow some code to be simplified.
BUG=None
Review-Url: https://codereview.webrtc.org/2934103002
Cr-Commit-Position: refs/heads/master@{#18613}
Currently there is a hard limit for the estimated captured frame
interval of 45ms. As the encoder utilization is calculated as
(input frame interval)/(encode time), overuse signals can be triggered
even though there is plenty of time to go around if the fps is low.
However, in order to avoid falsly estimating low encode usage in case
the capturer has a dynamic frame rate, set the frame interval based on
the actual current max framerate.
BUG=webrtc:4172
Review-Url: https://codereview.webrtc.org/2918143003
Cr-Commit-Position: refs/heads/master@{#18610}
The following changes have been made:
- command line args wired,
- user output added,
- final polishing.
BUG=webrtc:7218
Review-Url: https://codereview.webrtc.org/2808053002
Cr-Commit-Position: refs/heads/master@{#18609}
Reason for revert:
Build file causing google3 compilation error fixed
Original issue's description:
> Revert of Conversational speech tool, simualtor + unit tests (patchset #12 id:220001 of https://codereview.webrtc.org/2790933002/ )
>
> Reason for revert:
> Compile Error.
>
> Original issue's description:
> > The simulator puts into action the schedule of speech turns encoded in a MultiEndCall instance. The output is a set of audio track pairs. There is one set for each speaker and each set contains one near-end and one far-end audio track. The tracks are directly written into wav files instead of creating them in memory. To speed up the creation of the output wav files, *all* the source audio tracks (i.e., the atomic speech turns) are pre-loaded.
> >
> > The ConversationalSpeechTest.MultiEndCallSimulator unit test defines a conversational speech sequence and creates two wav files (with pure tones at 440 and 880 Hz) that are used as atomic speech turn tracks.
> >
> > This CL also patches MultiEndCall in order to allow input audio tracks with same sample rate and single channel only.
> >
> > BUG=webrtc:7218
> >
> > Review-Url: https://codereview.webrtc.org/2790933002
> > Cr-Commit-Position: refs/heads/master@{#18480}
> > Committed: 6b648c4697
>
> TBR=minyue@webrtc.org,alessiob@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:7218
>
> Review-Url: https://codereview.webrtc.org/2925123003
> Cr-Commit-Position: refs/heads/master@{#18481}
> Committed: 4c72cf43dfTBR=minyue@webrtc.org,charujain@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7218
Review-Url: https://codereview.webrtc.org/2930853002
Cr-Commit-Position: refs/heads/master@{#18606}
Currently the debug dump functionality of WebRTC (a log of all
AudioProcessing operations) was tested by the following tests:
1. ApmTest.VerifyDebugDump* which configures and runs AudioProcessing
from a debug dump, and verifies that the same debug dump is
recorded.
2. DebugDumpTest.* which is a comprehensive test of the debug dump
operations. AudioProcessing configuration is changed, and the dump
is scanned for the change.
3. ApmTest::{DebugDump, DebugDumpFromFileHandle} that verify that
debug dumping can be started and files written.
This CL replaces the debug dump mechanism in all these tests to
webrtc::AecDump. Some of the tests are adapted to the chenges of the
new API to AecDump {Start,Stop}DebugRecording: the old functions
signal errors when a file cannot be opened. With AecDump, the
AecDumpFactory instead returns a nullptr.
The CL also changes audioproc_f to use AecDump.
BUG=webrtc:7404
Review-Url: https://codereview.webrtc.org/2864373002
Cr-Commit-Position: refs/heads/master@{#18605}
This functionality is needed when sending C++ I420 buffers to Java
VideoSinks or Java encoders.
Bug: webrtc:7749
Change-Id: Ied783470b90b9d2e0cb5930795f35de4a296d499
Reviewed-on: https://chromium-review.googlesource.com/532961
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#18597}
