Remove MediaChannel usage from webrtc_video_engine_unittest

Bug: webrtc:13931
Change-Id: Ie45a25c6b204b38b749381ef5e9403cf036b8126
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/309660
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40323}
This commit is contained in:
Florent Castelli 2023-06-20 18:30:03 +00:00 committed by WebRTC LUCI CQ
parent ee97e6ad88
commit 4e434c313e
4 changed files with 644 additions and 593 deletions

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@ -43,7 +43,7 @@ class FakeNetworkInterface : public MediaChannelNetworkInterface {
recvbuf_size_(-1),
dscp_(rtc::DSCP_NO_CHANGE) {}
void SetDestination(MediaChannel* dest) { dest_ = dest; }
void SetDestination(MediaReceiveChannelInterface* dest) { dest_ = dest; }
// Conference mode is a mode where instead of simply forwarding the packets,
// the transport will send multiple copies of the packet with the specified
@ -207,7 +207,7 @@ class FakeNetworkInterface : public MediaChannelNetworkInterface {
}
webrtc::TaskQueueBase* thread_;
MediaChannel* dest_;
MediaReceiveChannelInterface* dest_;
bool conf_;
// The ssrcs used in sending out packets in conference mode.
std::vector<uint32_t> conf_sent_ssrcs_;

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@ -948,6 +948,14 @@ class VideoMediaSendChannelInterface : public MediaSendChannelInterface {
// Enable network condition based codec switching.
virtual void SetVideoCodecSwitchingEnabled(bool enabled) = 0;
virtual bool GetStats(VideoMediaSendInfo* stats) = 0;
// This fills the "bitrate parts" (rtx, video bitrate) of the
// BandwidthEstimationInfo, since that part that isn't possible to get
// through webrtc::Call::GetStats, as they are statistics of the send
// streams.
// TODO(holmer): We should change this so that either BWE graphs doesn't
// need access to bitrates of the streams, or change the (RTC)StatsCollector
// so that it's getting the send stream stats separately by calling
// GetStats(), and merges with BandwidthEstimationInfo by itself.
virtual void FillBitrateInfo(BandwidthEstimationInfo* bwe_info) = 0;
// Information queries to support SetReceiverFeedbackParameters
virtual webrtc::RtcpMode SendCodecRtcpMode() const = 0;

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@ -271,15 +271,6 @@ class VideoMediaChannel : public MediaChannel,
webrtc::VideoEncoderFactory::EncoderSelectorInterface*
encoder_selector) override {}
// This fills the "bitrate parts" (rtx, video bitrate) of the
// BandwidthEstimationInfo, since that part that isn't possible to get
// through webrtc::Call::GetStats, as they are statistics of the send
// streams.
// TODO(holmer): We should change this so that either BWE graphs doesn't
// need access to bitrates of the streams, or change the (RTC)StatsCollector
// so that it's getting the send stream stats separately by calling
// GetStats(), and merges with BandwidthEstimationInfo by itself.
void FillBitrateInfo(BandwidthEstimationInfo* bwe_info) override = 0;
// Gets quality stats for the channel.
virtual bool GetSendStats(VideoMediaSendInfo* info) = 0;
virtual bool GetReceiveStats(VideoMediaReceiveInfo* info) = 0;

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