Index: webrtc/video_engine/vie_channel.cc |
diff --git a/webrtc/video_engine/vie_channel.cc b/webrtc/video_engine/vie_channel.cc |
deleted file mode 100644 |
index 681f72cc06bde2f4d5d2f487f76d2d96a0a30d4f..0000000000000000000000000000000000000000 |
--- a/webrtc/video_engine/vie_channel.cc |
+++ /dev/null |
@@ -1,1203 +0,0 @@ |
-/* |
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. |
- * |
- * Use of this source code is governed by a BSD-style license |
- * that can be found in the LICENSE file in the root of the source |
- * tree. An additional intellectual property rights grant can be found |
- * in the file PATENTS. All contributing project authors may |
- * be found in the AUTHORS file in the root of the source tree. |
- */ |
- |
-#include "webrtc/video_engine/vie_channel.h" |
- |
-#include <algorithm> |
-#include <map> |
-#include <vector> |
- |
-#include "webrtc/base/checks.h" |
-#include "webrtc/base/logging.h" |
-#include "webrtc/base/platform_thread.h" |
-#include "webrtc/common.h" |
-#include "webrtc/common_video/include/incoming_video_stream.h" |
-#include "webrtc/common_video/libyuv/include/webrtc_libyuv.h" |
-#include "webrtc/frame_callback.h" |
-#include "webrtc/modules/pacing/paced_sender.h" |
-#include "webrtc/modules/pacing/packet_router.h" |
-#include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h" |
-#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h" |
-#include "webrtc/modules/utility/include/process_thread.h" |
-#include "webrtc/modules/video_coding/include/video_coding.h" |
-#include "webrtc/modules/video_processing/include/video_processing.h" |
-#include "webrtc/modules/video_render/video_render_defines.h" |
-#include "webrtc/system_wrappers/include/critical_section_wrapper.h" |
-#include "webrtc/system_wrappers/include/metrics.h" |
-#include "webrtc/video/receive_statistics_proxy.h" |
-#include "webrtc/video_engine/call_stats.h" |
-#include "webrtc/video_engine/payload_router.h" |
-#include "webrtc/video_engine/report_block_stats.h" |
- |
-namespace webrtc { |
- |
-const int kMaxDecodeWaitTimeMs = 50; |
-static const int kMaxTargetDelayMs = 10000; |
-const int kMinSendSidePacketHistorySize = 600; |
-const int kMaxPacketAgeToNack = 450; |
-const int kMaxNackListSize = 250; |
- |
-// Helper class receiving statistics callbacks. |
-class ChannelStatsObserver : public CallStatsObserver { |
- public: |
- explicit ChannelStatsObserver(ViEChannel* owner) : owner_(owner) {} |
- virtual ~ChannelStatsObserver() {} |
- |
- // Implements StatsObserver. |
- virtual void OnRttUpdate(int64_t avg_rtt_ms, int64_t max_rtt_ms) { |
- owner_->OnRttUpdate(avg_rtt_ms, max_rtt_ms); |
- } |
- |
- private: |
- ViEChannel* const owner_; |
-}; |
- |
-class ViEChannelProtectionCallback : public VCMProtectionCallback { |
- public: |
- explicit ViEChannelProtectionCallback(ViEChannel* owner) : owner_(owner) {} |
- ~ViEChannelProtectionCallback() {} |
- |
- |
- int ProtectionRequest( |
- const FecProtectionParams* delta_fec_params, |
- const FecProtectionParams* key_fec_params, |
- uint32_t* sent_video_rate_bps, |
- uint32_t* sent_nack_rate_bps, |
- uint32_t* sent_fec_rate_bps) override { |
- return owner_->ProtectionRequest(delta_fec_params, key_fec_params, |
- sent_video_rate_bps, sent_nack_rate_bps, |
- sent_fec_rate_bps); |
- } |
- private: |
- ViEChannel* owner_; |
-}; |
- |
-ViEChannel::ViEChannel(uint32_t number_of_cores, |
- Transport* transport, |
- ProcessThread* module_process_thread, |
- RtcpIntraFrameObserver* intra_frame_observer, |
- RtcpBandwidthObserver* bandwidth_observer, |
- TransportFeedbackObserver* transport_feedback_observer, |
- RemoteBitrateEstimator* remote_bitrate_estimator, |
- RtcpRttStats* rtt_stats, |
- PacedSender* paced_sender, |
- PacketRouter* packet_router, |
- size_t max_rtp_streams, |
- bool sender) |
- : number_of_cores_(number_of_cores), |
- sender_(sender), |
- module_process_thread_(module_process_thread), |
- crit_(CriticalSectionWrapper::CreateCriticalSection()), |
- send_payload_router_(new PayloadRouter()), |
- vcm_protection_callback_(new ViEChannelProtectionCallback(this)), |
- vcm_(VideoCodingModule::Create(Clock::GetRealTimeClock(), |
- nullptr, |
- nullptr)), |
- vie_receiver_(vcm_, remote_bitrate_estimator, this), |
- vie_sync_(vcm_), |
- stats_observer_(new ChannelStatsObserver(this)), |
- receive_stats_callback_(nullptr), |
- incoming_video_stream_(nullptr), |
- intra_frame_observer_(intra_frame_observer), |
- rtt_stats_(rtt_stats), |
- paced_sender_(paced_sender), |
- packet_router_(packet_router), |
- bandwidth_observer_(bandwidth_observer), |
- transport_feedback_observer_(transport_feedback_observer), |
- decode_thread_(ChannelDecodeThreadFunction, this, "DecodingThread"), |
- nack_history_size_sender_(kMinSendSidePacketHistorySize), |
- max_nack_reordering_threshold_(kMaxPacketAgeToNack), |
- pre_render_callback_(NULL), |
- report_block_stats_sender_(new ReportBlockStats()), |
- time_of_first_rtt_ms_(-1), |
- rtt_sum_ms_(0), |
- last_rtt_ms_(0), |
- num_rtts_(0), |
- rtp_rtcp_modules_( |
- CreateRtpRtcpModules(!sender, |
- vie_receiver_.GetReceiveStatistics(), |
- transport, |
- intra_frame_observer_, |
- bandwidth_observer_.get(), |
- transport_feedback_observer_, |
- rtt_stats_, |
- &rtcp_packet_type_counter_observer_, |
- remote_bitrate_estimator, |
- paced_sender_, |
- packet_router_, |
- &send_bitrate_observer_, |
- &send_frame_count_observer_, |
- &send_side_delay_observer_, |
- max_rtp_streams)), |
- num_active_rtp_rtcp_modules_(1) { |
- vie_receiver_.SetRtpRtcpModule(rtp_rtcp_modules_[0]); |
- vcm_->SetNackSettings(kMaxNackListSize, max_nack_reordering_threshold_, 0); |
-} |
- |
-int32_t ViEChannel::Init() { |
- static const int kDefaultRenderDelayMs = 10; |
- module_process_thread_->RegisterModule(vie_receiver_.GetReceiveStatistics()); |
- |
- // RTP/RTCP initialization. |
- module_process_thread_->RegisterModule(rtp_rtcp_modules_[0]); |
- |
- rtp_rtcp_modules_[0]->SetKeyFrameRequestMethod(kKeyFrameReqPliRtcp); |
- if (paced_sender_) { |
- for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) |
- rtp_rtcp->SetStorePacketsStatus(true, nack_history_size_sender_); |
- } |
- packet_router_->AddRtpModule(rtp_rtcp_modules_[0]); |
- if (sender_) { |
- std::list<RtpRtcp*> send_rtp_modules(1, rtp_rtcp_modules_[0]); |
- send_payload_router_->SetSendingRtpModules(send_rtp_modules); |
- RTC_DCHECK(!send_payload_router_->active()); |
- } |
- if (vcm_->RegisterReceiveCallback(this) != 0) { |
- return -1; |
- } |
- vcm_->RegisterFrameTypeCallback(this); |
- vcm_->RegisterReceiveStatisticsCallback(this); |
- vcm_->RegisterDecoderTimingCallback(this); |
- vcm_->SetRenderDelay(kDefaultRenderDelayMs); |
- |
- module_process_thread_->RegisterModule(vcm_); |
- module_process_thread_->RegisterModule(&vie_sync_); |
- |
- return 0; |
-} |
- |
-ViEChannel::~ViEChannel() { |
- UpdateHistograms(); |
- // Make sure we don't get more callbacks from the RTP module. |
- module_process_thread_->DeRegisterModule( |
- vie_receiver_.GetReceiveStatistics()); |
- module_process_thread_->DeRegisterModule(vcm_); |
- module_process_thread_->DeRegisterModule(&vie_sync_); |
- send_payload_router_->SetSendingRtpModules(std::list<RtpRtcp*>()); |
- for (size_t i = 0; i < num_active_rtp_rtcp_modules_; ++i) |
- packet_router_->RemoveRtpModule(rtp_rtcp_modules_[i]); |
- for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) { |
- module_process_thread_->DeRegisterModule(rtp_rtcp); |
- delete rtp_rtcp; |
- } |
- if (!sender_) |
- StopDecodeThread(); |
- // Release modules. |
- VideoCodingModule::Destroy(vcm_); |
-} |
- |
-void ViEChannel::UpdateHistograms() { |
- int64_t now = Clock::GetRealTimeClock()->TimeInMilliseconds(); |
- |
- { |
- CriticalSectionScoped cs(crit_.get()); |
- int64_t elapsed_sec = (now - time_of_first_rtt_ms_) / 1000; |
- if (time_of_first_rtt_ms_ != -1 && num_rtts_ > 0 && |
- elapsed_sec > metrics::kMinRunTimeInSeconds) { |
- int64_t avg_rtt_ms = (rtt_sum_ms_ + num_rtts_ / 2) / num_rtts_; |
- RTC_HISTOGRAM_COUNTS_10000( |
- "WebRTC.Video.AverageRoundTripTimeInMilliseconds", avg_rtt_ms); |
- } |
- } |
- |
- if (sender_) { |
- RtcpPacketTypeCounter rtcp_counter; |
- GetSendRtcpPacketTypeCounter(&rtcp_counter); |
- int64_t elapsed_sec = rtcp_counter.TimeSinceFirstPacketInMs(now) / 1000; |
- if (elapsed_sec > metrics::kMinRunTimeInSeconds) { |
- RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.NackPacketsReceivedPerMinute", |
- rtcp_counter.nack_packets * 60 / elapsed_sec); |
- RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.FirPacketsReceivedPerMinute", |
- rtcp_counter.fir_packets * 60 / elapsed_sec); |
- RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.PliPacketsReceivedPerMinute", |
- rtcp_counter.pli_packets * 60 / elapsed_sec); |
- if (rtcp_counter.nack_requests > 0) { |
- RTC_HISTOGRAM_PERCENTAGE( |
- "WebRTC.Video.UniqueNackRequestsReceivedInPercent", |
- rtcp_counter.UniqueNackRequestsInPercent()); |
- } |
- int fraction_lost = report_block_stats_sender_->FractionLostInPercent(); |
- if (fraction_lost != -1) { |
- RTC_HISTOGRAM_PERCENTAGE("WebRTC.Video.SentPacketsLostInPercent", |
- fraction_lost); |
- } |
- } |
- |
- StreamDataCounters rtp; |
- StreamDataCounters rtx; |
- GetSendStreamDataCounters(&rtp, &rtx); |
- StreamDataCounters rtp_rtx = rtp; |
- rtp_rtx.Add(rtx); |
- elapsed_sec = rtp_rtx.TimeSinceFirstPacketInMs( |
- Clock::GetRealTimeClock()->TimeInMilliseconds()) / |
- 1000; |
- if (elapsed_sec > metrics::kMinRunTimeInSeconds) { |
- RTC_HISTOGRAM_COUNTS_100000( |
- "WebRTC.Video.BitrateSentInKbps", |
- static_cast<int>(rtp_rtx.transmitted.TotalBytes() * 8 / elapsed_sec / |
- 1000)); |
- RTC_HISTOGRAM_COUNTS_10000( |
- "WebRTC.Video.MediaBitrateSentInKbps", |
- static_cast<int>(rtp.MediaPayloadBytes() * 8 / elapsed_sec / 1000)); |
- RTC_HISTOGRAM_COUNTS_10000( |
- "WebRTC.Video.PaddingBitrateSentInKbps", |
- static_cast<int>(rtp_rtx.transmitted.padding_bytes * 8 / elapsed_sec / |
- 1000)); |
- RTC_HISTOGRAM_COUNTS_10000( |
- "WebRTC.Video.RetransmittedBitrateSentInKbps", |
- static_cast<int>(rtp_rtx.retransmitted.TotalBytes() * 8 / |
- elapsed_sec / 1000)); |
- if (rtp_rtcp_modules_[0]->RtxSendStatus() != kRtxOff) { |
- RTC_HISTOGRAM_COUNTS_10000( |
- "WebRTC.Video.RtxBitrateSentInKbps", |
- static_cast<int>(rtx.transmitted.TotalBytes() * 8 / elapsed_sec / |
- 1000)); |
- } |
- bool fec_enabled = false; |
- uint8_t pltype_red; |
- uint8_t pltype_fec; |
- rtp_rtcp_modules_[0]->GenericFECStatus(fec_enabled, pltype_red, |
- pltype_fec); |
- if (fec_enabled) { |
- RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.FecBitrateSentInKbps", |
- static_cast<int>(rtp_rtx.fec.TotalBytes() * |
- 8 / elapsed_sec / 1000)); |
- } |
- } |
- } else if (vie_receiver_.GetRemoteSsrc() > 0) { |
- // Get receive stats if we are receiving packets, i.e. there is a remote |
- // ssrc. |
- RtcpPacketTypeCounter rtcp_counter; |
- GetReceiveRtcpPacketTypeCounter(&rtcp_counter); |
- int64_t elapsed_sec = rtcp_counter.TimeSinceFirstPacketInMs(now) / 1000; |
- if (elapsed_sec > metrics::kMinRunTimeInSeconds) { |
- RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.NackPacketsSentPerMinute", |
- rtcp_counter.nack_packets * 60 / elapsed_sec); |
- RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.FirPacketsSentPerMinute", |
- rtcp_counter.fir_packets * 60 / elapsed_sec); |
- RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.PliPacketsSentPerMinute", |
- rtcp_counter.pli_packets * 60 / elapsed_sec); |
- if (rtcp_counter.nack_requests > 0) { |
- RTC_HISTOGRAM_PERCENTAGE("WebRTC.Video.UniqueNackRequestsSentInPercent", |
- rtcp_counter.UniqueNackRequestsInPercent()); |
- } |
- } |
- |
- StreamDataCounters rtp; |
- StreamDataCounters rtx; |
- GetReceiveStreamDataCounters(&rtp, &rtx); |
- StreamDataCounters rtp_rtx = rtp; |
- rtp_rtx.Add(rtx); |
- elapsed_sec = rtp_rtx.TimeSinceFirstPacketInMs(now) / 1000; |
- if (elapsed_sec > metrics::kMinRunTimeInSeconds) { |
- RTC_HISTOGRAM_COUNTS_10000( |
- "WebRTC.Video.BitrateReceivedInKbps", |
- static_cast<int>(rtp_rtx.transmitted.TotalBytes() * 8 / elapsed_sec / |
- 1000)); |
- RTC_HISTOGRAM_COUNTS_10000( |
- "WebRTC.Video.MediaBitrateReceivedInKbps", |
- static_cast<int>(rtp.MediaPayloadBytes() * 8 / elapsed_sec / 1000)); |
- RTC_HISTOGRAM_COUNTS_10000( |
- "WebRTC.Video.PaddingBitrateReceivedInKbps", |
- static_cast<int>(rtp_rtx.transmitted.padding_bytes * 8 / elapsed_sec / |
- 1000)); |
- RTC_HISTOGRAM_COUNTS_10000( |
- "WebRTC.Video.RetransmittedBitrateReceivedInKbps", |
- static_cast<int>(rtp_rtx.retransmitted.TotalBytes() * 8 / |
- elapsed_sec / 1000)); |
- uint32_t ssrc = 0; |
- if (vie_receiver_.GetRtxSsrc(&ssrc)) { |
- RTC_HISTOGRAM_COUNTS_10000( |
- "WebRTC.Video.RtxBitrateReceivedInKbps", |
- static_cast<int>(rtx.transmitted.TotalBytes() * 8 / elapsed_sec / |
- 1000)); |
- } |
- if (vie_receiver_.IsFecEnabled()) { |
- RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.FecBitrateReceivedInKbps", |
- static_cast<int>(rtp_rtx.fec.TotalBytes() * |
- 8 / elapsed_sec / 1000)); |
- } |
- } |
- } |
-} |
- |
-int32_t ViEChannel::SetSendCodec(const VideoCodec& video_codec, |
- bool new_stream) { |
- RTC_DCHECK(sender_); |
- if (video_codec.codecType == kVideoCodecRED || |
- video_codec.codecType == kVideoCodecULPFEC) { |
- LOG_F(LS_ERROR) << "Not a valid send codec " << video_codec.codecType; |
- return -1; |
- } |
- if (kMaxSimulcastStreams < video_codec.numberOfSimulcastStreams) { |
- LOG_F(LS_ERROR) << "Incorrect config " |
- << video_codec.numberOfSimulcastStreams; |
- return -1; |
- } |
- // Update the RTP module with the settings. |
- // Stop and Start the RTP module -> trigger new SSRC, if an SSRC hasn't been |
- // set explicitly. |
- // The first layer is always active, so the first module can be checked for |
- // sending status. |
- bool is_sending = rtp_rtcp_modules_[0]->Sending(); |
- bool router_was_active = send_payload_router_->active(); |
- send_payload_router_->set_active(false); |
- send_payload_router_->SetSendingRtpModules(std::list<RtpRtcp*>()); |
- |
- std::vector<RtpRtcp*> registered_modules; |
- std::vector<RtpRtcp*> deregistered_modules; |
- size_t num_active_modules = video_codec.numberOfSimulcastStreams > 0 |
- ? video_codec.numberOfSimulcastStreams |
- : 1; |
- size_t num_prev_active_modules; |
- { |
- // Cache which modules are active so StartSend can know which ones to start. |
- CriticalSectionScoped cs(crit_.get()); |
- num_prev_active_modules = num_active_rtp_rtcp_modules_; |
- num_active_rtp_rtcp_modules_ = num_active_modules; |
- } |
- for (size_t i = 0; i < num_active_modules; ++i) |
- registered_modules.push_back(rtp_rtcp_modules_[i]); |
- |
- for (size_t i = num_active_modules; i < rtp_rtcp_modules_.size(); ++i) |
- deregistered_modules.push_back(rtp_rtcp_modules_[i]); |
- |
- // Disable inactive modules. |
- for (RtpRtcp* rtp_rtcp : deregistered_modules) { |
- rtp_rtcp->SetSendingStatus(false); |
- rtp_rtcp->SetSendingMediaStatus(false); |
- } |
- |
- // Configure active modules. |
- for (RtpRtcp* rtp_rtcp : registered_modules) { |
- rtp_rtcp->DeRegisterSendPayload(video_codec.plType); |
- if (rtp_rtcp->RegisterSendPayload(video_codec) != 0) { |
- return -1; |
- } |
- rtp_rtcp->SetSendingStatus(is_sending); |
- rtp_rtcp->SetSendingMediaStatus(is_sending); |
- } |
- |
- // |RegisterSimulcastRtpRtcpModules| resets all old weak pointers and old |
- // modules can be deleted after this step. |
- vie_receiver_.RegisterRtpRtcpModules(registered_modules); |
- |
- // Update the packet and payload routers with the sending RtpRtcp modules. |
- if (sender_) { |
- std::list<RtpRtcp*> active_send_modules; |
- for (RtpRtcp* rtp_rtcp : registered_modules) |
- active_send_modules.push_back(rtp_rtcp); |
- send_payload_router_->SetSendingRtpModules(active_send_modules); |
- } |
- |
- if (router_was_active) |
- send_payload_router_->set_active(true); |
- |
- // Deregister previously registered modules. |
- for (size_t i = num_active_modules; i < num_prev_active_modules; ++i) { |
- module_process_thread_->DeRegisterModule(rtp_rtcp_modules_[i]); |
- packet_router_->RemoveRtpModule(rtp_rtcp_modules_[i]); |
- } |
- // Register new active modules. |
- for (size_t i = num_prev_active_modules; i < num_active_modules; ++i) { |
- module_process_thread_->RegisterModule(rtp_rtcp_modules_[i]); |
- packet_router_->AddRtpModule(rtp_rtcp_modules_[i]); |
- } |
- return 0; |
-} |
- |
-int32_t ViEChannel::SetReceiveCodec(const VideoCodec& video_codec) { |
- RTC_DCHECK(!sender_); |
- if (!vie_receiver_.SetReceiveCodec(video_codec)) { |
- return -1; |
- } |
- |
- if (video_codec.codecType != kVideoCodecRED && |
- video_codec.codecType != kVideoCodecULPFEC) { |
- // Register codec type with VCM, but do not register RED or ULPFEC. |
- if (vcm_->RegisterReceiveCodec(&video_codec, number_of_cores_, false) != |
- VCM_OK) { |
- return -1; |
- } |
- } |
- return 0; |
-} |
- |
- |
-int32_t ViEChannel::RegisterExternalDecoder(const uint8_t pl_type, |
- VideoDecoder* decoder, |
- bool buffered_rendering, |
- int32_t render_delay) { |
- RTC_DCHECK(!sender_); |
- vcm_->RegisterExternalDecoder(decoder, pl_type, buffered_rendering); |
- return vcm_->SetRenderDelay(render_delay); |
-} |
- |
-int32_t ViEChannel::ReceiveCodecStatistics(uint32_t* num_key_frames, |
- uint32_t* num_delta_frames) { |
- CriticalSectionScoped cs(crit_.get()); |
- *num_key_frames = receive_frame_counts_.key_frames; |
- *num_delta_frames = receive_frame_counts_.delta_frames; |
- return 0; |
-} |
- |
-uint32_t ViEChannel::DiscardedPackets() const { |
- return vcm_->DiscardedPackets(); |
-} |
- |
-int ViEChannel::ReceiveDelay() const { |
- return vcm_->Delay(); |
-} |
- |
-void ViEChannel::SetRTCPMode(const RtcpMode rtcp_mode) { |
- for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) |
- rtp_rtcp->SetRTCPStatus(rtcp_mode); |
-} |
- |
-void ViEChannel::SetProtectionMode(bool enable_nack, |
- bool enable_fec, |
- int payload_type_red, |
- int payload_type_fec) { |
- // Validate payload types. |
- if (enable_fec) { |
- RTC_DCHECK_GE(payload_type_red, 0); |
- RTC_DCHECK_GE(payload_type_fec, 0); |
- RTC_DCHECK_LE(payload_type_red, 127); |
- RTC_DCHECK_LE(payload_type_fec, 127); |
- } else { |
- RTC_DCHECK_EQ(payload_type_red, -1); |
- RTC_DCHECK_EQ(payload_type_fec, -1); |
- // Set to valid uint8_ts to be castable later without signed overflows. |
- payload_type_red = 0; |
- payload_type_fec = 0; |
- } |
- |
- VCMVideoProtection protection_method; |
- if (enable_nack) { |
- protection_method = enable_fec ? kProtectionNackFEC : kProtectionNack; |
- } else { |
- protection_method = kProtectionNone; |
- } |
- |
- vcm_->SetVideoProtection(protection_method, true); |
- |
- // Set NACK. |
- ProcessNACKRequest(enable_nack); |
- |
- // Set FEC. |
- for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) { |
- rtp_rtcp->SetGenericFECStatus(enable_fec, |
- static_cast<uint8_t>(payload_type_red), |
- static_cast<uint8_t>(payload_type_fec)); |
- } |
-} |
- |
-void ViEChannel::ProcessNACKRequest(const bool enable) { |
- if (enable) { |
- // Turn on NACK. |
- if (rtp_rtcp_modules_[0]->RTCP() == RtcpMode::kOff) |
- return; |
- vie_receiver_.SetNackStatus(true, max_nack_reordering_threshold_); |
- |
- for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) |
- rtp_rtcp->SetStorePacketsStatus(true, nack_history_size_sender_); |
- |
- vcm_->RegisterPacketRequestCallback(this); |
- // Don't introduce errors when NACK is enabled. |
- vcm_->SetDecodeErrorMode(kNoErrors); |
- } else { |
- vcm_->RegisterPacketRequestCallback(NULL); |
- if (paced_sender_ == nullptr) { |
- for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) |
- rtp_rtcp->SetStorePacketsStatus(false, 0); |
- } |
- vie_receiver_.SetNackStatus(false, max_nack_reordering_threshold_); |
- // When NACK is off, allow decoding with errors. Otherwise, the video |
- // will freeze, and will only recover with a complete key frame. |
- vcm_->SetDecodeErrorMode(kWithErrors); |
- } |
-} |
- |
-bool ViEChannel::IsSendingFecEnabled() { |
- bool fec_enabled = false; |
- uint8_t pltype_red = 0; |
- uint8_t pltype_fec = 0; |
- |
- for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) { |
- rtp_rtcp->GenericFECStatus(fec_enabled, pltype_red, pltype_fec); |
- if (fec_enabled) |
- return true; |
- } |
- return false; |
-} |
- |
-int ViEChannel::SetSenderBufferingMode(int target_delay_ms) { |
- if ((target_delay_ms < 0) || (target_delay_ms > kMaxTargetDelayMs)) { |
- LOG(LS_ERROR) << "Invalid send buffer value."; |
- return -1; |
- } |
- if (target_delay_ms == 0) { |
- // Real-time mode. |
- nack_history_size_sender_ = kMinSendSidePacketHistorySize; |
- } else { |
- nack_history_size_sender_ = GetRequiredNackListSize(target_delay_ms); |
- // Don't allow a number lower than the default value. |
- if (nack_history_size_sender_ < kMinSendSidePacketHistorySize) { |
- nack_history_size_sender_ = kMinSendSidePacketHistorySize; |
- } |
- } |
- for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) |
- rtp_rtcp->SetStorePacketsStatus(true, nack_history_size_sender_); |
- return 0; |
-} |
- |
-int ViEChannel::GetRequiredNackListSize(int target_delay_ms) { |
- // The max size of the nack list should be large enough to accommodate the |
- // the number of packets (frames) resulting from the increased delay. |
- // Roughly estimating for ~40 packets per frame @ 30fps. |
- return target_delay_ms * 40 * 30 / 1000; |
-} |
- |
-int ViEChannel::SetSendTimestampOffsetStatus(bool enable, int id) { |
- // Disable any previous registrations of this extension to avoid errors. |
- for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) { |
- rtp_rtcp->DeregisterSendRtpHeaderExtension( |
- kRtpExtensionTransmissionTimeOffset); |
- } |
- if (!enable) |
- return 0; |
- // Enable the extension. |
- int error = 0; |
- for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) { |
- error |= rtp_rtcp->RegisterSendRtpHeaderExtension( |
- kRtpExtensionTransmissionTimeOffset, id); |
- } |
- return error; |
-} |
- |
-int ViEChannel::SetReceiveTimestampOffsetStatus(bool enable, int id) { |
- return vie_receiver_.SetReceiveTimestampOffsetStatus(enable, id) ? 0 : -1; |
-} |
- |
-int ViEChannel::SetSendAbsoluteSendTimeStatus(bool enable, int id) { |
- // Disable any previous registrations of this extension to avoid errors. |
- for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) |
- rtp_rtcp->DeregisterSendRtpHeaderExtension(kRtpExtensionAbsoluteSendTime); |
- if (!enable) |
- return 0; |
- // Enable the extension. |
- int error = 0; |
- for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) { |
- error |= rtp_rtcp->RegisterSendRtpHeaderExtension( |
- kRtpExtensionAbsoluteSendTime, id); |
- } |
- return error; |
-} |
- |
-int ViEChannel::SetReceiveAbsoluteSendTimeStatus(bool enable, int id) { |
- return vie_receiver_.SetReceiveAbsoluteSendTimeStatus(enable, id) ? 0 : -1; |
-} |
- |
-int ViEChannel::SetSendVideoRotationStatus(bool enable, int id) { |
- // Disable any previous registrations of this extension to avoid errors. |
- for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) |
- rtp_rtcp->DeregisterSendRtpHeaderExtension(kRtpExtensionVideoRotation); |
- if (!enable) |
- return 0; |
- // Enable the extension. |
- int error = 0; |
- for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) { |
- error |= rtp_rtcp->RegisterSendRtpHeaderExtension( |
- kRtpExtensionVideoRotation, id); |
- } |
- return error; |
-} |
- |
-int ViEChannel::SetReceiveVideoRotationStatus(bool enable, int id) { |
- return vie_receiver_.SetReceiveVideoRotationStatus(enable, id) ? 0 : -1; |
-} |
- |
-int ViEChannel::SetSendTransportSequenceNumber(bool enable, int id) { |
- // Disable any previous registrations of this extension to avoid errors. |
- for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) { |
- rtp_rtcp->DeregisterSendRtpHeaderExtension( |
- kRtpExtensionTransportSequenceNumber); |
- } |
- if (!enable) |
- return 0; |
- // Enable the extension. |
- int error = 0; |
- for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) { |
- error |= rtp_rtcp->RegisterSendRtpHeaderExtension( |
- kRtpExtensionTransportSequenceNumber, id); |
- } |
- return error; |
-} |
- |
-int ViEChannel::SetReceiveTransportSequenceNumber(bool enable, int id) { |
- return vie_receiver_.SetReceiveTransportSequenceNumber(enable, id) ? 0 : -1; |
-} |
- |
-void ViEChannel::SetRtcpXrRrtrStatus(bool enable) { |
- rtp_rtcp_modules_[0]->SetRtcpXrRrtrStatus(enable); |
-} |
- |
-void ViEChannel::EnableTMMBR(bool enable) { |
- rtp_rtcp_modules_[0]->SetTMMBRStatus(enable); |
-} |
- |
-int32_t ViEChannel::SetSSRC(const uint32_t SSRC, |
- const StreamType usage, |
- const uint8_t simulcast_idx) { |
- RtpRtcp* rtp_rtcp = rtp_rtcp_modules_[simulcast_idx]; |
- if (usage == kViEStreamTypeRtx) { |
- rtp_rtcp->SetRtxSsrc(SSRC); |
- } else { |
- rtp_rtcp->SetSSRC(SSRC); |
- } |
- return 0; |
-} |
- |
-int32_t ViEChannel::SetRemoteSSRCType(const StreamType usage, |
- const uint32_t SSRC) { |
- vie_receiver_.SetRtxSsrc(SSRC); |
- return 0; |
-} |
- |
-int32_t ViEChannel::GetLocalSSRC(uint8_t idx, unsigned int* ssrc) { |
- RTC_DCHECK_LE(idx, rtp_rtcp_modules_.size()); |
- *ssrc = rtp_rtcp_modules_[idx]->SSRC(); |
- return 0; |
-} |
- |
-uint32_t ViEChannel::GetRemoteSSRC() { |
- return vie_receiver_.GetRemoteSsrc(); |
-} |
- |
-int ViEChannel::SetRtxSendPayloadType(int payload_type, |
- int associated_payload_type) { |
- for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) |
- rtp_rtcp->SetRtxSendPayloadType(payload_type, associated_payload_type); |
- SetRtxSendStatus(true); |
- return 0; |
-} |
- |
-void ViEChannel::SetRtxSendStatus(bool enable) { |
- int rtx_settings = |
- enable ? kRtxRetransmitted | kRtxRedundantPayloads : kRtxOff; |
- for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) |
- rtp_rtcp->SetRtxSendStatus(rtx_settings); |
-} |
- |
-void ViEChannel::SetRtxReceivePayloadType(int payload_type, |
- int associated_payload_type) { |
- vie_receiver_.SetRtxPayloadType(payload_type, associated_payload_type); |
-} |
- |
-void ViEChannel::SetUseRtxPayloadMappingOnRestore(bool val) { |
- vie_receiver_.SetUseRtxPayloadMappingOnRestore(val); |
-} |
- |
-void ViEChannel::SetRtpStateForSsrc(uint32_t ssrc, const RtpState& rtp_state) { |
- RTC_DCHECK(!rtp_rtcp_modules_[0]->Sending()); |
- for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) { |
- if (rtp_rtcp->SetRtpStateForSsrc(ssrc, rtp_state)) |
- return; |
- } |
-} |
- |
-RtpState ViEChannel::GetRtpStateForSsrc(uint32_t ssrc) { |
- RTC_DCHECK(!rtp_rtcp_modules_[0]->Sending()); |
- RtpState rtp_state; |
- for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) { |
- if (rtp_rtcp->GetRtpStateForSsrc(ssrc, &rtp_state)) |
- return rtp_state; |
- } |
- LOG(LS_ERROR) << "Couldn't get RTP state for ssrc: " << ssrc; |
- return rtp_state; |
-} |
- |
-// TODO(pbos): Set CNAME on all modules. |
-int32_t ViEChannel::SetRTCPCName(const char* rtcp_cname) { |
- RTC_DCHECK(!rtp_rtcp_modules_[0]->Sending()); |
- return rtp_rtcp_modules_[0]->SetCNAME(rtcp_cname); |
-} |
- |
-int32_t ViEChannel::GetRemoteRTCPCName(char rtcp_cname[]) { |
- uint32_t remoteSSRC = vie_receiver_.GetRemoteSsrc(); |
- return rtp_rtcp_modules_[0]->RemoteCNAME(remoteSSRC, rtcp_cname); |
-} |
- |
-int32_t ViEChannel::GetSendRtcpStatistics(uint16_t* fraction_lost, |
- uint32_t* cumulative_lost, |
- uint32_t* extended_max, |
- uint32_t* jitter_samples, |
- int64_t* rtt_ms) { |
- // Aggregate the report blocks associated with streams sent on this channel. |
- std::vector<RTCPReportBlock> report_blocks; |
- for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) |
- rtp_rtcp->RemoteRTCPStat(&report_blocks); |
- |
- if (report_blocks.empty()) |
- return -1; |
- |
- uint32_t remote_ssrc = vie_receiver_.GetRemoteSsrc(); |
- std::vector<RTCPReportBlock>::const_iterator it = report_blocks.begin(); |
- for (; it != report_blocks.end(); ++it) { |
- if (it->remoteSSRC == remote_ssrc) |
- break; |
- } |
- if (it == report_blocks.end()) { |
- // We have not received packets with an SSRC matching the report blocks. To |
- // have a chance of calculating an RTT we will try with the SSRC of the |
- // first report block received. |
- // This is very important for send-only channels where we don't know the |
- // SSRC of the other end. |
- remote_ssrc = report_blocks[0].remoteSSRC; |
- } |
- |
- // TODO(asapersson): Change report_block_stats to not rely on |
- // GetSendRtcpStatistics to be called. |
- RTCPReportBlock report = |
- report_block_stats_sender_->AggregateAndStore(report_blocks); |
- *fraction_lost = report.fractionLost; |
- *cumulative_lost = report.cumulativeLost; |
- *extended_max = report.extendedHighSeqNum; |
- *jitter_samples = report.jitter; |
- |
- int64_t dummy; |
- int64_t rtt = 0; |
- if (rtp_rtcp_modules_[0]->RTT(remote_ssrc, &rtt, &dummy, &dummy, &dummy) != |
- 0) { |
- return -1; |
- } |
- *rtt_ms = rtt; |
- return 0; |
-} |
- |
-void ViEChannel::RegisterSendChannelRtcpStatisticsCallback( |
- RtcpStatisticsCallback* callback) { |
- for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) |
- rtp_rtcp->RegisterRtcpStatisticsCallback(callback); |
-} |
- |
-void ViEChannel::RegisterReceiveChannelRtcpStatisticsCallback( |
- RtcpStatisticsCallback* callback) { |
- vie_receiver_.GetReceiveStatistics()->RegisterRtcpStatisticsCallback( |
- callback); |
- rtp_rtcp_modules_[0]->RegisterRtcpStatisticsCallback(callback); |
-} |
- |
-void ViEChannel::RegisterRtcpPacketTypeCounterObserver( |
- RtcpPacketTypeCounterObserver* observer) { |
- rtcp_packet_type_counter_observer_.Set(observer); |
-} |
- |
-void ViEChannel::GetSendStreamDataCounters( |
- StreamDataCounters* rtp_counters, |
- StreamDataCounters* rtx_counters) const { |
- *rtp_counters = StreamDataCounters(); |
- *rtx_counters = StreamDataCounters(); |
- for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) { |
- StreamDataCounters rtp_data; |
- StreamDataCounters rtx_data; |
- rtp_rtcp->GetSendStreamDataCounters(&rtp_data, &rtx_data); |
- rtp_counters->Add(rtp_data); |
- rtx_counters->Add(rtx_data); |
- } |
-} |
- |
-void ViEChannel::GetReceiveStreamDataCounters( |
- StreamDataCounters* rtp_counters, |
- StreamDataCounters* rtx_counters) const { |
- StreamStatistician* statistician = vie_receiver_.GetReceiveStatistics()-> |
- GetStatistician(vie_receiver_.GetRemoteSsrc()); |
- if (statistician) { |
- statistician->GetReceiveStreamDataCounters(rtp_counters); |
- } |
- uint32_t rtx_ssrc = 0; |
- if (vie_receiver_.GetRtxSsrc(&rtx_ssrc)) { |
- StreamStatistician* statistician = |
- vie_receiver_.GetReceiveStatistics()->GetStatistician(rtx_ssrc); |
- if (statistician) { |
- statistician->GetReceiveStreamDataCounters(rtx_counters); |
- } |
- } |
-} |
- |
-void ViEChannel::RegisterSendChannelRtpStatisticsCallback( |
- StreamDataCountersCallback* callback) { |
- for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) |
- rtp_rtcp->RegisterSendChannelRtpStatisticsCallback(callback); |
-} |
- |
-void ViEChannel::RegisterReceiveChannelRtpStatisticsCallback( |
- StreamDataCountersCallback* callback) { |
- vie_receiver_.GetReceiveStatistics()->RegisterRtpStatisticsCallback(callback); |
-} |
- |
-void ViEChannel::GetSendRtcpPacketTypeCounter( |
- RtcpPacketTypeCounter* packet_counter) const { |
- std::map<uint32_t, RtcpPacketTypeCounter> counter_map = |
- rtcp_packet_type_counter_observer_.GetPacketTypeCounterMap(); |
- |
- RtcpPacketTypeCounter counter; |
- for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) |
- counter.Add(counter_map[rtp_rtcp->SSRC()]); |
- *packet_counter = counter; |
-} |
- |
-void ViEChannel::GetReceiveRtcpPacketTypeCounter( |
- RtcpPacketTypeCounter* packet_counter) const { |
- std::map<uint32_t, RtcpPacketTypeCounter> counter_map = |
- rtcp_packet_type_counter_observer_.GetPacketTypeCounterMap(); |
- |
- RtcpPacketTypeCounter counter; |
- counter.Add(counter_map[vie_receiver_.GetRemoteSsrc()]); |
- |
- *packet_counter = counter; |
-} |
- |
-void ViEChannel::RegisterSendSideDelayObserver( |
- SendSideDelayObserver* observer) { |
- send_side_delay_observer_.Set(observer); |
-} |
- |
-void ViEChannel::RegisterSendBitrateObserver( |
- BitrateStatisticsObserver* observer) { |
- send_bitrate_observer_.Set(observer); |
-} |
- |
-int32_t ViEChannel::StartSend() { |
- CriticalSectionScoped cs(crit_.get()); |
- |
- if (rtp_rtcp_modules_[0]->Sending()) |
- return -1; |
- |
- for (size_t i = 0; i < num_active_rtp_rtcp_modules_; ++i) { |
- RtpRtcp* rtp_rtcp = rtp_rtcp_modules_[i]; |
- rtp_rtcp->SetSendingMediaStatus(true); |
- rtp_rtcp->SetSendingStatus(true); |
- } |
- send_payload_router_->set_active(true); |
- return 0; |
-} |
- |
-int32_t ViEChannel::StopSend() { |
- send_payload_router_->set_active(false); |
- for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) |
- rtp_rtcp->SetSendingMediaStatus(false); |
- |
- if (!rtp_rtcp_modules_[0]->Sending()) { |
- return -1; |
- } |
- |
- for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) { |
- rtp_rtcp->SetSendingStatus(false); |
- } |
- return 0; |
-} |
- |
-bool ViEChannel::Sending() { |
- return rtp_rtcp_modules_[0]->Sending(); |
-} |
- |
-void ViEChannel::StartReceive() { |
- if (!sender_) |
- StartDecodeThread(); |
- vie_receiver_.StartReceive(); |
-} |
- |
-void ViEChannel::StopReceive() { |
- vie_receiver_.StopReceive(); |
- if (!sender_) { |
- StopDecodeThread(); |
- vcm_->ResetDecoder(); |
- } |
-} |
- |
-int32_t ViEChannel::ReceivedRTPPacket(const void* rtp_packet, |
- size_t rtp_packet_length, |
- const PacketTime& packet_time) { |
- return vie_receiver_.ReceivedRTPPacket( |
- rtp_packet, rtp_packet_length, packet_time); |
-} |
- |
-int32_t ViEChannel::ReceivedRTCPPacket(const void* rtcp_packet, |
- size_t rtcp_packet_length) { |
- return vie_receiver_.ReceivedRTCPPacket(rtcp_packet, rtcp_packet_length); |
-} |
- |
-int32_t ViEChannel::SetMTU(uint16_t mtu) { |
- for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) |
- rtp_rtcp->SetMaxTransferUnit(mtu); |
- return 0; |
-} |
- |
-RtpRtcp* ViEChannel::rtp_rtcp() { |
- return rtp_rtcp_modules_[0]; |
-} |
- |
-rtc::scoped_refptr<PayloadRouter> ViEChannel::send_payload_router() { |
- return send_payload_router_; |
-} |
- |
-VCMProtectionCallback* ViEChannel::vcm_protection_callback() { |
- return vcm_protection_callback_.get(); |
-} |
- |
-CallStatsObserver* ViEChannel::GetStatsObserver() { |
- return stats_observer_.get(); |
-} |
- |
-// Do not acquire the lock of |vcm_| in this function. Decode callback won't |
-// necessarily be called from the decoding thread. The decoding thread may have |
-// held the lock when calling VideoDecoder::Decode, Reset, or Release. Acquiring |
-// the same lock in the path of decode callback can deadlock. |
-int32_t ViEChannel::FrameToRender(VideoFrame& video_frame) { // NOLINT |
- CriticalSectionScoped cs(crit_.get()); |
- |
- if (pre_render_callback_ != NULL) |
- pre_render_callback_->FrameCallback(&video_frame); |
- |
- // TODO(pbos): Remove stream id argument. |
- incoming_video_stream_->RenderFrame(0xFFFFFFFF, video_frame); |
- return 0; |
-} |
- |
-int32_t ViEChannel::ReceivedDecodedReferenceFrame( |
- const uint64_t picture_id) { |
- return rtp_rtcp_modules_[0]->SendRTCPReferencePictureSelection(picture_id); |
-} |
- |
-void ViEChannel::OnIncomingPayloadType(int payload_type) { |
- CriticalSectionScoped cs(crit_.get()); |
- if (receive_stats_callback_) |
- receive_stats_callback_->OnIncomingPayloadType(payload_type); |
-} |
- |
-void ViEChannel::OnReceiveRatesUpdated(uint32_t bit_rate, uint32_t frame_rate) { |
- CriticalSectionScoped cs(crit_.get()); |
- if (receive_stats_callback_) |
- receive_stats_callback_->OnIncomingRate(frame_rate, bit_rate); |
-} |
- |
-void ViEChannel::OnDiscardedPacketsUpdated(int discarded_packets) { |
- CriticalSectionScoped cs(crit_.get()); |
- if (receive_stats_callback_) |
- receive_stats_callback_->OnDiscardedPacketsUpdated(discarded_packets); |
-} |
- |
-void ViEChannel::OnFrameCountsUpdated(const FrameCounts& frame_counts) { |
- CriticalSectionScoped cs(crit_.get()); |
- receive_frame_counts_ = frame_counts; |
- if (receive_stats_callback_) |
- receive_stats_callback_->OnFrameCountsUpdated(frame_counts); |
-} |
- |
-void ViEChannel::OnDecoderTiming(int decode_ms, |
- int max_decode_ms, |
- int current_delay_ms, |
- int target_delay_ms, |
- int jitter_buffer_ms, |
- int min_playout_delay_ms, |
- int render_delay_ms) { |
- CriticalSectionScoped cs(crit_.get()); |
- if (!receive_stats_callback_) |
- return; |
- receive_stats_callback_->OnDecoderTiming( |
- decode_ms, max_decode_ms, current_delay_ms, target_delay_ms, |
- jitter_buffer_ms, min_playout_delay_ms, render_delay_ms, last_rtt_ms_); |
-} |
- |
-int32_t ViEChannel::RequestKeyFrame() { |
- return rtp_rtcp_modules_[0]->RequestKeyFrame(); |
-} |
- |
-int32_t ViEChannel::SliceLossIndicationRequest( |
- const uint64_t picture_id) { |
- return rtp_rtcp_modules_[0]->SendRTCPSliceLossIndication( |
- static_cast<uint8_t>(picture_id)); |
-} |
- |
-int32_t ViEChannel::ResendPackets(const uint16_t* sequence_numbers, |
- uint16_t length) { |
- return rtp_rtcp_modules_[0]->SendNACK(sequence_numbers, length); |
-} |
- |
-bool ViEChannel::ChannelDecodeThreadFunction(void* obj) { |
- return static_cast<ViEChannel*>(obj)->ChannelDecodeProcess(); |
-} |
- |
-bool ViEChannel::ChannelDecodeProcess() { |
- vcm_->Decode(kMaxDecodeWaitTimeMs); |
- return true; |
-} |
- |
-void ViEChannel::OnRttUpdate(int64_t avg_rtt_ms, int64_t max_rtt_ms) { |
- vcm_->SetReceiveChannelParameters(max_rtt_ms); |
- |
- CriticalSectionScoped cs(crit_.get()); |
- if (time_of_first_rtt_ms_ == -1) |
- time_of_first_rtt_ms_ = Clock::GetRealTimeClock()->TimeInMilliseconds(); |
- rtt_sum_ms_ += avg_rtt_ms; |
- last_rtt_ms_ = avg_rtt_ms; |
- ++num_rtts_; |
-} |
- |
-int ViEChannel::ProtectionRequest(const FecProtectionParams* delta_fec_params, |
- const FecProtectionParams* key_fec_params, |
- uint32_t* video_rate_bps, |
- uint32_t* nack_rate_bps, |
- uint32_t* fec_rate_bps) { |
- *video_rate_bps = 0; |
- *nack_rate_bps = 0; |
- *fec_rate_bps = 0; |
- for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) { |
- uint32_t not_used = 0; |
- uint32_t module_video_rate = 0; |
- uint32_t module_fec_rate = 0; |
- uint32_t module_nack_rate = 0; |
- rtp_rtcp->SetFecParameters(delta_fec_params, key_fec_params); |
- rtp_rtcp->BitrateSent(¬_used, &module_video_rate, &module_fec_rate, |
- &module_nack_rate); |
- *video_rate_bps += module_video_rate; |
- *nack_rate_bps += module_nack_rate; |
- *fec_rate_bps += module_fec_rate; |
- } |
- return 0; |
-} |
- |
-std::vector<RtpRtcp*> ViEChannel::CreateRtpRtcpModules( |
- bool receiver_only, |
- ReceiveStatistics* receive_statistics, |
- Transport* outgoing_transport, |
- RtcpIntraFrameObserver* intra_frame_callback, |
- RtcpBandwidthObserver* bandwidth_callback, |
- TransportFeedbackObserver* transport_feedback_callback, |
- RtcpRttStats* rtt_stats, |
- RtcpPacketTypeCounterObserver* rtcp_packet_type_counter_observer, |
- RemoteBitrateEstimator* remote_bitrate_estimator, |
- RtpPacketSender* paced_sender, |
- TransportSequenceNumberAllocator* transport_sequence_number_allocator, |
- BitrateStatisticsObserver* send_bitrate_observer, |
- FrameCountObserver* send_frame_count_observer, |
- SendSideDelayObserver* send_side_delay_observer, |
- size_t num_modules) { |
- RTC_DCHECK_GT(num_modules, 0u); |
- RtpRtcp::Configuration configuration; |
- ReceiveStatistics* null_receive_statistics = configuration.receive_statistics; |
- configuration.audio = false; |
- configuration.receiver_only = receiver_only; |
- configuration.receive_statistics = receive_statistics; |
- configuration.outgoing_transport = outgoing_transport; |
- configuration.intra_frame_callback = intra_frame_callback; |
- configuration.rtt_stats = rtt_stats; |
- configuration.rtcp_packet_type_counter_observer = |
- rtcp_packet_type_counter_observer; |
- configuration.paced_sender = paced_sender; |
- configuration.transport_sequence_number_allocator = |
- transport_sequence_number_allocator; |
- configuration.send_bitrate_observer = send_bitrate_observer; |
- configuration.send_frame_count_observer = send_frame_count_observer; |
- configuration.send_side_delay_observer = send_side_delay_observer; |
- configuration.bandwidth_callback = bandwidth_callback; |
- configuration.transport_feedback_callback = transport_feedback_callback; |
- |
- std::vector<RtpRtcp*> modules; |
- for (size_t i = 0; i < num_modules; ++i) { |
- RtpRtcp* rtp_rtcp = RtpRtcp::CreateRtpRtcp(configuration); |
- rtp_rtcp->SetSendingStatus(false); |
- rtp_rtcp->SetSendingMediaStatus(false); |
- rtp_rtcp->SetRTCPStatus(RtcpMode::kCompound); |
- modules.push_back(rtp_rtcp); |
- // Receive statistics and remote bitrate estimator should only be set for |
- // the primary (first) module. |
- configuration.receive_statistics = null_receive_statistics; |
- configuration.remote_bitrate_estimator = nullptr; |
- } |
- return modules; |
-} |
- |
-void ViEChannel::StartDecodeThread() { |
- RTC_DCHECK(!sender_); |
- if (decode_thread_.IsRunning()) |
- return; |
- // Start the decode thread |
- decode_thread_.Start(); |
- decode_thread_.SetPriority(rtc::kHighestPriority); |
-} |
- |
-void ViEChannel::StopDecodeThread() { |
- vcm_->TriggerDecoderShutdown(); |
- |
- decode_thread_.Stop(); |
-} |
- |
-int32_t ViEChannel::SetVoiceChannel(int32_t ve_channel_id, |
- VoEVideoSync* ve_sync_interface) { |
- return vie_sync_.ConfigureSync(ve_channel_id, ve_sync_interface, |
- rtp_rtcp_modules_[0], |
- vie_receiver_.GetRtpReceiver()); |
-} |
- |
-int32_t ViEChannel::VoiceChannel() { |
- return vie_sync_.VoiceChannel(); |
-} |
- |
-void ViEChannel::RegisterPreRenderCallback( |
- I420FrameCallback* pre_render_callback) { |
- CriticalSectionScoped cs(crit_.get()); |
- pre_render_callback_ = pre_render_callback; |
-} |
- |
-void ViEChannel::RegisterPreDecodeImageCallback( |
- EncodedImageCallback* pre_decode_callback) { |
- vcm_->RegisterPreDecodeImageCallback(pre_decode_callback); |
-} |
- |
-// TODO(pbos): Remove OnInitializeDecoder which is called from the RTP module, |
-// any decoder resetting should be handled internally within the VCM. |
-int32_t ViEChannel::OnInitializeDecoder( |
- const int8_t payload_type, |
- const char payload_name[RTP_PAYLOAD_NAME_SIZE], |
- const int frequency, |
- const uint8_t channels, |
- const uint32_t rate) { |
- LOG(LS_INFO) << "OnInitializeDecoder " << static_cast<int>(payload_type) |
- << " " << payload_name; |
- vcm_->ResetDecoder(); |
- |
- return 0; |
-} |
- |
-void ViEChannel::OnIncomingSSRCChanged(const uint32_t ssrc) { |
- rtp_rtcp_modules_[0]->SetRemoteSSRC(ssrc); |
-} |
- |
-void ViEChannel::OnIncomingCSRCChanged(const uint32_t CSRC, const bool added) {} |
- |
-void ViEChannel::RegisterSendFrameCountObserver( |
- FrameCountObserver* observer) { |
- send_frame_count_observer_.Set(observer); |
-} |
- |
-void ViEChannel::RegisterReceiveStatisticsProxy( |
- ReceiveStatisticsProxy* receive_statistics_proxy) { |
- CriticalSectionScoped cs(crit_.get()); |
- receive_stats_callback_ = receive_statistics_proxy; |
-} |
- |
-void ViEChannel::SetIncomingVideoStream( |
- IncomingVideoStream* incoming_video_stream) { |
- CriticalSectionScoped cs(crit_.get()); |
- incoming_video_stream_ = incoming_video_stream; |
-} |
-} // namespace webrtc |