When starting from 720p this is necessary to achieve acceptable
quality at low bitrates.
BUG=webrtc:6495
Review-Url: https://codereview.webrtc.org/2538913003
Cr-Commit-Position: refs/heads/master@{#15356}
The goal with this CL is to move implementation details out from the
webrtc root (webrtc/video_decoder.h) to simplify the dependency graph.
Another goal is to streamline the creation of VideoDecoders in
webrtcvideoengine2.cc; it will now have two factories of the same
WebRtcVideoDecoderFactory type, one internal and one external.
Specifically, this CL:
* Removes webrtc::VideoDecoder::DecoderType and use webrtc::VideoCodecType
instead.
* Removes 'static VideoDecoder* Create(DecoderType codec_type)' and
moves the create function to the internal decoder factory instead.
* Removes video_decoder.cc. webrtc::VideoDecoder is now just an
interface without any static functions.
BUG=webrtc:6743
Review-Url: https://codereview.webrtc.org/2521203002
Cr-Commit-Position: refs/heads/master@{#15350}
This is to assess the performance penalty of the (current)
lack of integration with FlexFEC and BWE.
This CL also enables send-side BWE for the following tests:
- foreman_cif_net_delay_0_0_plr_0_VP9
- foreman_cif_net_delay_0_0_plr_0_H264
- foreman_cif_delay_50_0_plr_5_VP9
- foreman_cif_delay_50_0_plr_5_H264
- foreman_cif_delay_50_0_plr_5_H264_flexfec
- foreman_cif_delay_50_0_plr_5_H264_ulpfec
Perf alerts on these tests are therefore expected.
R=stefan@webrtc.org
BUG=webrtc:5654
Review-Url: https://codereview.webrtc.org/2534203004
Cr-Commit-Position: refs/heads/master@{#15339}
It is unused since cl https://codereview.webrtc.org/2386573002.
The new denoiser implementation and its tests are kept for now. This
code is also unused, but there are some plans to take this code into
use in the not too distant future.
BUG=None
Review-Url: https://codereview.webrtc.org/2496153002
Cr-Commit-Position: refs/heads/master@{#15338}
And delete the method CongestionController::packet_router.
BUG=None
Review-Url: https://codereview.webrtc.org/2516983004
Cr-Commit-Position: refs/heads/master@{#15323}
JitterBufferDelayInMs is used for the WebRTC-NewVideoJitterBuffer finch
experiment, and therefore needs to be calculated.
BUG=webrtc:5514
Review-Url: https://codereview.webrtc.org/2534093003
Cr-Commit-Position: refs/heads/master@{#15313}
This brings QualityScaler much more in line with OveruseFrameDetector.
The two classes are conceptually similar, and should be used in the
same way. The biggest changes in this CL are:
- Quality scaling is now only done in ViEEncoder and not in each
encoder implementation separately.
- QualityScaler now checks the average QP asynchronously, instead of
having to be polled on each frame.
- QualityScaler is no longer responsible for actually scaling the frames,
but has a callback to ViEEncoder that it uses to express it's desire
for lower resolution.
BUG=webrtc:6495
Review-Url: https://codereview.webrtc.org/2398963003
Cr-Commit-Position: refs/heads/master@{#15286}
Intervals when video is paused is no longer included in the stats:
"WebRTC.Video.InputFramesPerSecond"
"WebRTC.Video.SentFramesPerSecond"
BUG=webrtc:5283
Review-Url: https://codereview.webrtc.org/2536743002
Cr-Commit-Position: refs/heads/master@{#15285}
There's no longer any need to make the two arguments have the same
signedness, so we can remove a bunch of superfluous (and sometimes
dangerous) casts.
It turned out I also had to fix the safe_cmp functions to properly handle
enums that are implicitly convertible to integers.
NOPRESUBMIT=true
BUG=webrtc:6645
Review-Url: https://codereview.webrtc.org/2534683002
Cr-Commit-Position: refs/heads/master@{#15281}
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}
Now ProbeController can send periodic bandwidth probes when in
application-limited region. This will allow to maintain correct
bottleneck bandwidth estimate, even not all bandwidth is being used.
The feature is not enabled by default, but can be enabled with a flag.
Interval between probes is currently set to 5 seconds.
BUG=webrtc:6332
Review-Url: https://codereview.webrtc.org/2504023002
Cr-Commit-Position: refs/heads/master@{#15279}
In this CL:
- EndToEndTests is now parameterized.
- Added VP8 non-rotated unittest.
- CanReceiveUlpfec/CanReceiveFlexFec now use multisets for timestamps.
- pre_decode_image_callback_ is now called before decoding a frame
with the new video jitter buffer.
- Set video rotation when FrameObjects are created.
- Calculate KeyFramesReceivedInPermille in new video jitter buffer.
BUG=webrtc:5514
Review-Url: https://codereview.webrtc.org/2522493002
Cr-Commit-Position: refs/heads/master@{#15274}
Reason for revert:
Unfortunately, this change breaks internal projects. Specifically the change to the CongestionController interface means anything implementing it will be forced to change in lock-step.
Original issue's description:
> Pass time constanct to bwe smoothing filter.
>
> BUG=webrtc:6443, webrtc:6303
>
> Committed: https://crrev.com/9abbf5ae4ec7d688a9b4aa03a405f3faadb74b90
> Cr-Commit-Position: refs/heads/master@{#15266}
TBR=minyue@webrtc.org,stefan@webrtc.org,solenberg@webrtc.org,michaelt@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6443, webrtc:6303
Review-Url: https://codereview.webrtc.org/2532993002
Cr-Commit-Position: refs/heads/master@{#15272}
Pass the selected cricket::VideoCodec to H264EncoderImpl::H264EncoderImpl. The cricket::VideoCodec contains relevant information for H264 about selected profile and packetization mode.
BUG=chromium:600254,webrtc:6402, webrtc:6337
Review-Url: https://codereview.webrtc.org/2474993002
Cr-Commit-Position: refs/heads/master@{#15270}
transport.h defines an interface for sending rtp and rtcp packets,
which is used by MediaChannel in webrtc/media/engine,
{Audio|Video}{Send|Receive}Stream and in a few other
places. It was part of the build target //webrtc:webrtc, which is a monolithic target with
all webrtc production code. This CL moves the header to its own target in webrtc/api
and deprecates the old location.
Targets in webrtc/api should in general only depend on other
targets in webrtc/api. The target webrtc/api:call_api depends on
transport.h. This change also makes webrtc/voice_engine pass GN's header
include checker and is needed in order for webrtc/api:call_api to pass
it.
transport.h will be completely removed in a follow-up CL in a few weeks
after clients have updated their includes.
NOTRY=True
BUG=webrtc:5589, webrtc:5878, webrtc:6785
Review-Url: https://codereview.webrtc.org/2426563003
Cr-Commit-Position: refs/heads/master@{#15267}
Reason for revert:
Include fix; set profile information in CreatePayloadType for video.
Original issue's description:
> Revert of Add H264 profile to webrtc::VideoCodecH264 and webrtc::VideoPayload (patchset #1 id:40001 of https://codereview.webrtc.org/2525693003/ )
>
> Reason for revert:
> The CL doesn't actually set profile information in VideoPayload.
>
> Original issue's description:
> > Add H264 profile to webrtc::VideoCodecH264 and webrtc::VideoPayload
> >
> > It's necessary to check H264 profile information as well as payload name
> > in PayloadIsCompatible in RTPPayloadRegistry.
> >
> > BUG=webrtc:6743
> >
> > Committed: https://crrev.com/bdbc4b7ef578ba1d61ceec351bc47c33da115329
> > Cr-Commit-Position: refs/heads/master@{#15248}
>
> TBR=mflodman@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:6743
>
> Committed: https://crrev.com/d7e6ccbc53fc24acdcc7507a6f3a155626473d54
> Cr-Commit-Position: refs/heads/master@{#15251}
TBR=mflodman@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6743
Review-Url: https://codereview.webrtc.org/2529153002
Cr-Commit-Position: refs/heads/master@{#15252}
Reason for revert:
The CL doesn't actually set profile information in VideoPayload.
Original issue's description:
> Add H264 profile to webrtc::VideoCodecH264 and webrtc::VideoPayload
>
> It's necessary to check H264 profile information as well as payload name
> in PayloadIsCompatible in RTPPayloadRegistry.
>
> BUG=webrtc:6743
>
> Committed: https://crrev.com/bdbc4b7ef578ba1d61ceec351bc47c33da115329
> Cr-Commit-Position: refs/heads/master@{#15248}
TBR=mflodman@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6743
Review-Url: https://codereview.webrtc.org/2529143002
Cr-Commit-Position: refs/heads/master@{#15251}
It's necessary to check H264 profile information as well as payload name
in PayloadIsCompatible in RTPPayloadRegistry.
BUG=webrtc:6743
Review-Url: https://codereview.webrtc.org/2525693003
Cr-Commit-Position: refs/heads/master@{#15248}
Reason for revert:
Downstream code has been updated.
Original issue's description:
> Revert of Remove RTPPayloadStrategy and simplify RTPPayloadRegistry (patchset #7 id:240001 of https://codereview.webrtc.org/2524923002/ )
>
> Reason for revert:
> Breaks downstream projects.
>
> Original issue's description:
> > Remove RTPPayloadStrategy and simplify RTPPayloadRegistry
> >
> > This CL removes RTPPayloadStrategy that is currently used to handle
> > audio/video specific aspects of payload handling. Instead, the audio and
> > video specific aspects will now have different functions, with linear
> > code flow.
> >
> > This CL does not contain any functional changes, and is just a
> > preparation for future CL:s.
> >
> > The main purpose with this CL is to add this function:
> > bool PayloadIsCompatible(const RtpUtility::Payload& payload,
> > const webrtc::VideoCodec& video_codec);
> > that can easily be extended in a future CL to look at video codec
> > specific information.
> >
> > BUG=webrtc:6743
> >
> > Committed: https://crrev.com/b881254dc86d2cc80a52e08155433458be002166
> > Cr-Commit-Position: refs/heads/master@{#15232}
>
> TBR=danilchap@webrtc.org,solenberg@webrtc.org,mflodman@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:6743
>
> Committed: https://crrev.com/33c81d05613f45f65ee17224ed381c6cdd1c6c6f
> Cr-Commit-Position: refs/heads/master@{#15234}
TBR=danilchap@webrtc.org,solenberg@webrtc.org,mflodman@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6743
Review-Url: https://codereview.webrtc.org/2531043002
Cr-Commit-Position: refs/heads/master@{#15245}
It will be necessary to keep the H264 profile information in
VideoReceiveStream::Decoder. I think it will be easier now and for the
future to just store all of the codec parameters unmodified in
VideoReceiveStream::Decoder instead of extracting a subset of them to an
ad hoc class.
BUG=webrtc:6743,webrtc:5948
Review-Url: https://codereview.webrtc.org/2523773003
Cr-Commit-Position: refs/heads/master@{#15239}
WebRTC.Video.BitrateSentInKbps
WebRTC.Video.MediaBitrateSentInKbps
WebRTC.Video.PaddingBitrateSentInKbps
WebRTC.Video.RetransmittedBitrateSentInKbps
WebRTC.Video.FecBitrateSentInKbps
RtpSender has two StreamDataCounters: for the non-RTX and the RTX stream.
The same counter (for the non-RTX stream) is reported for both the media SSRC and the FlexFEC SSRC.
Bitrate stats are summed for all SSRCs, thus the counter for the non-RTX stream is counted twice.
Do not store the counter for the FlexFEC SSRC.
Do not include info from FlexFEC substreams in VideoSendStream::Stats::ToString (periodically logged during a call).
BUG=webrtc:6774
Review-Url: https://codereview.webrtc.org/2525293002
Cr-Commit-Position: refs/heads/master@{#15238}
Reason for revert:
Breaks downstream projects.
Original issue's description:
> Remove RTPPayloadStrategy and simplify RTPPayloadRegistry
>
> This CL removes RTPPayloadStrategy that is currently used to handle
> audio/video specific aspects of payload handling. Instead, the audio and
> video specific aspects will now have different functions, with linear
> code flow.
>
> This CL does not contain any functional changes, and is just a
> preparation for future CL:s.
>
> The main purpose with this CL is to add this function:
> bool PayloadIsCompatible(const RtpUtility::Payload& payload,
> const webrtc::VideoCodec& video_codec);
> that can easily be extended in a future CL to look at video codec
> specific information.
>
> BUG=webrtc:6743
>
> Committed: https://crrev.com/b881254dc86d2cc80a52e08155433458be002166
> Cr-Commit-Position: refs/heads/master@{#15232}
TBR=danilchap@webrtc.org,solenberg@webrtc.org,mflodman@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6743
Review-Url: https://codereview.webrtc.org/2528993002
Cr-Commit-Position: refs/heads/master@{#15234}
This CL removes RTPPayloadStrategy that is currently used to handle
audio/video specific aspects of payload handling. Instead, the audio and
video specific aspects will now have different functions, with linear
code flow.
This CL does not contain any functional changes, and is just a
preparation for future CL:s.
The main purpose with this CL is to add this function:
bool PayloadIsCompatible(const RtpUtility::Payload& payload,
const webrtc::VideoCodec& video_codec);
that can easily be extended in a future CL to look at video codec
specific information.
BUG=webrtc:6743
Review-Url: https://codereview.webrtc.org/2524923002
Cr-Commit-Position: refs/heads/master@{#15232}
The purpose with this CL is to be able to send video codec specific
information down to RTPPayloadRegistry. We already do this for audio
with explicit arguments for e.g. number of channels. Instead of
extracting the arguments from webrtc::CodecInst (audio) and
webrtc::VideoCodec, this CL sends the types unmodified all the way down
to RTPPayloadRegistry.
This CL does not contain any functional changes, and is just a
preparation for future CL:s.
In the dependent CL https://codereview.webrtc.org/2524923002/,
RTPPayloadStrategy is removed. RTPPayloadStrategy previously handled
audio/video specific aspects of payload handling. After this CL, we will
know if we get audio or video codecs without any dependency injection,
since we have different functions with different signatures for audio
vs video.
BUG=webrtc:6743
TBR=mflodman
Review-Url: https://codereview.webrtc.org/2523843002
Cr-Commit-Position: refs/heads/master@{#15231}
Reason for revert:
Internal bots failed.
Original issue's description:
> Reland "Move smoothing filter to common audio".
>
> The original CL was this https://codereview.webrtc.org/2484153002/
>
> Due to failure with internal trial servers, it was reverted. This CL tries to reland it.
>
> BUG=webrtc:6443
>
> Committed: https://crrev.com/223641f1b903e41e77d88f03199b4cdb65255ec8
> Cr-Commit-Position: refs/heads/master@{#15227}
TBR=tommi@webrtc.org,solenberg@webrtc.org,terelius@webrtc.org,kjellander@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6443
Review-Url: https://codereview.webrtc.org/2529943002
Cr-Commit-Position: refs/heads/master@{#15228}
Previously ProbeController was starting probing as soon as SetBitrates()
is called. As result these probes would often timeout while connection
is being established. Now ProbeController receives notifications about
network route changes. This allows to start probing only when transport
is connected. This also makes it possible to restart probing whenever
transport route changes (will be done in a separate change).
BUG=webrtc:6332
R=honghaiz@webrtc.org, philipel@webrtc.org, stefan@webrtc.org
Review URL: https://codereview.webrtc.org/2458863002 .
Committed: https://crrev.com/5c99c76255ee7bface3c742c25fb5617748ac86e
Cr-Original-Commit-Position: refs/heads/master@{#15094}
Cr-Commit-Position: refs/heads/master@{#15204}
The class VideoDecoderSoftwareFallbackWrapper is an implementation
detail of webrtc/media/engine/webrtcvideoengine2.cc and should not be
directly under webrtc/video_decoder.h. The main purpose is to improve
the dependency graph in WebRTC so that VideoDecoderSoftwareFallbackWrapper
can depend on cricket::VideoCodec.
The test for VideoDecoderSoftwareFallbackWrapper is also moved from
webrtc/video/video_decoder_unittest.cc to
webrtc/media/engine/videodecodersoftwarefallbackwrapper_unittest.cc.
BUG=webrtc:6743
TBR=stefan@webrtc.org
Review-Url: https://codereview.webrtc.org/2518263003
Cr-Commit-Position: refs/heads/master@{#15180}
Reason for revert:
The WebRtcBrowserTest.NegotiateUnsupportedVideoCodec test has been fixed in Chromium with the following change:
function removeVideoCodec(offerSdp) {
- offerSdp = offerSdp.replace('a=rtpmap:100 VP8/90000\r\n',
- 'a=rtpmap:100 XVP8/90000\r\n');
+ offerSdp = offerSdp.replace(/a=rtpmap:(\d+)\ VP8\/90000\r\n/,
+ 'a=rtpmap:$1 XVP8/90000\r\n');
return offerSdp;
}
Original issue's description:
> Revert of Stop using hardcoded payload types for video codecs (patchset #6 id:210001 of https://codereview.webrtc.org/2493133002/ )
>
> Reason for revert:
> Breaks chromium.fyi test:
> WebRtcBrowserTest.NegotiateUnsupportedVideoCodec
>
> Original issue's description:
> > Stop using hardcoded payload types for video codecs
> >
> > This CL stops using hardcoded payload types for different video codecs
> > and will dynamically assign them payload types incrementally from 96 to
> > 127 instead.
> >
> > This CL:
> > * Replaces 'std::vector<VideoCodec> DefaultVideoCodecList()' in
> > webrtcvideoengine2.cc with an explicit WebRtcVideoEncoderFactory for
> > internally supported software codecs instead. The purpose is to
> > streamline the payload type assignment in webrtcvideoengine2.cc which
> > will now have two encoder factories of the same
> > WebRtcVideoEncoderFactory type; one internal and one external.
> > * Removes webrtc::VideoEncoder::EncoderType and use cricket::VideoCodec
> > instead.
> > * Removes 'static VideoEncoder* Create(EncoderType codec_type)' and
> > moves the create function to the internal encoder factory instead.
> > * Removes video_encoder.cc. webrtc::VideoEncoder is now just an
> > interface without any static functions.
> > * The function GetSupportedCodecs in webrtcvideoengine2.cc unifies
> > the internal and external codecs and assigns them payload types
> > incrementally from 96 to 127.
> > * Updates webrtcvideoengine2_unittest.cc and removes assumptions about
> > what payload types will be used.
> >
> > BUG=webrtc:6677,webrtc:6705
> > R=hta@webrtc.org, ossu@webrtc.org, stefan@webrtc.org
> >
> > Committed: https://crrev.com/42043b95872b51321f508bf255d804ce3dff366b
> > Cr-Commit-Position: refs/heads/master@{#15135}
>
> TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:6677,webrtc:6705
>
> Committed: https://crrev.com/eacbaea920797ff751ca83050d140821f5055591
> Cr-Commit-Position: refs/heads/master@{#15140}
TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6677,webrtc:6705
Review-Url: https://codereview.webrtc.org/2511933002
Cr-Commit-Position: refs/heads/master@{#15148}
PayloadRouter::OnEncodedImage() was casing boolean result from
SendOutgoingData() to int, and then not handling it correctly, which
results in all errors in SendOutgoingData() being ignored. This issue
was introduced in
https://chromium.googlesource.com/external/webrtc/+/ad34dbe934
This bug masked another issue with VP9 codec (see
crbug.com/webrtc/6723 ) and that increased number of dropped frames.
BUG=634816
Review-Url: https://codereview.webrtc.org/2512543002
Cr-Commit-Position: refs/heads/master@{#15143}
Reason for revert:
Breaks chromium.fyi test:
WebRtcBrowserTest.NegotiateUnsupportedVideoCodec
Original issue's description:
> Stop using hardcoded payload types for video codecs
>
> This CL stops using hardcoded payload types for different video codecs
> and will dynamically assign them payload types incrementally from 96 to
> 127 instead.
>
> This CL:
> * Replaces 'std::vector<VideoCodec> DefaultVideoCodecList()' in
> webrtcvideoengine2.cc with an explicit WebRtcVideoEncoderFactory for
> internally supported software codecs instead. The purpose is to
> streamline the payload type assignment in webrtcvideoengine2.cc which
> will now have two encoder factories of the same
> WebRtcVideoEncoderFactory type; one internal and one external.
> * Removes webrtc::VideoEncoder::EncoderType and use cricket::VideoCodec
> instead.
> * Removes 'static VideoEncoder* Create(EncoderType codec_type)' and
> moves the create function to the internal encoder factory instead.
> * Removes video_encoder.cc. webrtc::VideoEncoder is now just an
> interface without any static functions.
> * The function GetSupportedCodecs in webrtcvideoengine2.cc unifies
> the internal and external codecs and assigns them payload types
> incrementally from 96 to 127.
> * Updates webrtcvideoengine2_unittest.cc and removes assumptions about
> what payload types will be used.
>
> BUG=webrtc:6677,webrtc:6705
> R=hta@webrtc.org, ossu@webrtc.org, stefan@webrtc.org
>
> Committed: https://crrev.com/42043b95872b51321f508bf255d804ce3dff366b
> Cr-Commit-Position: refs/heads/master@{#15135}
TBR=hta@webrtc.org,stefan@webrtc.org,ossu@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6677,webrtc:6705
Review-Url: https://codereview.webrtc.org/2513633002
Cr-Commit-Position: refs/heads/master@{#15140}
This CL stops using hardcoded payload types for different video codecs
and will dynamically assign them payload types incrementally from 96 to
127 instead.
This CL:
* Replaces 'std::vector<VideoCodec> DefaultVideoCodecList()' in
webrtcvideoengine2.cc with an explicit WebRtcVideoEncoderFactory for
internally supported software codecs instead. The purpose is to
streamline the payload type assignment in webrtcvideoengine2.cc which
will now have two encoder factories of the same
WebRtcVideoEncoderFactory type; one internal and one external.
* Removes webrtc::VideoEncoder::EncoderType and use cricket::VideoCodec
instead.
* Removes 'static VideoEncoder* Create(EncoderType codec_type)' and
moves the create function to the internal encoder factory instead.
* Removes video_encoder.cc. webrtc::VideoEncoder is now just an
interface without any static functions.
* The function GetSupportedCodecs in webrtcvideoengine2.cc unifies
the internal and external codecs and assigns them payload types
incrementally from 96 to 127.
* Updates webrtcvideoengine2_unittest.cc and removes assumptions about
what payload types will be used.
BUG=webrtc:6677,webrtc:6705
R=hta@webrtc.org, ossu@webrtc.org, stefan@webrtc.org
Review URL: https://codereview.webrtc.org/2493133002 .
Cr-Commit-Position: refs/heads/master@{#15135}
This is a refactoring change in preparation for enabling AudioMixer
with the goal to have a small CL as possible for passing audio through
the new audio mixer in WebRTC. The dependent CL https://codereview.webrtc.org/2436033002/
enables the mixer.
An object of class AudioState is shared across different webrtc audio
connections. It is created in tests and in
WebRTCVoiceEngine. AudioState is constructed by passing a Config
struct, where one argument is scoped_refptr<AudioMixer>.
Populating this field has now been mandatory. Tests and
WebRTCVoiceEngine create and pass either a AudioMixerImpl.
WebRTCVoiceEngine passes a real AudioMixer, which is
currently unused.
An alternative would have tests pass a mocked audio mixer. We
chose not to do that, because we believe that tests should use
the real thing unless there are reasons against it. Construction
time is not an issue, because the real mixer is relatively
lightweight.
We couldn't find a way to test any mixer-related changes in AudioState
before the mixes is connected. The next dependent CL
https://codereview.webrtc.org/2436033002/ contains unit tests for
mixer usage.
BUG=webrtc:6346
Review-Url: https://codereview.webrtc.org/2469743002
Cr-Commit-Position: refs/heads/master@{#15134}
This CL adds full stack tests that are used to measure the performance
of H264 with and without FlexFEC. In order to not increase the bot
run time, the CL also reduces the test time to 45 secs. This should
be OK, since the BWE is faster to ramp up nowadays.
Due to the test time change, there may be some performance alerts.
BUG=webrtc:5654
Review-Url: https://codereview.webrtc.org/2499273002
Cr-Commit-Position: refs/heads/master@{#15118}
