Pass Environment into RtcpSender
To remove usage of RtcpConfiguration fields that are passed through Environment Bug: webrtc:362762208 Change-Id: I1a0f218efe6a893c31ef2272cf2379c66fb7b205 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361746 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Åsa Persson <asapersson@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42984}
This commit is contained in:
Danil Chapovalov
WebRTC LUCI CQ
parent
363dc19f9d
commit
0acbb7745f
@ -22,6 +22,7 @@
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#include "absl/strings/string_view.h"
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#include "api/array_view.h"
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#include "api/environment/environment.h"
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#include "api/rtc_event_log/rtc_event_log.h"
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#include "api/rtp_headers.h"
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#include "api/units/data_rate.h"
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@ -137,10 +138,8 @@ RTCPSender::Configuration RTCPSender::Configuration::FromRtpRtcpConfiguration(
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RTCPSender::Configuration result;
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result.audio = configuration.audio;
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result.local_media_ssrc = configuration.local_media_ssrc;
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result.clock = configuration.clock;
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result.outgoing_transport = configuration.outgoing_transport;
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result.non_sender_rtt_measurement = configuration.non_sender_rtt_measurement;
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result.event_log = configuration.event_log;
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if (configuration.rtcp_report_interval_ms) {
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result.rtcp_report_interval =
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TimeDelta::Millis(configuration.rtcp_report_interval_ms);
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@ -151,13 +150,12 @@ RTCPSender::Configuration RTCPSender::Configuration::FromRtpRtcpConfiguration(
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return result;
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}
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RTCPSender::RTCPSender(Configuration config)
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: audio_(config.audio),
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RTCPSender::RTCPSender(const Environment& env, Configuration config)
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: env_(env),
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audio_(config.audio),
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ssrc_(config.local_media_ssrc),
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clock_(config.clock),
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random_(clock_->TimeInMicroseconds()),
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random_(env_.clock().TimeInMicroseconds()),
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method_(RtcpMode::kOff),
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event_log_(config.event_log),
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transport_(config.outgoing_transport),
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report_interval_(config.rtcp_report_interval.value_or(
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TimeDelta::Millis(config.audio ? kDefaultAudioReportInterval
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@ -243,9 +241,7 @@ int32_t RTCPSender::SendLossNotification(const FeedbackState& feedback_state,
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auto callback = [&](rtc::ArrayView<const uint8_t> packet) {
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transport_->SendRtcp(packet);
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error_code = 0;
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if (event_log_) {
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event_log_->Log(std::make_unique<RtcEventRtcpPacketOutgoing>(packet));
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}
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env_.event_log().Log(std::make_unique<RtcEventRtcpPacketOutgoing>(packet));
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};
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std::optional<PacketSender> sender;
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{
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@ -326,7 +322,7 @@ void RTCPSender::SetLastRtpTime(uint32_t rtp_timestamp,
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last_rtp_timestamp_ = rtp_timestamp;
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if (!capture_time.has_value()) {
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// We don't currently get a capture time from VoiceEngine.
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last_frame_capture_time_ = clock_->CurrentTime();
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last_frame_capture_time_ = env_.clock().CurrentTime();
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} else {
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last_frame_capture_time_ = *capture_time;
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}
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@ -360,7 +356,7 @@ int32_t RTCPSender::SetCNAME(absl::string_view c_name) {
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}
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bool RTCPSender::TimeToSendRTCPReport(bool send_keyframe_before_rtp) const {
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Timestamp now = clock_->CurrentTime();
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Timestamp now = env_.clock().CurrentTime();
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MutexLock lock(&mutex_rtcp_sender_);
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RTC_DCHECK(
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@ -400,7 +396,7 @@ void RTCPSender::BuildSR(const RtcpContext& ctx, PacketSender& sender) {
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rtcp::SenderReport report;
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report.SetSenderSsrc(ssrc_);
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report.SetNtp(clock_->ConvertTimestampToNtpTime(ctx.now_));
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report.SetNtp(env_.clock().ConvertTimestampToNtpTime(ctx.now_));
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report.SetRtpTimestamp(rtp_timestamp);
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report.SetPacketCount(ctx.feedback_state_.packets_sent);
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report.SetOctetCount(ctx.feedback_state_.media_bytes_sent);
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@ -568,7 +564,7 @@ void RTCPSender::BuildExtendedReports(const RtcpContext& ctx,
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if (!sending_ && xr_send_receiver_reference_time_enabled_) {
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rtcp::Rrtr rrtr;
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rrtr.SetNtp(clock_->ConvertTimestampToNtpTime(ctx.now_));
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rrtr.SetNtp(env_.clock().ConvertTimestampToNtpTime(ctx.now_));
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xr.SetRrtr(rrtr);
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}
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@ -602,9 +598,8 @@ int32_t RTCPSender::SendRTCP(const FeedbackState& feedback_state,
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auto callback = [&](rtc::ArrayView<const uint8_t> packet) {
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if (transport_->SendRtcp(packet)) {
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error_code = 0;
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if (event_log_) {
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event_log_->Log(std::make_unique<RtcEventRtcpPacketOutgoing>(packet));
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}
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env_.event_log().Log(
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std::make_unique<RtcEventRtcpPacketOutgoing>(packet));
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}
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};
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std::optional<PacketSender> sender;
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@ -654,7 +649,7 @@ std::optional<int32_t> RTCPSender::ComputeCompoundRTCPPacket(
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// We need to send our NTP even if we haven't received any reports.
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RtcpContext context(feedback_state, nack_size, nack_list,
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clock_->CurrentTime());
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env_.clock().CurrentTime());
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PrepareReport(feedback_state);
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@ -781,7 +776,7 @@ std::vector<rtcp::ReportBlock> RTCPSender::CreateReportBlocks(
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if (!result.empty() && feedback_state.last_rr.Valid()) {
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// Get our NTP as late as possible to avoid a race.
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uint32_t now = CompactNtp(clock_->CurrentNtpTime());
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uint32_t now = CompactNtp(env_.clock().CurrentNtpTime());
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uint32_t receive_time = CompactNtp(feedback_state.last_rr);
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uint32_t delay_since_last_sr = now - receive_time;
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@ -900,8 +895,8 @@ void RTCPSender::SendCombinedRtcpPacket(
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RTC_DCHECK_LE(max_packet_size, IP_PACKET_SIZE);
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auto callback = [&](rtc::ArrayView<const uint8_t> packet) {
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if (transport_->SendRtcp(packet)) {
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if (event_log_)
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event_log_->Log(std::make_unique<RtcEventRtcpPacketOutgoing>(packet));
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env_.event_log().Log(
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std::make_unique<RtcEventRtcpPacketOutgoing>(packet));
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}
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};
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PacketSender sender(callback, max_packet_size);
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@ -913,7 +908,7 @@ void RTCPSender::SendCombinedRtcpPacket(
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}
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void RTCPSender::SetNextRtcpSendEvaluationDuration(TimeDelta duration) {
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next_time_to_send_rtcp_ = clock_->CurrentTime() + duration;
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next_time_to_send_rtcp_ = env_.clock().CurrentTime() + duration;
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// TODO(bugs.webrtc.org/11581): make unconditional once downstream consumers
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// are using the callback method.
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if (schedule_next_rtcp_send_evaluation_function_)
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@ -23,6 +23,7 @@
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#include "absl/strings/string_view.h"
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#include "api/call/transport.h"
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#include "api/environment/environment.h"
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#include "api/rtp_headers.h"
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#include "api/units/data_rate.h"
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#include "api/units/time_delta.h"
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@ -46,7 +47,6 @@
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namespace webrtc {
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class RTCPReceiver;
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class RtcEventLog;
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class RTCPSender final {
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public:
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@ -62,8 +62,7 @@ class RTCPSender final {
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// SSRCs for media and retransmission, respectively.
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// FlexFec SSRC is fetched from `flexfec_sender`.
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uint32_t local_media_ssrc = 0;
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// The clock to use to read time. If nullptr then system clock will be used.
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Clock* clock = nullptr;
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// Transport object that will be called when packets are ready to be sent
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// out on the network.
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Transport* outgoing_transport = nullptr;
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@ -82,7 +81,6 @@ class RTCPSender final {
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// have migrated to the callback solution.
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std::function<void(TimeDelta)> schedule_next_rtcp_send_evaluation_function;
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RtcEventLog* event_log = nullptr;
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std::optional<TimeDelta> rtcp_report_interval;
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ReceiveStatisticsProvider* receive_statistics = nullptr;
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RtcpPacketTypeCounterObserver* rtcp_packet_type_counter_observer = nullptr;
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@ -107,13 +105,13 @@ class RTCPSender final {
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RTCPReceiver* receiver;
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};
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explicit RTCPSender(Configuration config);
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RTCPSender(const Environment& env, Configuration config);
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RTCPSender() = delete;
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RTCPSender(const RTCPSender&) = delete;
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RTCPSender& operator=(const RTCPSender&) = delete;
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virtual ~RTCPSender();
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~RTCPSender();
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RtcpMode Status() const RTC_LOCKS_EXCLUDED(mutex_rtcp_sender_);
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void SetRTCPStatus(RtcpMode method) RTC_LOCKS_EXCLUDED(mutex_rtcp_sender_);
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@ -238,17 +236,16 @@ class RTCPSender final {
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void SetNextRtcpSendEvaluationDuration(TimeDelta duration)
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RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_rtcp_sender_);
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const Environment env_;
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const bool audio_;
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// TODO(bugs.webrtc.org/11581): `mutex_rtcp_sender_` shouldn't be required if
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// we consistently run network related operations on the network thread.
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// This is currently not possible due to callbacks from the process thread in
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// ModuleRtpRtcpImpl2.
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uint32_t ssrc_ RTC_GUARDED_BY(mutex_rtcp_sender_);
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Clock* const clock_;
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Random random_ RTC_GUARDED_BY(mutex_rtcp_sender_);
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RtcpMode method_ RTC_GUARDED_BY(mutex_rtcp_sender_);
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RtcEventLog* const event_log_;
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Transport* const transport_;
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const TimeDelta report_interval_;
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@ -10,20 +10,40 @@
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#include "modules/rtp_rtcp/source/rtcp_sender.h"
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#include <cstddef>
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#include <cstdint>
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#include <memory>
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#include <optional>
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#include <utility>
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#include <vector>
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#include "absl/base/macros.h"
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#include "api/array_view.h"
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#include "api/call/transport.h"
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#include "api/environment/environment.h"
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#include "api/environment/environment_factory.h"
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#include "api/rtp_headers.h"
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#include "api/units/time_delta.h"
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#include "api/units/timestamp.h"
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#include "api/video/video_bitrate_allocation.h"
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#include "modules/rtp_rtcp/include/receive_statistics.h"
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#include "modules/rtp_rtcp/include/rtcp_statistics.h"
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#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
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#include "modules/rtp_rtcp/source/rtcp_packet.h"
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#include "modules/rtp_rtcp/source/rtcp_packet/bye.h"
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#include "modules/rtp_rtcp/source/rtcp_packet/common_header.h"
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#include "modules/rtp_rtcp/source/rtcp_packet/dlrr.h"
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#include "modules/rtp_rtcp/source/rtcp_packet/remote_estimate.h"
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#include "modules/rtp_rtcp/source/rtcp_packet/report_block.h"
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#include "modules/rtp_rtcp/source/rtcp_packet/target_bitrate.h"
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#include "modules/rtp_rtcp/source/rtcp_packet/tmmb_item.h"
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#include "modules/rtp_rtcp/source/rtp_packet_received.h"
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#include "modules/rtp_rtcp/source/rtp_rtcp_impl2.h"
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#include "modules/rtp_rtcp/source/rtp_rtcp_interface.h"
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#include "rtc_base/rate_limiter.h"
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#include "rtc_base/thread.h"
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#include "system_wrappers/include/clock.h"
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#include "system_wrappers/include/ntp_time.h"
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#include "test/gmock.h"
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#include "test/gtest.h"
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#include "test/mock_transport.h"
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@ -72,19 +92,6 @@ static const uint32_t kSenderSsrc = 0x11111111;
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static const uint32_t kRemoteSsrc = 0x22222222;
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static const uint32_t kStartRtpTimestamp = 0x34567;
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static const uint32_t kRtpTimestamp = 0x45678;
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std::unique_ptr<RTCPSender> CreateRtcpSender(
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const RTCPSender::Configuration& config,
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bool init_timestamps = true) {
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auto rtcp_sender = std::make_unique<RTCPSender>(config);
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rtcp_sender->SetRemoteSSRC(kRemoteSsrc);
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if (init_timestamps) {
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rtcp_sender->SetTimestampOffset(kStartRtpTimestamp);
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rtcp_sender->SetLastRtpTime(kRtpTimestamp, config.clock->CurrentTime(),
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/*payload_type=*/0);
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}
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return rtcp_sender;
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}
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} // namespace
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class RtcpSenderTest : public ::testing::Test {
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@ -98,7 +105,6 @@ class RtcpSenderTest : public ::testing::Test {
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RTCPSender::Configuration GetDefaultConfig() {
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RTCPSender::Configuration configuration;
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configuration.audio = false;
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configuration.clock = &clock_;
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configuration.outgoing_transport = &test_transport_;
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configuration.rtcp_report_interval = TimeDelta::Millis(1000);
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configuration.receive_statistics = receive_statistics_.get();
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@ -117,6 +123,19 @@ class RtcpSenderTest : public ::testing::Test {
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return result;
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}
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std::unique_ptr<RTCPSender> CreateRtcpSender(
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const RTCPSender::Configuration& config,
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bool init_timestamps = true) {
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auto rtcp_sender = std::make_unique<RTCPSender>(env_, config);
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rtcp_sender->SetRemoteSSRC(kRemoteSsrc);
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if (init_timestamps) {
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rtcp_sender->SetTimestampOffset(kStartRtpTimestamp);
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rtcp_sender->SetLastRtpTime(kRtpTimestamp, env_.clock().CurrentTime(),
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/*payload_type=*/0);
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}
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return rtcp_sender;
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}
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void InsertIncomingPacket(uint32_t remote_ssrc, uint16_t seq_num) {
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RtpPacketReceived packet;
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packet.SetSsrc(remote_ssrc);
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@ -217,7 +236,6 @@ TEST_F(RtcpSenderTest, SendConsecutiveSrWithExactSlope) {
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TEST_F(RtcpSenderTest, DoNotSendSrBeforeRtp) {
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RTCPSender::Configuration config;
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config.clock = &clock_;
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config.receive_statistics = receive_statistics_.get();
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config.outgoing_transport = &test_transport_;
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config.rtcp_report_interval = TimeDelta::Millis(1000);
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@ -238,7 +256,6 @@ TEST_F(RtcpSenderTest, DoNotSendSrBeforeRtp) {
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TEST_F(RtcpSenderTest, DoNotSendCompundBeforeRtp) {
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RTCPSender::Configuration config;
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config.clock = &clock_;
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config.receive_statistics = receive_statistics_.get();
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config.outgoing_transport = &test_transport_;
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config.rtcp_report_interval = TimeDelta::Millis(1000);
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@ -592,7 +609,6 @@ TEST_F(RtcpSenderTest, TestNoXrRrtrSentIfNotEnabled) {
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TEST_F(RtcpSenderTest, TestRegisterRtcpPacketTypeObserver) {
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RtcpPacketTypeCounterObserverImpl observer;
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RTCPSender::Configuration config;
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config.clock = &clock_;
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config.receive_statistics = receive_statistics_.get();
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config.outgoing_transport = &test_transport_;
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config.rtcp_packet_type_counter_observer = &observer;
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@ -687,7 +703,6 @@ TEST_F(RtcpSenderTest, ByeMustBeLast) {
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// Re-configure rtcp_sender with mock_transport_
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RTCPSender::Configuration config;
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config.clock = &clock_;
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config.receive_statistics = receive_statistics_.get();
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config.outgoing_transport = &mock_transport;
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config.rtcp_report_interval = TimeDelta::Millis(1000);
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@ -89,6 +89,7 @@ ModuleRtpRtcpImpl::ModuleRtpRtcpImpl(const Environment& env,
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const Configuration& configuration)
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: env_(env),
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rtcp_sender_(
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env_,
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RTCPSender::Configuration::FromRtpRtcpConfiguration(configuration)),
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rtcp_receiver_(env_, configuration, this),
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last_bitrate_process_time_(env_.clock().TimeInMilliseconds()),
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@ -111,11 +111,13 @@ ModuleRtpRtcpImpl2::ModuleRtpRtcpImpl2(TagConfigurationIncludesEnvironment,
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const Configuration& configuration)
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: env_(env),
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worker_queue_(TaskQueueBase::Current()),
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rtcp_sender_(AddRtcpSendEvaluationCallback(
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RTCPSender::Configuration::FromRtpRtcpConfiguration(configuration),
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[this](TimeDelta duration) {
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ScheduleRtcpSendEvaluation(duration);
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})),
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rtcp_sender_(env_,
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AddRtcpSendEvaluationCallback(
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RTCPSender::Configuration::FromRtpRtcpConfiguration(
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configuration),
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[this](TimeDelta duration) {
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ScheduleRtcpSendEvaluation(duration);
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})),
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rtcp_receiver_(env_, configuration, this),
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packet_overhead_(28), // IPV4 UDP.
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nack_last_time_sent_full_ms_(0),
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@ -806,15 +806,15 @@ if (rtc_include_tests) {
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":video_stream_buffer_controller",
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":video_stream_encoder_impl",
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":video_stream_encoder_interface",
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"..//test/network:simulated_network",
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"../api:array_view",
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"../api:bitrate_allocation",
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"../api:create_frame_generator",
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"../api:fake_frame_decryptor",
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"../api:fake_frame_encryptor",
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"../api:fec_controller_api",
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"../api:field_trials_view",
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"../api:frame_generator_api",
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"../api:libjingle_peerconnection_api",
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"../api:make_ref_counted",
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"../api:mock_fec_controller_override",
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"../api:mock_frame_decryptor",
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"../api:mock_frame_transformer",
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@ -827,9 +827,6 @@ if (rtc_include_tests) {
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"../api:sequence_checker",
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"../api:simulated_network_api",
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"../api:time_controller",
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"../api:transport_api",
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"../api/adaptation:resource_adaptation_api",
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"../api/adaptation:resource_adaptation_api",
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"../api/crypto:options",
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"../api/environment",
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"../api/environment:environment_factory",
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@ -837,9 +834,11 @@ if (rtc_include_tests) {
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"../api/rtc_event_log",
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"../api/task_queue",
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"../api/task_queue:default_task_queue_factory",
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"../api/task_queue:pending_task_safety_flag",
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"../api/test/metrics:global_metrics_logger_and_exporter",
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"../api/test/metrics:metric",
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"../api/test/video:function_video_factory",
|
||||
"../api/transport:bitrate_settings",
|
||||
"../api/units:data_rate",
|
||||
"../api/units:data_size",
|
||||
"../api/units:frequency",
|
||||
@ -851,6 +850,8 @@ if (rtc_include_tests) {
|
||||
"../api/video:resolution",
|
||||
"../api/video:video_adaptation",
|
||||
"../api/video:video_bitrate_allocation",
|
||||
"../api/video:video_bitrate_allocator",
|
||||
"../api/video:video_bitrate_allocator_factory",
|
||||
"../api/video:video_frame",
|
||||
"../api/video:video_frame_type",
|
||||
"../api/video:video_layers_allocation",
|
||||
@ -877,20 +878,15 @@ if (rtc_include_tests) {
|
||||
"../media:media_constants",
|
||||
"../media:rtc_audio_video",
|
||||
"../media:rtc_internal_video_codecs",
|
||||
"../media:rtc_media",
|
||||
"../media:rtc_media_tests_utils",
|
||||
"../media:rtc_simulcast_encoder_adapter",
|
||||
"../media:video_adapter",
|
||||
"../modules:module_api_public",
|
||||
"../modules/pacing",
|
||||
"../modules/rtp_rtcp",
|
||||
"../modules/rtp_rtcp:mock_rtp_rtcp",
|
||||
"../modules/rtp_rtcp:rtp_rtcp_format",
|
||||
"../modules/utility:utility",
|
||||
"../modules/video_coding",
|
||||
"../modules/video_coding:codec_globals_headers",
|
||||
"../modules/video_coding:encoded_frame",
|
||||
"../modules/video_coding:h26x_packet_buffer",
|
||||
"../modules/video_coding:packet_buffer",
|
||||
"../modules/video_coding:video_codec_interface",
|
||||
"../modules/video_coding:video_coding_utility",
|
||||
@ -908,7 +904,7 @@ if (rtc_include_tests) {
|
||||
"../rtc_base:gunit_helpers",
|
||||
"../rtc_base:logging",
|
||||
"../rtc_base:macromagic",
|
||||
"../rtc_base:platform_thread",
|
||||
"../rtc_base:network_route",
|
||||
"../rtc_base:rate_limiter",
|
||||
"../rtc_base:rate_statistics",
|
||||
"../rtc_base:refcount",
|
||||
@ -933,7 +929,6 @@ if (rtc_include_tests) {
|
||||
"../test:fake_encoded_frame",
|
||||
"../test:fake_video_codecs",
|
||||
"../test:field_trial",
|
||||
"../test:fileutils",
|
||||
"../test:frame_generator_capturer",
|
||||
"../test:frame_utils",
|
||||
"../test:mock_transport",
|
||||
@ -945,8 +940,8 @@ if (rtc_include_tests) {
|
||||
"../test:test_support",
|
||||
"../test:video_test_common",
|
||||
"../test:video_test_constants",
|
||||
"../test/network:simulated_network",
|
||||
"../test/time_controller",
|
||||
"adaptation:video_adaptation",
|
||||
"config:encoder_config",
|
||||
"config:streams_config",
|
||||
"config:video_config_tests",
|
||||
|
||||
@ -8,42 +8,82 @@
|
||||
* be found in the AUTHORS file in the root of the source tree.
|
||||
*/
|
||||
#include <algorithm> // max
|
||||
#include <cstddef>
|
||||
#include <cstdint>
|
||||
#include <cstring>
|
||||
#include <map>
|
||||
#include <memory>
|
||||
#include <optional>
|
||||
#include <set>
|
||||
#include <string>
|
||||
#include <tuple>
|
||||
#include <utility>
|
||||
#include <vector>
|
||||
|
||||
#include "absl/algorithm/container.h"
|
||||
#include "absl/strings/match.h"
|
||||
#include "absl/types/variant.h"
|
||||
#include "api/array_view.h"
|
||||
#include "api/environment/environment.h"
|
||||
#include "api/environment/environment_factory.h"
|
||||
#include "api/fec_controller_override.h"
|
||||
#include "api/make_ref_counted.h"
|
||||
#include "api/rtp_headers.h"
|
||||
#include "api/rtp_parameters.h"
|
||||
#include "api/scoped_refptr.h"
|
||||
#include "api/sequence_checker.h"
|
||||
#include "api/task_queue/pending_task_safety_flag.h"
|
||||
#include "api/task_queue/task_queue_base.h"
|
||||
#include "api/test/metrics/global_metrics_logger_and_exporter.h"
|
||||
#include "api/test/metrics/metric.h"
|
||||
#include "api/test/simulated_network.h"
|
||||
#include "api/test/video/function_video_encoder_factory.h"
|
||||
#include "api/transport/bitrate_settings.h"
|
||||
#include "api/units/data_rate.h"
|
||||
#include "api/units/time_delta.h"
|
||||
#include "api/video/builtin_video_bitrate_allocator_factory.h"
|
||||
#include "api/video/encoded_image.h"
|
||||
#include "api/video/video_bitrate_allocation.h"
|
||||
#include "api/video/video_bitrate_allocator.h"
|
||||
#include "api/video/video_bitrate_allocator_factory.h"
|
||||
#include "api/video/video_codec_type.h"
|
||||
#include "api/video/video_content_type.h"
|
||||
#include "api/video/video_frame_type.h"
|
||||
#include "api/video/video_rotation.h"
|
||||
#include "api/video/video_sink_interface.h"
|
||||
#include "api/video/video_source_interface.h"
|
||||
#include "api/video_codecs/scalability_mode.h"
|
||||
#include "api/video_codecs/sdp_video_format.h"
|
||||
#include "api/video_codecs/video_codec.h"
|
||||
#include "api/video_codecs/video_encoder.h"
|
||||
#include "call/audio_receive_stream.h"
|
||||
#include "call/audio_send_stream.h"
|
||||
#include "call/call.h"
|
||||
#include "call/fake_network_pipe.h"
|
||||
#include "call/rtp_transport_controller_send.h"
|
||||
#include "call/video_receive_stream.h"
|
||||
#include "call/video_send_stream.h"
|
||||
#include "media/base/video_common.h"
|
||||
#include "media/engine/internal_encoder_factory.h"
|
||||
#include "media/engine/simulcast_encoder_adapter.h"
|
||||
#include "media/engine/webrtc_video_engine.h"
|
||||
#include "modules/include/module_common_types_public.h"
|
||||
#include "modules/rtp_rtcp/include/receive_statistics.h"
|
||||
#include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
|
||||
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
|
||||
#include "modules/rtp_rtcp/source/create_video_rtp_depacketizer.h"
|
||||
#include "modules/rtp_rtcp/source/rtcp_packet/report_block.h"
|
||||
#include "modules/rtp_rtcp/source/rtcp_sender.h"
|
||||
#include "modules/rtp_rtcp/source/rtp_header_extensions.h"
|
||||
#include "modules/rtp_rtcp/source/rtp_packet.h"
|
||||
#include "modules/rtp_rtcp/source/rtp_rtcp_impl2.h"
|
||||
#include "modules/rtp_rtcp/source/rtp_rtcp_interface.h"
|
||||
#include "modules/rtp_rtcp/source/rtp_util.h"
|
||||
#include "modules/rtp_rtcp/source/video_rtp_depacketizer.h"
|
||||
#include "modules/rtp_rtcp/source/video_rtp_depacketizer_vp9.h"
|
||||
#include "modules/video_coding/codecs/interface/common_constants.h"
|
||||
#include "modules/video_coding/codecs/vp8/include/vp8.h"
|
||||
#include "modules/video_coding/codecs/vp8/include/vp8_globals.h"
|
||||
#include "modules/video_coding/codecs/vp9/include/vp9.h"
|
||||
#include "modules/video_coding/codecs/vp9/include/vp9_globals.h"
|
||||
#include "modules/video_coding/svc/create_scalability_structure.h"
|
||||
#include "modules/video_coding/svc/scalability_mode_util.h"
|
||||
#include "modules/video_coding/svc/scalable_video_controller.h"
|
||||
@ -51,31 +91,33 @@
|
||||
#include "rtc_base/event.h"
|
||||
#include "rtc_base/experiments/alr_experiment.h"
|
||||
#include "rtc_base/logging.h"
|
||||
#include "rtc_base/platform_thread.h"
|
||||
#include "rtc_base/network_route.h"
|
||||
#include "rtc_base/rate_limiter.h"
|
||||
#include "rtc_base/strings/string_builder.h"
|
||||
#include "rtc_base/synchronization/mutex.h"
|
||||
#include "rtc_base/task_queue_for_test.h"
|
||||
#include "rtc_base/thread.h"
|
||||
#include "rtc_base/thread_annotations.h"
|
||||
#include "rtc_base/time_utils.h"
|
||||
#include "rtc_base/unique_id_generator.h"
|
||||
#include "system_wrappers/include/sleep.h"
|
||||
#include "test/call_test.h"
|
||||
#include "test/configurable_frame_size_encoder.h"
|
||||
#include "test/encoder_settings.h"
|
||||
#include "test/fake_encoder.h"
|
||||
#include "test/fake_texture_frame.h"
|
||||
#include "test/field_trial.h"
|
||||
#include "test/frame_forwarder.h"
|
||||
#include "test/frame_generator_capturer.h"
|
||||
#include "test/frame_utils.h"
|
||||
#include "test/gmock.h"
|
||||
#include "test/gtest.h"
|
||||
#include "test/network/simulated_network.h"
|
||||
#include "test/null_transport.h"
|
||||
#include "test/rtcp_packet_parser.h"
|
||||
#include "test/rtp_rtcp_observer.h"
|
||||
#include "test/scoped_key_value_config.h"
|
||||
#include "test/video_encoder_proxy_factory.h"
|
||||
#include "test/video_test_constants.h"
|
||||
#include "video/config/encoder_stream_factory.h"
|
||||
#include "video/send_statistics_proxy.h"
|
||||
#include "video/config/video_encoder_config.h"
|
||||
#include "video/transport_adapter.h"
|
||||
#include "video/video_send_stream_impl.h"
|
||||
|
||||
@ -991,12 +1033,11 @@ void VideoSendStreamTest::TestNackRetransmission(
|
||||
// NACK packets at arbitrary points.
|
||||
if (send_count_ % 25 == 0) {
|
||||
RTCPSender::Configuration config;
|
||||
config.clock = Clock::GetRealTimeClock();
|
||||
config.outgoing_transport = transport_adapter_.get();
|
||||
config.rtcp_report_interval = TimeDelta::Millis(kRtcpIntervalMs);
|
||||
config.local_media_ssrc =
|
||||
test::VideoTestConstants::kReceiverLocalVideoSsrc;
|
||||
RTCPSender rtcp_sender(config);
|
||||
RTCPSender rtcp_sender(env_, config);
|
||||
|
||||
rtcp_sender.SetRTCPStatus(RtcpMode::kReducedSize);
|
||||
rtcp_sender.SetRemoteSSRC(test::VideoTestConstants::kVideoSendSsrcs[0]);
|
||||
@ -1069,6 +1110,7 @@ void VideoSendStreamTest::TestNackRetransmission(
|
||||
EXPECT_TRUE(Wait()) << "Timed out while waiting for NACK retransmission.";
|
||||
}
|
||||
|
||||
const Environment env_ = CreateEnvironment();
|
||||
std::unique_ptr<internal::TransportAdapter> transport_adapter_;
|
||||
int send_count_;
|
||||
int retransmit_count_;
|
||||
@ -1230,12 +1272,11 @@ void VideoSendStreamTest::TestPacketFragmentationSize(VideoFormat format,
|
||||
packets_lost_, // Cumulative lost.
|
||||
loss_ratio); // Loss percent.
|
||||
RTCPSender::Configuration config;
|
||||
config.clock = Clock::GetRealTimeClock();
|
||||
config.receive_statistics = &lossy_receive_stats;
|
||||
config.outgoing_transport = transport_adapter_.get();
|
||||
config.rtcp_report_interval = TimeDelta::Millis(kRtcpIntervalMs);
|
||||
config.local_media_ssrc = test::VideoTestConstants::kVideoSendSsrcs[0];
|
||||
RTCPSender rtcp_sender(config);
|
||||
RTCPSender rtcp_sender(env_, config);
|
||||
|
||||
rtcp_sender.SetRTCPStatus(RtcpMode::kReducedSize);
|
||||
rtcp_sender.SetRemoteSSRC(test::VideoTestConstants::kVideoSendSsrcs[0]);
|
||||
@ -1296,6 +1337,7 @@ void VideoSendStreamTest::TestPacketFragmentationSize(VideoFormat format,
|
||||
EXPECT_TRUE(Wait()) << "Timed out while observing incoming RTP packets.";
|
||||
}
|
||||
|
||||
const Environment env_ = CreateEnvironment();
|
||||
std::unique_ptr<internal::TransportAdapter> transport_adapter_;
|
||||
test::ConfigurableFrameSizeEncoder encoder_;
|
||||
test::VideoEncoderProxyFactory encoder_factory_;
|
||||
|
||||
Loading…
x
Reference in New Issue
Block a user