| 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
|
|
|