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Revert "Fix target bitrate RTCP messages behavior for SVC streams"

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This reverts commit ab65d8aab5fe63619033371fca1ce2711c2c2137.

Reason for revert: Fails video_engine_tests ExtendedReportsEndToEndTest.TestExtendedReportsCanSignalZeroTargetBitrate
https://ci.chromium.org/p/webrtc/builders/ci/Linux%20MSan/18366

Original change's description:
> Fix target bitrate RTCP messages behavior for SVC streams
>
> Before this CL for SVC streams (e.g VP9) still 3 separate RTP_RTCP senders
> were created. The RTCP target bitrate messages were treated as simulcast
> and were split and send for each separate spatial layer in a separate SSRC.
>
> To fix that an svc flag is now wired to VideoSendStream config
> and filled based on the encoder config in WebrtcVideoEngine. This flag is
> used to differentiate between simulcast and SVC mode in RtpVideoSender.
>
> Bug: webrtc:10485
> Change-Id: Ifa01d12a7d4f01fcbe448ad11e0cc39ab2d1df55
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/129929
> Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27345}

TBR=ilnik@webrtc.org,nisse@webrtc.org,sprang@webrtc.org

Change-Id: I184f87289d9dccc67de165038d76a5690158a3b5
No-Tree-Checks: True
No-Try: True
Bug: webrtc:10485
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130467
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27355}
This commit is contained in:
Oleh Prypin
2019-03-29 15:33:01 +00:00
committed by Commit Bot
parent e8bc3a0a5a
commit e8964903a9
12 changed files with 6 additions and 24 deletions
Download Patch File Download Diff File Expand all files Collapse all files

3
call/rtp_transport_controller_send.cc
View File

@ -104,14 +104,13 @@ RtpVideoSenderInterface* RtpTransportControllerSend::CreateRtpVideoSender(
const RtpConfig& rtp_config,
int rtcp_report_interval_ms,
Transport* send_transport,
bool is_svc,
const RtpSenderObservers& observers,
RtcEventLog* event_log,
std::unique_ptr<FecController> fec_controller,
const RtpSenderFrameEncryptionConfig& frame_encryption_config) {
video_rtp_senders_.push_back(absl::make_unique<RtpVideoSender>(
clock_, suspended_ssrcs, states, rtp_config, rtcp_report_interval_ms,
send_transport, is_svc, observers,
send_transport, observers,
// TODO(holmer): Remove this circular dependency by injecting
// the parts of RtpTransportControllerSendInterface that are really used.
this, event_log, &retransmission_rate_limiter_, std::move(fec_controller),

1
call/rtp_transport_controller_send.h
View File

@ -60,7 +60,6 @@ class RtpTransportControllerSend final
const RtpConfig& rtp_config,
int rtcp_report_interval_ms,
Transport* send_transport,
bool is_svc,
const RtpSenderObservers& observers,
RtcEventLog* event_log,
std::unique_ptr<FecController> fec_controller,

1
call/rtp_transport_controller_send_interface.h
View File

@ -104,7 +104,6 @@ class RtpTransportControllerSendInterface {
const RtpConfig& rtp_config,
int rtcp_report_interval_ms,
Transport* send_transport,
bool is_svc,
const RtpSenderObservers& observers,
RtcEventLog* event_log,
std::unique_ptr<FecController> fec_controller,

6
call/rtp_video_sender.cc
View File

@ -212,7 +212,6 @@ RtpVideoSender::RtpVideoSender(
const RtpConfig& rtp_config,
int rtcp_report_interval_ms,
Transport* send_transport,
bool is_svc,
const RtpSenderObservers& observers,
RtpTransportControllerSendInterface* transport,
RtcEventLog* event_log,
@ -254,8 +253,7 @@ RtpVideoSender::RtpVideoSender(
overhead_bytes_per_packet_(0),
encoder_target_rate_bps_(0),
frame_counts_(rtp_config.ssrcs.size()),
frame_count_observer_(observers.frame_count_observer),
is_svc_(is_svc) {
frame_count_observer_(observers.frame_count_observer) {
RTC_DCHECK_EQ(rtp_config.ssrcs.size(), rtp_streams_.size());
module_process_thread_checker_.DetachFromThread();
// SSRCs are assumed to be sorted in the same order as |rtp_modules|.
@ -450,7 +448,7 @@ void RtpVideoSender::OnBitrateAllocationUpdated(
const VideoBitrateAllocation& bitrate) {
rtc::CritScope lock(&crit_);
if (IsActive()) {
if (rtp_streams_.size() == 1 || is_svc_) {
if (rtp_streams_.size() == 1) {
// If spatial scalability is enabled, it is covered by a single stream.
rtp_streams_[0].rtp_rtcp->SetVideoBitrateAllocation(bitrate);
} else {

3
call/rtp_video_sender.h
View File

@ -76,7 +76,6 @@ class RtpVideoSender : public RtpVideoSenderInterface,
const RtpConfig& rtp_config,
int rtcp_report_interval_ms,
Transport* send_transport,
bool is_svc,
const RtpSenderObservers& observers,
RtpTransportControllerSendInterface* transport,
RtcEventLog* event_log,
@ -191,8 +190,6 @@ class RtpVideoSender : public RtpVideoSenderInterface,
std::vector<FrameCounts> frame_counts_ RTC_GUARDED_BY(crit_);
FrameCountObserver* const frame_count_observer_;
const bool is_svc_;
RTC_DISALLOW_COPY_AND_ASSIGN(RtpVideoSender);
};

2
call/rtp_video_sender_unittest.cc
View File

@ -99,7 +99,7 @@ class RtpVideoSenderTestFixture {
std::map<uint32_t, RtpState> suspended_ssrcs;
router_ = absl::make_unique<RtpVideoSender>(
&clock_, suspended_ssrcs, suspended_payload_states, config_.rtp,
config_.rtcp_report_interval_ms, &transport_, config_.is_svc,
config_.rtcp_report_interval_ms, &transport_,
CreateObservers(&call_stats_, &encoder_feedback_, &stats_proxy_,
&stats_proxy_, &stats_proxy_, frame_count_observer,
&stats_proxy_, &stats_proxy_, &send_delay_stats_),

3
call/test/mock_rtp_transport_controller_send.h
View File

@ -32,14 +32,13 @@ namespace webrtc {
class MockRtpTransportControllerSend
: public RtpTransportControllerSendInterface {
public:
MOCK_METHOD10(
MOCK_METHOD9(
CreateRtpVideoSender,
RtpVideoSenderInterface*(std::map<uint32_t, RtpState>,
const std::map<uint32_t, RtpPayloadState>&,
const RtpConfig&,
int rtcp_report_interval_ms,
Transport*,
bool,
const RtpSenderObservers&,
RtcEventLog*,
std::unique_ptr<FecController>,

4
call/video_send_stream.h
View File

@ -151,10 +151,6 @@ class VideoSendStream {
// Per PeerConnection cryptography options.
CryptoOptions crypto_options;
// Forces spatial scalability to be implemented via spatial layers
// instead of simulcast.
bool is_svc;
private:
// Access to the copy constructor is private to force use of the Copy()
// method for those exceptional cases where we do use it.

1
media/engine/webrtc_video_engine.cc
View File

@ -2302,7 +2302,6 @@ void WebRtcVideoChannel::WebRtcVideoSendStream::RecreateWebRtcStream() {
"payload type the set codec. Ignoring RTX.";
config.rtp.rtx.ssrcs.clear();
}
config.is_svc = parameters_.encoder_config.number_of_streams == 1;
stream_ = call_->CreateVideoSendStream(std::move(config),
parameters_.encoder_config.Copy());

3
video/video_quality_test.cc
View File

@ -762,9 +762,6 @@ void VideoQualityTest::SetupVideo(Transport* send_transport,
video_send_configs_[video_idx].suspend_below_min_bitrate =
params_.video[video_idx].suspend_below_min_bitrate;
video_send_configs_[video_idx].is_svc =
params_.ss[video_idx].streams.size() == 1;
video_encoder_configs_[video_idx].number_of_streams =
params_.ss[video_idx].streams.size();
video_encoder_configs_[video_idx].max_bitrate_bps = 0;

1
video/video_send_stream_impl.cc
View File

@ -236,7 +236,6 @@ VideoSendStreamImpl::VideoSendStreamImpl(
config_->rtp,
config_->rtcp_report_interval_ms,
config_->send_transport,
config_->is_svc,
CreateObservers(call_stats,
&encoder_feedback_,
stats_proxy_,

2
video/video_send_stream_impl_unittest.cc
View File

@ -96,7 +96,7 @@ class VideoSendStreamImplTest : public ::testing::Test {
EXPECT_CALL(transport_controller_, packet_router())
.WillRepeatedly(Return(&packet_router_));
EXPECT_CALL(transport_controller_,
CreateRtpVideoSender(_, _, _, _, _, _, _, _, _, _))
CreateRtpVideoSender(_, _, _, _, _, _, _, _, _))
.WillRepeatedly(Return(&rtp_video_sender_));
EXPECT_CALL(rtp_video_sender_, SetActive(_))
.WillRepeatedly(testing::Invoke(