
TEST=trybots BUG=1812 R=mflodman@webrtc.org, pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/4169004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5209 4adac7df-926f-26a2-2b94-8c16560cd09d
1975 lines
71 KiB
C++
1975 lines
71 KiB
C++
/*
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* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "webrtc/video_engine/vie_channel.h"
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#include <algorithm>
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#include <vector>
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#include "webrtc/common.h"
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#include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
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#include "webrtc/experiments.h"
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#include "webrtc/modules/pacing/include/paced_sender.h"
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#include "webrtc/modules/rtp_rtcp/interface/rtp_receiver.h"
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#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
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#include "webrtc/modules/utility/interface/process_thread.h"
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#include "webrtc/modules/video_coding/main/interface/video_coding.h"
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#include "webrtc/modules/video_processing/main/interface/video_processing.h"
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#include "webrtc/modules/video_render/include/video_render_defines.h"
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#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
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#include "webrtc/system_wrappers/interface/thread_wrapper.h"
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#include "webrtc/system_wrappers/interface/trace.h"
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#include "webrtc/video_engine/call_stats.h"
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#include "webrtc/video_engine/include/vie_codec.h"
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#include "webrtc/video_engine/include/vie_errors.h"
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#include "webrtc/video_engine/include/vie_image_process.h"
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#include "webrtc/video_engine/include/vie_rtp_rtcp.h"
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#include "webrtc/frame_callback.h"
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#include "webrtc/video_engine/vie_defines.h"
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namespace webrtc {
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const int kMaxDecodeWaitTimeMs = 50;
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const int kInvalidRtpExtensionId = 0;
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static const int kMaxTargetDelayMs = 10000;
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static const float kMaxIncompleteTimeMultiplier = 3.5f;
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// Helper class receiving statistics callbacks.
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class ChannelStatsObserver : public CallStatsObserver {
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public:
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explicit ChannelStatsObserver(ViEChannel* owner) : owner_(owner) {}
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virtual ~ChannelStatsObserver() {}
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// Implements StatsObserver.
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virtual void OnRttUpdate(uint32_t rtt) {
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owner_->OnRttUpdate(rtt);
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}
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private:
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ViEChannel* owner_;
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};
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ViEChannel::ViEChannel(int32_t channel_id,
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int32_t engine_id,
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uint32_t number_of_cores,
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const Config& config,
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ProcessThread& module_process_thread,
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RtcpIntraFrameObserver* intra_frame_observer,
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RtcpBandwidthObserver* bandwidth_observer,
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RemoteBitrateEstimator* remote_bitrate_estimator,
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RtcpRttStats* rtt_stats,
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PacedSender* paced_sender,
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RtpRtcp* default_rtp_rtcp,
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bool sender)
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: ViEFrameProviderBase(channel_id, engine_id),
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channel_id_(channel_id),
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engine_id_(engine_id),
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number_of_cores_(number_of_cores),
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num_socket_threads_(kViESocketThreads),
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callback_cs_(CriticalSectionWrapper::CreateCriticalSection()),
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rtp_rtcp_cs_(CriticalSectionWrapper::CreateCriticalSection()),
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default_rtp_rtcp_(default_rtp_rtcp),
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vcm_(*VideoCodingModule::Create(ViEModuleId(engine_id, channel_id))),
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vie_receiver_(channel_id, &vcm_, remote_bitrate_estimator, this),
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vie_sender_(channel_id),
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vie_sync_(&vcm_, this),
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stats_observer_(new ChannelStatsObserver(this)),
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module_process_thread_(module_process_thread),
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codec_observer_(NULL),
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do_key_frame_callbackRequest_(false),
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rtp_observer_(NULL),
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rtcp_observer_(NULL),
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intra_frame_observer_(intra_frame_observer),
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rtt_stats_(rtt_stats),
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paced_sender_(paced_sender),
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bandwidth_observer_(bandwidth_observer),
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send_timestamp_extension_id_(kInvalidRtpExtensionId),
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absolute_send_time_extension_id_(kInvalidRtpExtensionId),
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receive_absolute_send_time_enabled_(false),
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external_transport_(NULL),
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decoder_reset_(true),
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wait_for_key_frame_(false),
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decode_thread_(NULL),
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external_encryption_(NULL),
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effect_filter_(NULL),
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color_enhancement_(false),
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mtu_(0),
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sender_(sender),
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nack_history_size_sender_(kSendSidePacketHistorySize),
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max_nack_reordering_threshold_(kMaxPacketAgeToNack),
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pre_render_callback_(NULL),
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config_(config) {
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WEBRTC_TRACE(kTraceMemory, kTraceVideo, ViEId(engine_id, channel_id),
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"ViEChannel::ViEChannel(channel_id: %d, engine_id: %d)",
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channel_id, engine_id);
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RtpRtcp::Configuration configuration;
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configuration.id = ViEModuleId(engine_id, channel_id);
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configuration.audio = false;
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configuration.default_module = default_rtp_rtcp;
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configuration.outgoing_transport = &vie_sender_;
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configuration.rtcp_feedback = this;
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configuration.intra_frame_callback = intra_frame_observer;
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configuration.bandwidth_callback = bandwidth_observer;
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configuration.rtt_stats = rtt_stats;
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configuration.remote_bitrate_estimator = remote_bitrate_estimator;
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configuration.paced_sender = paced_sender;
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configuration.receive_statistics = vie_receiver_.GetReceiveStatistics();
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rtp_rtcp_.reset(RtpRtcp::CreateRtpRtcp(configuration));
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vie_receiver_.SetRtpRtcpModule(rtp_rtcp_.get());
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vcm_.SetNackSettings(kMaxNackListSize, max_nack_reordering_threshold_, 0);
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}
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int32_t ViEChannel::Init() {
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WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
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"%s: channel_id: %d, engine_id: %d)", __FUNCTION__, channel_id_,
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engine_id_);
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if (module_process_thread_.RegisterModule(
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vie_receiver_.GetReceiveStatistics()) != 0) {
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WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
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"%s: Failed to register receive-statistics to process thread",
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__FUNCTION__);
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return -1;
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}
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// RTP/RTCP initialization.
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if (rtp_rtcp_->SetSendingMediaStatus(false) != 0) {
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WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
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"%s: RTP::SetSendingMediaStatus failure", __FUNCTION__);
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return -1;
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}
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if (module_process_thread_.RegisterModule(rtp_rtcp_.get()) != 0) {
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WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
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"%s: RTP::RegisterModule failure", __FUNCTION__);
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return -1;
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}
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if (rtp_rtcp_->SetKeyFrameRequestMethod(kKeyFrameReqFirRtp) != 0) {
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WEBRTC_TRACE(kTraceWarning, kTraceVideo, ViEId(engine_id_, channel_id_),
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"%s: RTP::SetKeyFrameRequestMethod failure", __FUNCTION__);
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}
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if (rtp_rtcp_->SetRTCPStatus(kRtcpCompound) != 0) {
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WEBRTC_TRACE(kTraceWarning, kTraceVideo, ViEId(engine_id_, channel_id_),
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"%s: RTP::SetRTCPStatus failure", __FUNCTION__);
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}
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if (paced_sender_) {
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if (rtp_rtcp_->SetStorePacketsStatus(true, nack_history_size_sender_) !=
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0) {
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WEBRTC_TRACE(kTraceWarning, kTraceVideo, ViEId(engine_id_, channel_id_),
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"%s:SetStorePacketsStatus failure", __FUNCTION__);
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}
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}
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// VCM initialization
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if (vcm_.InitializeReceiver() != 0) {
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WEBRTC_TRACE(kTraceError, kTraceVideo,
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ViEId(engine_id_, channel_id_),
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"%s: VCM::InitializeReceiver failure", __FUNCTION__);
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return -1;
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}
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if (vcm_.SetVideoProtection(kProtectionKeyOnLoss, true)) {
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WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
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"%s: VCM::SetVideoProtection failure", __FUNCTION__);
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return -1;
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}
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if (vcm_.RegisterReceiveCallback(this) != 0) {
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WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
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"%s: VCM::RegisterReceiveCallback failure", __FUNCTION__);
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return -1;
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}
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if (vcm_.RegisterFrameTypeCallback(this) != 0) {
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WEBRTC_TRACE(kTraceWarning, kTraceVideo, ViEId(engine_id_, channel_id_),
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"%s: VCM::RegisterFrameTypeCallback failure", __FUNCTION__);
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}
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if (vcm_.RegisterReceiveStatisticsCallback(this) != 0) {
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WEBRTC_TRACE(kTraceWarning, kTraceVideo, ViEId(engine_id_, channel_id_),
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"%s: VCM::RegisterReceiveStatisticsCallback failure",
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__FUNCTION__);
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}
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if (vcm_.RegisterDecoderTimingCallback(this) != 0) {
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WEBRTC_TRACE(kTraceWarning, kTraceVideo, ViEId(engine_id_, channel_id_),
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"%s: VCM::RegisterDecoderTimingCallback failure",
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__FUNCTION__);
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}
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if (vcm_.SetRenderDelay(kViEDefaultRenderDelayMs) != 0) {
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WEBRTC_TRACE(kTraceWarning, kTraceVideo, ViEId(engine_id_, channel_id_),
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"%s: VCM::SetRenderDelay failure", __FUNCTION__);
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}
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if (module_process_thread_.RegisterModule(&vcm_) != 0) {
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WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
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"%s: VCM::RegisterModule(vcm) failure", __FUNCTION__);
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return -1;
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}
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#ifdef VIDEOCODEC_VP8
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VideoCodec video_codec;
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if (vcm_.Codec(kVideoCodecVP8, &video_codec) == VCM_OK) {
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rtp_rtcp_->RegisterSendPayload(video_codec);
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// TODO(holmer): Can we call SetReceiveCodec() here instead?
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if (!vie_receiver_.RegisterPayload(video_codec)) {
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return -1;
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}
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vcm_.RegisterReceiveCodec(&video_codec, number_of_cores_);
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vcm_.RegisterSendCodec(&video_codec, number_of_cores_,
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rtp_rtcp_->MaxDataPayloadLength());
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} else {
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assert(false);
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}
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#endif
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return 0;
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}
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ViEChannel::~ViEChannel() {
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WEBRTC_TRACE(kTraceMemory, kTraceVideo, ViEId(engine_id_, channel_id_),
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"ViEChannel Destructor, channel_id: %d, engine_id: %d",
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channel_id_, engine_id_);
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// Make sure we don't get more callbacks from the RTP module.
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module_process_thread_.DeRegisterModule(vie_receiver_.GetReceiveStatistics());
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module_process_thread_.DeRegisterModule(rtp_rtcp_.get());
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module_process_thread_.DeRegisterModule(&vcm_);
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module_process_thread_.DeRegisterModule(&vie_sync_);
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while (simulcast_rtp_rtcp_.size() > 0) {
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std::list<RtpRtcp*>::iterator it = simulcast_rtp_rtcp_.begin();
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RtpRtcp* rtp_rtcp = *it;
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module_process_thread_.DeRegisterModule(rtp_rtcp);
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delete rtp_rtcp;
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simulcast_rtp_rtcp_.erase(it);
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}
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while (removed_rtp_rtcp_.size() > 0) {
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std::list<RtpRtcp*>::iterator it = removed_rtp_rtcp_.begin();
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delete *it;
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removed_rtp_rtcp_.erase(it);
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}
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if (decode_thread_) {
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StopDecodeThread();
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}
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// Release modules.
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VideoCodingModule::Destroy(&vcm_);
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}
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int32_t ViEChannel::SetSendCodec(const VideoCodec& video_codec,
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bool new_stream) {
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WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
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"%s: codec_type: %d", __FUNCTION__, video_codec.codecType);
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if (!sender_) {
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return 0;
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}
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if (video_codec.codecType == kVideoCodecRED ||
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video_codec.codecType == kVideoCodecULPFEC) {
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WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
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"%s: codec_type: %d is not a valid send codec.", __FUNCTION__,
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video_codec.codecType);
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return -1;
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}
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if (kMaxSimulcastStreams < video_codec.numberOfSimulcastStreams) {
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WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
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"%s: Too many simulcast streams", __FUNCTION__);
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return -1;
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}
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// Update the RTP module with the settings.
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// Stop and Start the RTP module -> trigger new SSRC, if an SSRC hasn't been
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// set explicitly.
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bool restart_rtp = false;
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if (rtp_rtcp_->Sending() && new_stream) {
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restart_rtp = true;
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rtp_rtcp_->SetSendingStatus(false);
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for (std::list<RtpRtcp*>::iterator it = simulcast_rtp_rtcp_.begin();
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it != simulcast_rtp_rtcp_.end(); ++it) {
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(*it)->SetSendingStatus(false);
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(*it)->SetSendingMediaStatus(false);
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}
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}
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bool fec_enabled = false;
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uint8_t payload_type_red;
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uint8_t payload_type_fec;
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rtp_rtcp_->GenericFECStatus(fec_enabled, payload_type_red, payload_type_fec);
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CriticalSectionScoped cs(rtp_rtcp_cs_.get());
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if (video_codec.numberOfSimulcastStreams > 0) {
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// Set correct bitrate to base layer.
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// Create our simulcast RTP modules.
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int num_modules_to_add = video_codec.numberOfSimulcastStreams -
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simulcast_rtp_rtcp_.size() - 1;
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if (num_modules_to_add < 0) {
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num_modules_to_add = 0;
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}
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while (removed_rtp_rtcp_.size() > 0 && num_modules_to_add > 0) {
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RtpRtcp* rtp_rtcp = removed_rtp_rtcp_.front();
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removed_rtp_rtcp_.pop_front();
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simulcast_rtp_rtcp_.push_back(rtp_rtcp);
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rtp_rtcp->SetSendingStatus(rtp_rtcp_->Sending());
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rtp_rtcp->SetSendingMediaStatus(rtp_rtcp_->SendingMedia());
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module_process_thread_.RegisterModule(rtp_rtcp);
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--num_modules_to_add;
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}
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for (int i = 0; i < num_modules_to_add; ++i) {
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RtpRtcp::Configuration configuration;
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configuration.id = ViEModuleId(engine_id_, channel_id_);
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configuration.audio = false; // Video.
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configuration.default_module = default_rtp_rtcp_;
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configuration.outgoing_transport = &vie_sender_;
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configuration.intra_frame_callback = intra_frame_observer_;
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configuration.bandwidth_callback = bandwidth_observer_.get();
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configuration.rtt_stats = rtt_stats_;
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configuration.paced_sender = paced_sender_;
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RtpRtcp* rtp_rtcp = RtpRtcp::CreateRtpRtcp(configuration);
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// Silently ignore error.
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module_process_thread_.RegisterModule(rtp_rtcp);
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if (rtp_rtcp->SetRTCPStatus(rtp_rtcp_->RTCP()) != 0) {
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WEBRTC_TRACE(kTraceWarning, kTraceVideo, ViEId(engine_id_, channel_id_),
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"%s: RTP::SetRTCPStatus failure", __FUNCTION__);
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}
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if (rtp_rtcp_->StorePackets()) {
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rtp_rtcp->SetStorePacketsStatus(true, nack_history_size_sender_);
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} else if (paced_sender_) {
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rtp_rtcp->SetStorePacketsStatus(true, nack_history_size_sender_);
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}
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if (fec_enabled) {
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rtp_rtcp->SetGenericFECStatus(fec_enabled, payload_type_red,
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payload_type_fec);
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}
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rtp_rtcp->SetSendingStatus(rtp_rtcp_->Sending());
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rtp_rtcp->SetSendingMediaStatus(rtp_rtcp_->SendingMedia());
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simulcast_rtp_rtcp_.push_back(rtp_rtcp);
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}
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// Remove last in list if we have too many.
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for (int j = simulcast_rtp_rtcp_.size();
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j > (video_codec.numberOfSimulcastStreams - 1);
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j--) {
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RtpRtcp* rtp_rtcp = simulcast_rtp_rtcp_.back();
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module_process_thread_.DeRegisterModule(rtp_rtcp);
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rtp_rtcp->SetSendingStatus(false);
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rtp_rtcp->SetSendingMediaStatus(false);
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simulcast_rtp_rtcp_.pop_back();
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removed_rtp_rtcp_.push_front(rtp_rtcp);
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}
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uint8_t idx = 0;
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// Configure all simulcast modules.
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for (std::list<RtpRtcp*>::iterator it = simulcast_rtp_rtcp_.begin();
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it != simulcast_rtp_rtcp_.end();
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it++) {
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idx++;
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RtpRtcp* rtp_rtcp = *it;
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rtp_rtcp->DeRegisterSendPayload(video_codec.plType);
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if (rtp_rtcp->RegisterSendPayload(video_codec) != 0) {
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WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
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"%s: could not register payload type", __FUNCTION__);
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return -1;
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}
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if (mtu_ != 0) {
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rtp_rtcp->SetMaxTransferUnit(mtu_);
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}
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if (restart_rtp) {
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rtp_rtcp->SetSendingStatus(true);
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rtp_rtcp->SetSendingMediaStatus(true);
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}
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if (send_timestamp_extension_id_ != kInvalidRtpExtensionId) {
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// Deregister in case the extension was previously enabled.
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rtp_rtcp->DeregisterSendRtpHeaderExtension(
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kRtpExtensionTransmissionTimeOffset);
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if (rtp_rtcp->RegisterSendRtpHeaderExtension(
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kRtpExtensionTransmissionTimeOffset,
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send_timestamp_extension_id_) != 0) {
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WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
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"%s: could not register transmission time extension",
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__FUNCTION__);
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}
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} else {
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rtp_rtcp->DeregisterSendRtpHeaderExtension(
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kRtpExtensionTransmissionTimeOffset);
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}
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if (absolute_send_time_extension_id_ != kInvalidRtpExtensionId) {
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// Deregister in case the extension was previously enabled.
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rtp_rtcp->DeregisterSendRtpHeaderExtension(
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kRtpExtensionAbsoluteSendTime);
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if (rtp_rtcp->RegisterSendRtpHeaderExtension(
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kRtpExtensionAbsoluteSendTime,
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absolute_send_time_extension_id_) != 0) {
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WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
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"%s: could not register absolute send time extension",
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__FUNCTION__);
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}
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} else {
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rtp_rtcp->DeregisterSendRtpHeaderExtension(
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kRtpExtensionAbsoluteSendTime);
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}
|
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rtp_rtcp->SetRtcpXrRrtrStatus(rtp_rtcp_->RtcpXrRrtrStatus());
|
|
}
|
|
// |RegisterSimulcastRtpRtcpModules| resets all old weak pointers and old
|
|
// modules can be deleted after this step.
|
|
vie_receiver_.RegisterSimulcastRtpRtcpModules(simulcast_rtp_rtcp_);
|
|
} else {
|
|
while (!simulcast_rtp_rtcp_.empty()) {
|
|
RtpRtcp* rtp_rtcp = simulcast_rtp_rtcp_.back();
|
|
module_process_thread_.DeRegisterModule(rtp_rtcp);
|
|
rtp_rtcp->SetSendingStatus(false);
|
|
rtp_rtcp->SetSendingMediaStatus(false);
|
|
simulcast_rtp_rtcp_.pop_back();
|
|
removed_rtp_rtcp_.push_front(rtp_rtcp);
|
|
}
|
|
// Clear any previous modules.
|
|
vie_receiver_.RegisterSimulcastRtpRtcpModules(simulcast_rtp_rtcp_);
|
|
}
|
|
// Enable this if H264 is available.
|
|
// This sets the wanted packetization mode.
|
|
// if (video_codec.plType == kVideoCodecH264) {
|
|
// if (video_codec.codecSpecific.H264.packetization == kH264SingleMode) {
|
|
// rtp_rtcp_->SetH264PacketizationMode(H264_SINGLE_NAL_MODE);
|
|
// } else {
|
|
// rtp_rtcp_->SetH264PacketizationMode(H264_NON_INTERLEAVED_MODE);
|
|
// }
|
|
// if (video_codec.codecSpecific.H264.configParametersSize > 0) {
|
|
// rtp_rtcp_->SetH264SendModeNALU_PPS_SPS(true);
|
|
// }
|
|
// }
|
|
|
|
// Don't log this error, no way to check in advance if this pl_type is
|
|
// registered or not...
|
|
rtp_rtcp_->DeRegisterSendPayload(video_codec.plType);
|
|
if (rtp_rtcp_->RegisterSendPayload(video_codec) != 0) {
|
|
WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
|
|
"%s: could not register payload type", __FUNCTION__);
|
|
return -1;
|
|
}
|
|
if (restart_rtp) {
|
|
rtp_rtcp_->SetSendingStatus(true);
|
|
for (std::list<RtpRtcp*>::iterator it = simulcast_rtp_rtcp_.begin();
|
|
it != simulcast_rtp_rtcp_.end(); ++it) {
|
|
(*it)->SetSendingStatus(true);
|
|
(*it)->SetSendingMediaStatus(true);
|
|
}
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
int32_t ViEChannel::SetReceiveCodec(const VideoCodec& video_codec) {
|
|
// We will not receive simulcast streams, so no need to handle that use case.
|
|
WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
|
|
"%s", __FUNCTION__);
|
|
|
|
if (!vie_receiver_.SetReceiveCodec(video_codec)) {
|
|
WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
|
|
"%s: Could not register receive payload type", __FUNCTION__);
|
|
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_,
|
|
wait_for_key_frame_) != VCM_OK) {
|
|
WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
|
|
"%s: Could not register decoder", __FUNCTION__);
|
|
return -1;
|
|
}
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
int32_t ViEChannel::GetReceiveCodec(VideoCodec* video_codec) {
|
|
WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
|
|
"%s", __FUNCTION__);
|
|
|
|
if (vcm_.ReceiveCodec(video_codec) != 0) {
|
|
WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
|
|
"%s: Could not get receive codec", __FUNCTION__);
|
|
return -1;
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
int32_t ViEChannel::RegisterCodecObserver(ViEDecoderObserver* observer) {
|
|
CriticalSectionScoped cs(callback_cs_.get());
|
|
if (observer) {
|
|
if (codec_observer_) {
|
|
WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
|
|
"%s: already added", __FUNCTION__);
|
|
return -1;
|
|
}
|
|
WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
|
|
"%s: observer added", __FUNCTION__);
|
|
codec_observer_ = observer;
|
|
} else {
|
|
if (!codec_observer_) {
|
|
WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
|
|
"%s: no observer added", __FUNCTION__);
|
|
return -1;
|
|
}
|
|
WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
|
|
"%s: observer removed", __FUNCTION__);
|
|
codec_observer_ = NULL;
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
int32_t ViEChannel::RegisterExternalDecoder(const uint8_t pl_type,
|
|
VideoDecoder* decoder,
|
|
bool buffered_rendering,
|
|
int32_t render_delay) {
|
|
WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
|
|
"%s", __FUNCTION__);
|
|
|
|
int32_t result;
|
|
result = vcm_.RegisterExternalDecoder(decoder, pl_type, buffered_rendering);
|
|
if (result != VCM_OK) {
|
|
WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
|
|
"%s: Could not register external decoder with VCM.",
|
|
__FUNCTION__);
|
|
return result;
|
|
}
|
|
return vcm_.SetRenderDelay(render_delay);
|
|
}
|
|
|
|
int32_t ViEChannel::DeRegisterExternalDecoder(
|
|
const uint8_t pl_type) {
|
|
WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
|
|
"%s pl_type", __FUNCTION__, pl_type);
|
|
|
|
VideoCodec current_receive_codec;
|
|
int32_t result = 0;
|
|
result = vcm_.ReceiveCodec(¤t_receive_codec);
|
|
if (vcm_.RegisterExternalDecoder(NULL, pl_type, false) != VCM_OK) {
|
|
return -1;
|
|
}
|
|
|
|
if (result == 0 && current_receive_codec.plType == pl_type) {
|
|
result = vcm_.RegisterReceiveCodec(¤t_receive_codec, number_of_cores_,
|
|
wait_for_key_frame_);
|
|
}
|
|
return result;
|
|
}
|
|
|
|
int32_t ViEChannel::ReceiveCodecStatistics(
|
|
uint32_t* num_key_frames, uint32_t* num_delta_frames) {
|
|
WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
|
|
"%s", __FUNCTION__);
|
|
|
|
VCMFrameCount received_frames;
|
|
if (vcm_.ReceivedFrameCount(received_frames) != VCM_OK) {
|
|
WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
|
|
"%s: Could not get received frame information", __FUNCTION__);
|
|
return -1;
|
|
}
|
|
*num_key_frames = received_frames.numKeyFrames;
|
|
*num_delta_frames = received_frames.numDeltaFrames;
|
|
return 0;
|
|
}
|
|
|
|
uint32_t ViEChannel::DiscardedPackets() const {
|
|
WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_), "%s",
|
|
__FUNCTION__);
|
|
return vcm_.DiscardedPackets();
|
|
}
|
|
|
|
int ViEChannel::ReceiveDelay() const {
|
|
return vcm_.Delay();
|
|
}
|
|
|
|
int32_t ViEChannel::WaitForKeyFrame(bool wait) {
|
|
WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
|
|
"%s(wait: %d)", __FUNCTION__, wait);
|
|
wait_for_key_frame_ = wait;
|
|
return 0;
|
|
}
|
|
|
|
int32_t ViEChannel::SetSignalPacketLossStatus(bool enable,
|
|
bool only_key_frames) {
|
|
WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
|
|
"%s(enable: %d)", __FUNCTION__, enable);
|
|
if (enable) {
|
|
if (only_key_frames) {
|
|
vcm_.SetVideoProtection(kProtectionKeyOnLoss, false);
|
|
if (vcm_.SetVideoProtection(kProtectionKeyOnKeyLoss, true) != VCM_OK) {
|
|
WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
|
|
"%s failed %d", __FUNCTION__, enable);
|
|
return -1;
|
|
}
|
|
} else {
|
|
vcm_.SetVideoProtection(kProtectionKeyOnKeyLoss, false);
|
|
if (vcm_.SetVideoProtection(kProtectionKeyOnLoss, true) != VCM_OK) {
|
|
WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
|
|
"%s failed %d", __FUNCTION__, enable);
|
|
return -1;
|
|
}
|
|
}
|
|
} else {
|
|
vcm_.SetVideoProtection(kProtectionKeyOnLoss, false);
|
|
vcm_.SetVideoProtection(kProtectionKeyOnKeyLoss, false);
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
int32_t ViEChannel::SetRTCPMode(const RTCPMethod rtcp_mode) {
|
|
WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
|
|
"%s: %d", __FUNCTION__, rtcp_mode);
|
|
|
|
CriticalSectionScoped cs(rtp_rtcp_cs_.get());
|
|
for (std::list<RtpRtcp*>::iterator it = simulcast_rtp_rtcp_.begin();
|
|
it != simulcast_rtp_rtcp_.end();
|
|
it++) {
|
|
RtpRtcp* rtp_rtcp = *it;
|
|
rtp_rtcp->SetRTCPStatus(rtcp_mode);
|
|
}
|
|
return rtp_rtcp_->SetRTCPStatus(rtcp_mode);
|
|
}
|
|
|
|
int32_t ViEChannel::GetRTCPMode(RTCPMethod* rtcp_mode) {
|
|
WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
|
|
"%s", __FUNCTION__);
|
|
*rtcp_mode = rtp_rtcp_->RTCP();
|
|
return 0;
|
|
}
|
|
|
|
int32_t ViEChannel::SetNACKStatus(const bool enable) {
|
|
WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
|
|
"%s(enable: %d)", __FUNCTION__, enable);
|
|
|
|
// Update the decoding VCM.
|
|
if (vcm_.SetVideoProtection(kProtectionNack, enable) != VCM_OK) {
|
|
WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
|
|
"%s: Could not set VCM NACK protection: %d", __FUNCTION__,
|
|
enable);
|
|
return -1;
|
|
}
|
|
if (enable) {
|
|
// Disable possible FEC.
|
|
SetFECStatus(false, 0, 0);
|
|
}
|
|
// Update the decoding VCM.
|
|
if (vcm_.SetVideoProtection(kProtectionNack, enable) != VCM_OK) {
|
|
WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
|
|
"%s: Could not set VCM NACK protection: %d", __FUNCTION__,
|
|
enable);
|
|
return -1;
|
|
}
|
|
return ProcessNACKRequest(enable);
|
|
}
|
|
|
|
int32_t ViEChannel::ProcessNACKRequest(const bool enable) {
|
|
WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
|
|
"%s(enable: %d)", __FUNCTION__, enable);
|
|
|
|
if (enable) {
|
|
// Turn on NACK.
|
|
NACKMethod nackMethod = kNackRtcp;
|
|
if (rtp_rtcp_->RTCP() == kRtcpOff) {
|
|
WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
|
|
"%s: Could not enable NACK, RTPC not on ", __FUNCTION__);
|
|
return -1;
|
|
}
|
|
vie_receiver_.SetNackStatus(true, max_nack_reordering_threshold_);
|
|
WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
|
|
"%s: Using NACK method %d", __FUNCTION__, nackMethod);
|
|
rtp_rtcp_->SetStorePacketsStatus(true, nack_history_size_sender_);
|
|
|
|
vcm_.RegisterPacketRequestCallback(this);
|
|
|
|
CriticalSectionScoped cs(rtp_rtcp_cs_.get());
|
|
|
|
for (std::list<RtpRtcp*>::iterator it = simulcast_rtp_rtcp_.begin();
|
|
it != simulcast_rtp_rtcp_.end();
|
|
it++) {
|
|
RtpRtcp* rtp_rtcp = *it;
|
|
rtp_rtcp->SetStorePacketsStatus(true, nack_history_size_sender_);
|
|
}
|
|
// Don't introduce errors when NACK is enabled.
|
|
vcm_.SetDecodeErrorMode(kNoErrors);
|
|
} else {
|
|
CriticalSectionScoped cs(rtp_rtcp_cs_.get());
|
|
for (std::list<RtpRtcp*>::iterator it = simulcast_rtp_rtcp_.begin();
|
|
it != simulcast_rtp_rtcp_.end();
|
|
it++) {
|
|
RtpRtcp* rtp_rtcp = *it;
|
|
if (paced_sender_ == NULL) {
|
|
rtp_rtcp->SetStorePacketsStatus(false, 0);
|
|
}
|
|
}
|
|
vcm_.RegisterPacketRequestCallback(NULL);
|
|
if (paced_sender_ == NULL) {
|
|
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);
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
int32_t ViEChannel::SetFECStatus(const bool enable,
|
|
const unsigned char payload_typeRED,
|
|
const unsigned char payload_typeFEC) {
|
|
// Disable possible NACK.
|
|
if (enable) {
|
|
SetNACKStatus(false);
|
|
}
|
|
|
|
return ProcessFECRequest(enable, payload_typeRED, payload_typeFEC);
|
|
}
|
|
|
|
int32_t ViEChannel::ProcessFECRequest(
|
|
const bool enable,
|
|
const unsigned char payload_typeRED,
|
|
const unsigned char payload_typeFEC) {
|
|
WEBRTC_TRACE(kTraceApiCall, kTraceVideo, ViEId(engine_id_, channel_id_),
|
|
"%s(enable: %d, payload_typeRED: %u, payload_typeFEC: %u)",
|
|
__FUNCTION__, enable, payload_typeRED, payload_typeFEC);
|
|
|
|
if (rtp_rtcp_->SetGenericFECStatus(enable, payload_typeRED,
|
|
payload_typeFEC) != 0) {
|
|
WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
|
|
"%s: Could not change FEC status to %d", __FUNCTION__,
|
|
enable);
|
|
return -1;
|
|
}
|
|
CriticalSectionScoped cs(rtp_rtcp_cs_.get());
|
|
for (std::list<RtpRtcp*>::iterator it = simulcast_rtp_rtcp_.begin();
|
|
it != simulcast_rtp_rtcp_.end();
|
|
it++) {
|
|
RtpRtcp* rtp_rtcp = *it;
|
|
rtp_rtcp->SetGenericFECStatus(enable, payload_typeRED, payload_typeFEC);
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
int32_t ViEChannel::SetHybridNACKFECStatus(
|
|
const bool enable,
|
|
const unsigned char payload_typeRED,
|
|
const unsigned char payload_typeFEC) {
|
|
// Update the decoding VCM with hybrid mode.
|
|
if (vcm_.SetVideoProtection(kProtectionNackFEC, enable) != VCM_OK) {
|
|
WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
|
|
"%s: Could not set VCM NACK protection: %d", __FUNCTION__,
|
|
enable);
|
|
return -1;
|
|
}
|
|
|
|
int32_t ret_val = 0;
|
|
ret_val = ProcessNACKRequest(enable);
|
|
if (ret_val < 0) {
|
|
return ret_val;
|
|
}
|
|
return ProcessFECRequest(enable, payload_typeRED, payload_typeFEC);
|
|
}
|
|
|
|
int ViEChannel::SetSenderBufferingMode(int target_delay_ms) {
|
|
if ((target_delay_ms < 0) || (target_delay_ms > kMaxTargetDelayMs)) {
|
|
WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
|
|
"%s: Target sender buffering delay out of bounds: %d",
|
|
__FUNCTION__, target_delay_ms);
|
|
return -1;
|
|
}
|
|
if (target_delay_ms == 0) {
|
|
// Real-time mode.
|
|
nack_history_size_sender_ = kSendSidePacketHistorySize;
|
|
} else {
|
|
nack_history_size_sender_ = GetRequiredNackListSize(target_delay_ms);
|
|
// Don't allow a number lower than the default value.
|
|
if (nack_history_size_sender_ < kSendSidePacketHistorySize) {
|
|
nack_history_size_sender_ = kSendSidePacketHistorySize;
|
|
}
|
|
}
|
|
// Setting nack_history_size_.
|
|
// First disabling (forcing free) and then resetting to desired value.
|
|
if (rtp_rtcp_->SetStorePacketsStatus(false, 0) != 0) {
|
|
WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
|
|
"%s:SetStorePacketsStatus failure", __FUNCTION__);
|
|
return -1;
|
|
}
|
|
if (rtp_rtcp_->SetStorePacketsStatus(true, nack_history_size_sender_) != 0) {
|
|
WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
|
|
"%s:SetStorePacketsStatus failure", __FUNCTION__);
|
|
return -1;
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
int ViEChannel::SetReceiverBufferingMode(int target_delay_ms) {
|
|
if ((target_delay_ms < 0) || (target_delay_ms > kMaxTargetDelayMs)) {
|
|
WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
|
|
"%s: Target receiver buffering delay out of bounds: %d",
|
|
__FUNCTION__, target_delay_ms);
|
|
return -1;
|
|
}
|
|
int max_nack_list_size;
|
|
int max_incomplete_time_ms;
|
|
if (target_delay_ms == 0) {
|
|
// Real-time mode - restore default settings.
|
|
max_nack_reordering_threshold_ = kMaxPacketAgeToNack;
|
|
max_nack_list_size = kMaxNackListSize;
|
|
max_incomplete_time_ms = 0;
|
|
} else {
|
|
max_nack_list_size = 3 * GetRequiredNackListSize(target_delay_ms) / 4;
|
|
max_nack_reordering_threshold_ = max_nack_list_size;
|
|
// Calculate the max incomplete time and round to int.
|
|
max_incomplete_time_ms = static_cast<int>(kMaxIncompleteTimeMultiplier *
|
|
target_delay_ms + 0.5f);
|
|
}
|
|
vcm_.SetNackSettings(max_nack_list_size, max_nack_reordering_threshold_,
|
|
max_incomplete_time_ms);
|
|
vcm_.SetMinReceiverDelay(target_delay_ms);
|
|
if (vie_sync_.SetTargetBufferingDelay(target_delay_ms) < 0)
|
|
return -1;
|
|
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;
|
|
}
|
|
|
|
int32_t ViEChannel::SetKeyFrameRequestMethod(
|
|
const KeyFrameRequestMethod method) {
|
|
WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
|
|
"%s: %d", __FUNCTION__, method);
|
|
return rtp_rtcp_->SetKeyFrameRequestMethod(method);
|
|
}
|
|
|
|
bool ViEChannel::EnableRemb(bool enable) {
|
|
WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
|
|
"ViEChannel::EnableRemb: %d", enable);
|
|
if (rtp_rtcp_->SetREMBStatus(enable) != 0)
|
|
return false;
|
|
return true;
|
|
}
|
|
|
|
int ViEChannel::SetSendTimestampOffsetStatus(bool enable, int id) {
|
|
CriticalSectionScoped cs(rtp_rtcp_cs_.get());
|
|
int error = 0;
|
|
if (enable) {
|
|
// Enable the extension, but disable possible old id to avoid errors.
|
|
send_timestamp_extension_id_ = id;
|
|
rtp_rtcp_->DeregisterSendRtpHeaderExtension(
|
|
kRtpExtensionTransmissionTimeOffset);
|
|
error = rtp_rtcp_->RegisterSendRtpHeaderExtension(
|
|
kRtpExtensionTransmissionTimeOffset, id);
|
|
for (std::list<RtpRtcp*>::iterator it = simulcast_rtp_rtcp_.begin();
|
|
it != simulcast_rtp_rtcp_.end(); it++) {
|
|
(*it)->DeregisterSendRtpHeaderExtension(
|
|
kRtpExtensionTransmissionTimeOffset);
|
|
error |= (*it)->RegisterSendRtpHeaderExtension(
|
|
kRtpExtensionTransmissionTimeOffset, id);
|
|
}
|
|
} else {
|
|
// Disable the extension.
|
|
send_timestamp_extension_id_ = kInvalidRtpExtensionId;
|
|
rtp_rtcp_->DeregisterSendRtpHeaderExtension(
|
|
kRtpExtensionTransmissionTimeOffset);
|
|
for (std::list<RtpRtcp*>::iterator it = simulcast_rtp_rtcp_.begin();
|
|
it != simulcast_rtp_rtcp_.end(); it++) {
|
|
(*it)->DeregisterSendRtpHeaderExtension(
|
|
kRtpExtensionTransmissionTimeOffset);
|
|
}
|
|
}
|
|
return error;
|
|
}
|
|
|
|
int ViEChannel::SetReceiveTimestampOffsetStatus(bool enable, int id) {
|
|
return vie_receiver_.SetReceiveTimestampOffsetStatus(enable, id) ? 0 : -1;
|
|
}
|
|
|
|
int ViEChannel::SetSendAbsoluteSendTimeStatus(bool enable, int id) {
|
|
CriticalSectionScoped cs(rtp_rtcp_cs_.get());
|
|
int error = 0;
|
|
if (enable) {
|
|
// Enable the extension, but disable possible old id to avoid errors.
|
|
absolute_send_time_extension_id_ = id;
|
|
rtp_rtcp_->DeregisterSendRtpHeaderExtension(
|
|
kRtpExtensionAbsoluteSendTime);
|
|
error = rtp_rtcp_->RegisterSendRtpHeaderExtension(
|
|
kRtpExtensionAbsoluteSendTime, id);
|
|
for (std::list<RtpRtcp*>::iterator it = simulcast_rtp_rtcp_.begin();
|
|
it != simulcast_rtp_rtcp_.end(); it++) {
|
|
(*it)->DeregisterSendRtpHeaderExtension(
|
|
kRtpExtensionAbsoluteSendTime);
|
|
error |= (*it)->RegisterSendRtpHeaderExtension(
|
|
kRtpExtensionAbsoluteSendTime, id);
|
|
}
|
|
} else {
|
|
// Disable the extension.
|
|
absolute_send_time_extension_id_ = kInvalidRtpExtensionId;
|
|
rtp_rtcp_->DeregisterSendRtpHeaderExtension(
|
|
kRtpExtensionAbsoluteSendTime);
|
|
for (std::list<RtpRtcp*>::iterator it = simulcast_rtp_rtcp_.begin();
|
|
it != simulcast_rtp_rtcp_.end(); it++) {
|
|
(*it)->DeregisterSendRtpHeaderExtension(
|
|
kRtpExtensionAbsoluteSendTime);
|
|
}
|
|
}
|
|
return error;
|
|
}
|
|
|
|
int ViEChannel::SetReceiveAbsoluteSendTimeStatus(bool enable, int id) {
|
|
receive_absolute_send_time_enabled_ = enable;
|
|
return vie_receiver_.SetReceiveAbsoluteSendTimeStatus(enable, id) ? 0 : -1;
|
|
}
|
|
|
|
bool ViEChannel::GetReceiveAbsoluteSendTimeStatus() const {
|
|
return receive_absolute_send_time_enabled_;
|
|
}
|
|
|
|
void ViEChannel::SetRtcpXrRrtrStatus(bool enable) {
|
|
CriticalSectionScoped cs(rtp_rtcp_cs_.get());
|
|
rtp_rtcp_->SetRtcpXrRrtrStatus(enable);
|
|
for (std::list<RtpRtcp*>::iterator it = simulcast_rtp_rtcp_.begin();
|
|
it != simulcast_rtp_rtcp_.end(); it++) {
|
|
(*it)->SetRtcpXrRrtrStatus(enable);
|
|
}
|
|
}
|
|
|
|
void ViEChannel::SetTransmissionSmoothingStatus(bool enable) {
|
|
assert(paced_sender_ && "No paced sender registered.");
|
|
paced_sender_->SetStatus(enable);
|
|
}
|
|
|
|
int32_t ViEChannel::EnableTMMBR(const bool enable) {
|
|
WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
|
|
"%s: %d", __FUNCTION__, enable);
|
|
return rtp_rtcp_->SetTMMBRStatus(enable);
|
|
}
|
|
|
|
int32_t ViEChannel::EnableKeyFrameRequestCallback(const bool enable) {
|
|
WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
|
|
"%s: %d", __FUNCTION__, enable);
|
|
|
|
CriticalSectionScoped cs(callback_cs_.get());
|
|
if (enable && !codec_observer_) {
|
|
WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
|
|
"%s: No ViECodecObserver set", __FUNCTION__, enable);
|
|
return -1;
|
|
}
|
|
do_key_frame_callbackRequest_ = enable;
|
|
return 0;
|
|
}
|
|
|
|
int32_t ViEChannel::SetSSRC(const uint32_t SSRC,
|
|
const StreamType usage,
|
|
const uint8_t simulcast_idx) {
|
|
WEBRTC_TRACE(webrtc::kTraceInfo,
|
|
webrtc::kTraceVideo,
|
|
ViEId(engine_id_, channel_id_),
|
|
"%s(usage:%d, SSRC: 0x%x, idx:%u)",
|
|
__FUNCTION__, usage, SSRC, simulcast_idx);
|
|
int rtx_settings = kRtxRetransmitted;
|
|
if (config_.Get<PaddingStrategy>().redundant_payloads)
|
|
rtx_settings |= kRtxRedundantPayloads;
|
|
if (simulcast_idx == 0) {
|
|
if (usage == kViEStreamTypeRtx) {
|
|
return rtp_rtcp_->SetRTXSendStatus(rtx_settings, true, SSRC);
|
|
}
|
|
return rtp_rtcp_->SetSSRC(SSRC);
|
|
}
|
|
CriticalSectionScoped cs(rtp_rtcp_cs_.get());
|
|
if (simulcast_idx > simulcast_rtp_rtcp_.size()) {
|
|
return -1;
|
|
}
|
|
std::list<RtpRtcp*>::const_iterator it = simulcast_rtp_rtcp_.begin();
|
|
for (int i = 1; i < simulcast_idx; ++i, ++it) {
|
|
if (it == simulcast_rtp_rtcp_.end()) {
|
|
return -1;
|
|
}
|
|
}
|
|
RtpRtcp* rtp_rtcp_module = *it;
|
|
if (usage == kViEStreamTypeRtx) {
|
|
return rtp_rtcp_module->SetRTXSendStatus(rtx_settings, true, SSRC);
|
|
}
|
|
return rtp_rtcp_module->SetSSRC(SSRC);
|
|
}
|
|
|
|
int32_t ViEChannel::SetRemoteSSRCType(const StreamType usage,
|
|
const uint32_t SSRC) {
|
|
WEBRTC_TRACE(webrtc::kTraceInfo,
|
|
webrtc::kTraceVideo,
|
|
ViEId(engine_id_, channel_id_),
|
|
"%s(usage:%d, SSRC: 0x%x)",
|
|
__FUNCTION__, usage, SSRC);
|
|
|
|
vie_receiver_.SetRtxStatus(true, SSRC);
|
|
return 0;
|
|
}
|
|
|
|
// TODO(mflodman) Add kViEStreamTypeRtx.
|
|
int32_t ViEChannel::GetLocalSSRC(uint8_t idx, unsigned int* ssrc) {
|
|
WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
|
|
"%s", __FUNCTION__);
|
|
|
|
if (idx == 0) {
|
|
*ssrc = rtp_rtcp_->SSRC();
|
|
return 0;
|
|
}
|
|
CriticalSectionScoped cs(rtp_rtcp_cs_.get());
|
|
if (idx > simulcast_rtp_rtcp_.size()) {
|
|
return -1;
|
|
}
|
|
std::list<RtpRtcp*>::const_iterator it = simulcast_rtp_rtcp_.begin();
|
|
for (int i = 1; i < idx; ++i, ++it) {
|
|
if (it == simulcast_rtp_rtcp_.end()) {
|
|
return -1;
|
|
}
|
|
}
|
|
*ssrc = (*it)->SSRC();
|
|
return 0;
|
|
}
|
|
|
|
int32_t ViEChannel::GetRemoteSSRC(uint32_t* ssrc) {
|
|
WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_), "%s",
|
|
__FUNCTION__);
|
|
|
|
*ssrc = vie_receiver_.GetRemoteSsrc();
|
|
return 0;
|
|
}
|
|
|
|
int32_t ViEChannel::GetRemoteCSRC(uint32_t CSRCs[kRtpCsrcSize]) {
|
|
WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_), "%s",
|
|
__FUNCTION__);
|
|
|
|
uint32_t arrayCSRC[kRtpCsrcSize];
|
|
memset(arrayCSRC, 0, sizeof(arrayCSRC));
|
|
|
|
int num_csrcs = vie_receiver_.GetCsrcs(arrayCSRC);
|
|
if (num_csrcs > 0) {
|
|
memcpy(CSRCs, arrayCSRC, num_csrcs * sizeof(uint32_t));
|
|
for (int idx = 0; idx < num_csrcs; idx++) {
|
|
WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
|
|
"\tCSRC[%d] = %lu", idx, CSRCs[idx]);
|
|
}
|
|
} else {
|
|
WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
|
|
"%s: CSRC list is empty", __FUNCTION__);
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
int ViEChannel::SetRtxSendPayloadType(int payload_type) {
|
|
if (rtp_rtcp_->Sending()) {
|
|
WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
|
|
"%s: already sending", __FUNCTION__);
|
|
return -1;
|
|
}
|
|
rtp_rtcp_->SetRtxSendPayloadType(payload_type);
|
|
CriticalSectionScoped cs(rtp_rtcp_cs_.get());
|
|
for (std::list<RtpRtcp*>::iterator it = simulcast_rtp_rtcp_.begin();
|
|
it != simulcast_rtp_rtcp_.end(); it++) {
|
|
(*it)->SetRtxSendPayloadType(payload_type);
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
void ViEChannel::SetRtxReceivePayloadType(int payload_type) {
|
|
vie_receiver_.SetRtxPayloadType(payload_type);
|
|
}
|
|
|
|
int32_t ViEChannel::SetStartSequenceNumber(uint16_t sequence_number) {
|
|
WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_), "%s",
|
|
__FUNCTION__);
|
|
|
|
if (rtp_rtcp_->Sending()) {
|
|
WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
|
|
"%s: already sending", __FUNCTION__);
|
|
return -1;
|
|
}
|
|
return rtp_rtcp_->SetSequenceNumber(sequence_number);
|
|
}
|
|
|
|
int32_t ViEChannel::SetRTCPCName(const char rtcp_cname[]) {
|
|
WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
|
|
"%s", __FUNCTION__);
|
|
if (rtp_rtcp_->Sending()) {
|
|
WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
|
|
"%s: already sending", __FUNCTION__);
|
|
return -1;
|
|
}
|
|
return rtp_rtcp_->SetCNAME(rtcp_cname);
|
|
}
|
|
|
|
int32_t ViEChannel::GetRTCPCName(char rtcp_cname[]) {
|
|
WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
|
|
"%s", __FUNCTION__);
|
|
return rtp_rtcp_->CNAME(rtcp_cname);
|
|
}
|
|
|
|
int32_t ViEChannel::GetRemoteRTCPCName(char rtcp_cname[]) {
|
|
WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_), "%s",
|
|
__FUNCTION__);
|
|
|
|
uint32_t remoteSSRC = vie_receiver_.GetRemoteSsrc();
|
|
return rtp_rtcp_->RemoteCNAME(remoteSSRC, rtcp_cname);
|
|
}
|
|
|
|
int32_t ViEChannel::RegisterRtpObserver(ViERTPObserver* observer) {
|
|
CriticalSectionScoped cs(callback_cs_.get());
|
|
if (observer) {
|
|
if (rtp_observer_) {
|
|
WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
|
|
"%s: observer alread added", __FUNCTION__);
|
|
return -1;
|
|
}
|
|
WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
|
|
"%s: observer added", __FUNCTION__);
|
|
rtp_observer_ = observer;
|
|
} else {
|
|
if (!rtp_observer_) {
|
|
WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
|
|
"%s: no observer added", __FUNCTION__);
|
|
return -1;
|
|
}
|
|
WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
|
|
"%s: observer removed", __FUNCTION__);
|
|
rtp_observer_ = NULL;
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
int32_t ViEChannel::RegisterRtcpObserver(ViERTCPObserver* observer) {
|
|
CriticalSectionScoped cs(callback_cs_.get());
|
|
if (observer) {
|
|
if (rtcp_observer_) {
|
|
WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
|
|
"%s: observer alread added", __FUNCTION__);
|
|
return -1;
|
|
}
|
|
WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
|
|
"%s: observer added", __FUNCTION__);
|
|
rtcp_observer_ = observer;
|
|
} else {
|
|
if (!rtcp_observer_) {
|
|
WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
|
|
"%s: no observer added", __FUNCTION__);
|
|
return -1;
|
|
}
|
|
WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
|
|
"%s: observer removed", __FUNCTION__);
|
|
rtcp_observer_ = NULL;
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
int32_t ViEChannel::SendApplicationDefinedRTCPPacket(
|
|
const uint8_t sub_type,
|
|
uint32_t name,
|
|
const uint8_t* data,
|
|
uint16_t data_length_in_bytes) {
|
|
WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_), "%s",
|
|
__FUNCTION__);
|
|
if (!rtp_rtcp_->Sending()) {
|
|
WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
|
|
"%s: not sending", __FUNCTION__);
|
|
return -1;
|
|
}
|
|
if (!data) {
|
|
WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
|
|
"%s: no input argument", __FUNCTION__);
|
|
return -1;
|
|
}
|
|
if (data_length_in_bytes % 4 != 0) {
|
|
WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
|
|
"%s: input length error", __FUNCTION__);
|
|
return -1;
|
|
}
|
|
RTCPMethod rtcp_method = rtp_rtcp_->RTCP();
|
|
if (rtcp_method == kRtcpOff) {
|
|
WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
|
|
"%s: RTCP not enabled", __FUNCTION__);
|
|
return -1;
|
|
}
|
|
// Create and send packet.
|
|
if (rtp_rtcp_->SetRTCPApplicationSpecificData(sub_type, name, data,
|
|
data_length_in_bytes) != 0) {
|
|
WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
|
|
"%s: Could not send RTCP application data", __FUNCTION__);
|
|
return -1;
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
int32_t ViEChannel::GetSendRtcpStatistics(uint16_t* fraction_lost,
|
|
uint32_t* cumulative_lost,
|
|
uint32_t* extended_max,
|
|
uint32_t* jitter_samples,
|
|
int32_t* rtt_ms) {
|
|
WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_), "%s",
|
|
__FUNCTION__);
|
|
|
|
// TODO(pwestin) how do we do this for simulcast ? average for all
|
|
// except cumulative_lost that is the sum ?
|
|
// CriticalSectionScoped cs(rtp_rtcp_cs_.get());
|
|
|
|
// for (std::list<RtpRtcp*>::const_iterator it = simulcast_rtp_rtcp_.begin();
|
|
// it != simulcast_rtp_rtcp_.end();
|
|
// it++) {
|
|
// RtpRtcp* rtp_rtcp = *it;
|
|
// }
|
|
uint32_t remote_ssrc = vie_receiver_.GetRemoteSsrc();
|
|
|
|
// Get all RTCP receiver report blocks that have been received on this
|
|
// channel. If we receive RTP packets from a remote source we know the
|
|
// remote SSRC and use the report block from him.
|
|
// Otherwise use the first report block.
|
|
std::vector<RTCPReportBlock> remote_stats;
|
|
if (rtp_rtcp_->RemoteRTCPStat(&remote_stats) != 0 || remote_stats.empty()) {
|
|
WEBRTC_TRACE(kTraceWarning, kTraceVideo, ViEId(engine_id_, channel_id_),
|
|
"%s: Could not get remote stats", __FUNCTION__);
|
|
return -1;
|
|
}
|
|
std::vector<RTCPReportBlock>::const_iterator statistics =
|
|
remote_stats.begin();
|
|
for (; statistics != remote_stats.end(); ++statistics) {
|
|
if (statistics->remoteSSRC == remote_ssrc)
|
|
break;
|
|
}
|
|
|
|
if (statistics == remote_stats.end()) {
|
|
// If we have not received any RTCP packets from this SSRC it probably means
|
|
// we have not received any RTP packets.
|
|
// Use the first received report block instead.
|
|
statistics = remote_stats.begin();
|
|
remote_ssrc = statistics->remoteSSRC;
|
|
}
|
|
|
|
*fraction_lost = statistics->fractionLost;
|
|
*cumulative_lost = statistics->cumulativeLost;
|
|
*extended_max = statistics->extendedHighSeqNum;
|
|
*jitter_samples = statistics->jitter;
|
|
|
|
uint16_t dummy;
|
|
uint16_t rtt = 0;
|
|
if (rtp_rtcp_->RTT(remote_ssrc, &rtt, &dummy, &dummy, &dummy) != 0) {
|
|
WEBRTC_TRACE(kTraceWarning, kTraceVideo, ViEId(engine_id_, channel_id_),
|
|
"%s: Could not get RTT", __FUNCTION__);
|
|
return -1;
|
|
}
|
|
*rtt_ms = rtt;
|
|
return 0;
|
|
}
|
|
|
|
// TODO(holmer): This is a bad function name as it implies that it returns the
|
|
// received RTCP, while it actually returns the statistics which will be sent
|
|
// in the RTCP.
|
|
int32_t ViEChannel::GetReceivedRtcpStatistics(uint16_t* fraction_lost,
|
|
uint32_t* cumulative_lost,
|
|
uint32_t* extended_max,
|
|
uint32_t* jitter_samples,
|
|
int32_t* rtt_ms) {
|
|
WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
|
|
"%s", __FUNCTION__);
|
|
|
|
uint32_t remote_ssrc = vie_receiver_.GetRemoteSsrc();
|
|
uint8_t frac_lost = 0;
|
|
StreamStatistician* statistician =
|
|
vie_receiver_.GetReceiveStatistics()->GetStatistician(remote_ssrc);
|
|
StreamStatistician::Statistics receive_stats;
|
|
if (!statistician || !statistician->GetStatistics(
|
|
&receive_stats, rtp_rtcp_->RTCP() == kRtcpOff)) {
|
|
WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
|
|
"%s: Could not get received RTP statistics", __FUNCTION__);
|
|
return -1;
|
|
}
|
|
*fraction_lost = receive_stats.fraction_lost;
|
|
*cumulative_lost = receive_stats.cumulative_lost;
|
|
*extended_max = receive_stats.extended_max_sequence_number;
|
|
*jitter_samples = receive_stats.jitter;
|
|
*fraction_lost = frac_lost;
|
|
|
|
uint16_t dummy = 0;
|
|
uint16_t rtt = 0;
|
|
if (rtp_rtcp_->RTT(remote_ssrc, &rtt, &dummy, &dummy, &dummy) != 0) {
|
|
WEBRTC_TRACE(kTraceStateInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
|
|
"%s: Could not get RTT", __FUNCTION__);
|
|
}
|
|
*rtt_ms = rtt;
|
|
return 0;
|
|
}
|
|
|
|
int32_t ViEChannel::GetRtpStatistics(uint32_t* bytes_sent,
|
|
uint32_t* packets_sent,
|
|
uint32_t* bytes_received,
|
|
uint32_t* packets_received) const {
|
|
WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_), "%s",
|
|
__FUNCTION__);
|
|
|
|
StreamStatistician* statistician = vie_receiver_.GetReceiveStatistics()->
|
|
GetStatistician(vie_receiver_.GetRemoteSsrc());
|
|
*bytes_received = 0;
|
|
*packets_received = 0;
|
|
if (statistician)
|
|
statistician->GetDataCounters(bytes_received, packets_received);
|
|
if (rtp_rtcp_->DataCountersRTP(bytes_sent, packets_sent) != 0) {
|
|
WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
|
|
"%s: Could not get counters", __FUNCTION__);
|
|
return -1;
|
|
}
|
|
CriticalSectionScoped cs(rtp_rtcp_cs_.get());
|
|
for (std::list<RtpRtcp*>::const_iterator it = simulcast_rtp_rtcp_.begin();
|
|
it != simulcast_rtp_rtcp_.end();
|
|
it++) {
|
|
uint32_t bytes_sent_temp = 0;
|
|
uint32_t packets_sent_temp = 0;
|
|
RtpRtcp* rtp_rtcp = *it;
|
|
rtp_rtcp->DataCountersRTP(&bytes_sent_temp, &packets_sent_temp);
|
|
bytes_sent += bytes_sent_temp;
|
|
packets_sent += packets_sent_temp;
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
void ViEChannel::GetBandwidthUsage(uint32_t* total_bitrate_sent,
|
|
uint32_t* video_bitrate_sent,
|
|
uint32_t* fec_bitrate_sent,
|
|
uint32_t* nackBitrateSent) const {
|
|
WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_), "%s",
|
|
__FUNCTION__);
|
|
|
|
rtp_rtcp_->BitrateSent(total_bitrate_sent, video_bitrate_sent,
|
|
fec_bitrate_sent, nackBitrateSent);
|
|
CriticalSectionScoped cs(rtp_rtcp_cs_.get());
|
|
for (std::list<RtpRtcp*>::const_iterator it = simulcast_rtp_rtcp_.begin();
|
|
it != simulcast_rtp_rtcp_.end(); it++) {
|
|
uint32_t stream_rate = 0;
|
|
uint32_t video_rate = 0;
|
|
uint32_t fec_rate = 0;
|
|
uint32_t nackRate = 0;
|
|
RtpRtcp* rtp_rtcp = *it;
|
|
rtp_rtcp->BitrateSent(&stream_rate, &video_rate, &fec_rate, &nackRate);
|
|
*total_bitrate_sent += stream_rate;
|
|
*video_bitrate_sent += video_rate;
|
|
*fec_bitrate_sent += fec_rate;
|
|
*nackBitrateSent += nackRate;
|
|
}
|
|
}
|
|
|
|
void ViEChannel::GetEstimatedReceiveBandwidth(
|
|
uint32_t* estimated_bandwidth) const {
|
|
vie_receiver_.EstimatedReceiveBandwidth(estimated_bandwidth);
|
|
}
|
|
|
|
int32_t ViEChannel::StartRTPDump(const char file_nameUTF8[1024],
|
|
RTPDirections direction) {
|
|
WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_), "%s",
|
|
__FUNCTION__);
|
|
|
|
if (direction != kRtpIncoming && direction != kRtpOutgoing) {
|
|
WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
|
|
"%s: invalid input", __FUNCTION__);
|
|
return -1;
|
|
}
|
|
|
|
if (direction == kRtpIncoming) {
|
|
return vie_receiver_.StartRTPDump(file_nameUTF8);
|
|
} else {
|
|
return vie_sender_.StartRTPDump(file_nameUTF8);
|
|
}
|
|
}
|
|
|
|
int32_t ViEChannel::StopRTPDump(RTPDirections direction) {
|
|
WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
|
|
"%s", __FUNCTION__);
|
|
|
|
if (direction != kRtpIncoming && direction != kRtpOutgoing) {
|
|
WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
|
|
"%s: invalid input", __FUNCTION__);
|
|
return -1;
|
|
}
|
|
|
|
if (direction == kRtpIncoming) {
|
|
return vie_receiver_.StopRTPDump();
|
|
} else {
|
|
return vie_sender_.StopRTPDump();
|
|
}
|
|
}
|
|
|
|
int32_t ViEChannel::StartSend() {
|
|
CriticalSectionScoped cs(callback_cs_.get());
|
|
WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
|
|
"%s", __FUNCTION__);
|
|
|
|
if (!external_transport_) {
|
|
WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
|
|
"%s: send sockets not initialized", __FUNCTION__);
|
|
return -1;
|
|
}
|
|
rtp_rtcp_->SetSendingMediaStatus(true);
|
|
|
|
if (rtp_rtcp_->Sending()) {
|
|
// Already sending.
|
|
WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
|
|
"%s: Already sending", __FUNCTION__);
|
|
return kViEBaseAlreadySending;
|
|
}
|
|
if (rtp_rtcp_->SetSendingStatus(true) != 0) {
|
|
WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
|
|
"%s: Could not start sending RTP", __FUNCTION__);
|
|
return -1;
|
|
}
|
|
CriticalSectionScoped cs_rtp(rtp_rtcp_cs_.get());
|
|
for (std::list<RtpRtcp*>::const_iterator it = simulcast_rtp_rtcp_.begin();
|
|
it != simulcast_rtp_rtcp_.end();
|
|
it++) {
|
|
RtpRtcp* rtp_rtcp = *it;
|
|
rtp_rtcp->SetSendingMediaStatus(true);
|
|
rtp_rtcp->SetSendingStatus(true);
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
int32_t ViEChannel::StopSend() {
|
|
WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_), "%s",
|
|
__FUNCTION__);
|
|
|
|
CriticalSectionScoped cs(rtp_rtcp_cs_.get());
|
|
rtp_rtcp_->SetSendingMediaStatus(false);
|
|
for (std::list<RtpRtcp*>::iterator it = simulcast_rtp_rtcp_.begin();
|
|
it != simulcast_rtp_rtcp_.end();
|
|
it++) {
|
|
RtpRtcp* rtp_rtcp = *it;
|
|
rtp_rtcp->SetSendingMediaStatus(false);
|
|
}
|
|
if (!rtp_rtcp_->Sending()) {
|
|
WEBRTC_TRACE(kTraceWarning, kTraceVideo, ViEId(engine_id_, channel_id_),
|
|
"%s: Not sending", __FUNCTION__);
|
|
return kViEBaseNotSending;
|
|
}
|
|
|
|
// Reset.
|
|
rtp_rtcp_->ResetSendDataCountersRTP();
|
|
if (rtp_rtcp_->SetSendingStatus(false) != 0) {
|
|
WEBRTC_TRACE(kTraceWarning, kTraceVideo, ViEId(engine_id_, channel_id_),
|
|
"%s: could not stop RTP sending", __FUNCTION__);
|
|
return -1;
|
|
}
|
|
for (std::list<RtpRtcp*>::iterator it = simulcast_rtp_rtcp_.begin();
|
|
it != simulcast_rtp_rtcp_.end();
|
|
it++) {
|
|
RtpRtcp* rtp_rtcp = *it;
|
|
rtp_rtcp->ResetSendDataCountersRTP();
|
|
rtp_rtcp->SetSendingStatus(false);
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
bool ViEChannel::Sending() {
|
|
return rtp_rtcp_->Sending();
|
|
}
|
|
|
|
int32_t ViEChannel::StartReceive() {
|
|
CriticalSectionScoped cs(callback_cs_.get());
|
|
WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_), "%s",
|
|
__FUNCTION__);
|
|
|
|
if (StartDecodeThread() != 0) {
|
|
WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
|
|
"%s: could not start decoder thread", __FUNCTION__);
|
|
|
|
vie_receiver_.StopReceive();
|
|
return -1;
|
|
}
|
|
vie_receiver_.StartReceive();
|
|
return 0;
|
|
}
|
|
|
|
int32_t ViEChannel::StopReceive() {
|
|
WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_), "%s",
|
|
__FUNCTION__);
|
|
|
|
vie_receiver_.StopReceive();
|
|
StopDecodeThread();
|
|
vcm_.ResetDecoder();
|
|
return 0;
|
|
}
|
|
|
|
int32_t ViEChannel::RegisterSendTransport(Transport* transport) {
|
|
WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_), "%s",
|
|
__FUNCTION__);
|
|
|
|
if (rtp_rtcp_->Sending()) {
|
|
WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
|
|
"%s: Sending", __FUNCTION__);
|
|
return -1;
|
|
}
|
|
|
|
CriticalSectionScoped cs(callback_cs_.get());
|
|
if (external_transport_) {
|
|
WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
|
|
"%s: transport already registered", __FUNCTION__);
|
|
return -1;
|
|
}
|
|
external_transport_ = transport;
|
|
vie_sender_.RegisterSendTransport(transport);
|
|
WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
|
|
"%s: Transport registered: 0x%p", __FUNCTION__,
|
|
&external_transport_);
|
|
|
|
return 0;
|
|
}
|
|
|
|
int32_t ViEChannel::DeregisterSendTransport() {
|
|
CriticalSectionScoped cs(callback_cs_.get());
|
|
WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_), "%s",
|
|
__FUNCTION__);
|
|
|
|
if (!external_transport_) {
|
|
WEBRTC_TRACE(kTraceWarning, kTraceVideo, ViEId(engine_id_, channel_id_),
|
|
"%s: no transport registered", __FUNCTION__);
|
|
return -1;
|
|
}
|
|
if (rtp_rtcp_->Sending()) {
|
|
WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
|
|
"%s: Sending", __FUNCTION__);
|
|
return -1;
|
|
}
|
|
external_transport_ = NULL;
|
|
vie_sender_.DeregisterSendTransport();
|
|
return 0;
|
|
}
|
|
|
|
int32_t ViEChannel::ReceivedRTPPacket(
|
|
const void* rtp_packet, const int32_t rtp_packet_length) {
|
|
{
|
|
CriticalSectionScoped cs(callback_cs_.get());
|
|
if (!external_transport_) {
|
|
return -1;
|
|
}
|
|
}
|
|
return vie_receiver_.ReceivedRTPPacket(rtp_packet, rtp_packet_length);
|
|
}
|
|
|
|
int32_t ViEChannel::ReceivedRTCPPacket(
|
|
const void* rtcp_packet, const int32_t rtcp_packet_length) {
|
|
{
|
|
CriticalSectionScoped cs(callback_cs_.get());
|
|
if (!external_transport_) {
|
|
return -1;
|
|
}
|
|
}
|
|
return vie_receiver_.ReceivedRTCPPacket(rtcp_packet, rtcp_packet_length);
|
|
}
|
|
|
|
int32_t ViEChannel::SetMTU(uint16_t mtu) {
|
|
WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_), "%s",
|
|
__FUNCTION__);
|
|
if (rtp_rtcp_->SetMaxTransferUnit(mtu) != 0) {
|
|
// Logging done.
|
|
return -1;
|
|
}
|
|
CriticalSectionScoped cs(rtp_rtcp_cs_.get());
|
|
for (std::list<RtpRtcp*>::iterator it = simulcast_rtp_rtcp_.begin();
|
|
it != simulcast_rtp_rtcp_.end();
|
|
it++) {
|
|
RtpRtcp* rtp_rtcp = *it;
|
|
rtp_rtcp->SetMaxTransferUnit(mtu);
|
|
}
|
|
mtu_ = mtu;
|
|
return 0;
|
|
}
|
|
|
|
uint16_t ViEChannel::MaxDataPayloadLength() const {
|
|
WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
|
|
"%s", __FUNCTION__);
|
|
return rtp_rtcp_->MaxDataPayloadLength();
|
|
}
|
|
|
|
int32_t ViEChannel::EnableColorEnhancement(bool enable) {
|
|
WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
|
|
"%s(enable: %d)", __FUNCTION__, enable);
|
|
|
|
CriticalSectionScoped cs(callback_cs_.get());
|
|
color_enhancement_ = enable;
|
|
return 0;
|
|
}
|
|
|
|
RtpRtcp* ViEChannel::rtp_rtcp() {
|
|
return rtp_rtcp_.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(
|
|
I420VideoFrame& video_frame) { // NOLINT
|
|
CriticalSectionScoped cs(callback_cs_.get());
|
|
|
|
if (decoder_reset_) {
|
|
// Trigger a callback to the user if the incoming codec has changed.
|
|
if (codec_observer_) {
|
|
// The codec set by RegisterReceiveCodec might not be the size we're
|
|
// actually decoding.
|
|
receive_codec_.width = static_cast<uint16_t>(video_frame.width());
|
|
receive_codec_.height = static_cast<uint16_t>(video_frame.height());
|
|
codec_observer_->IncomingCodecChanged(channel_id_, receive_codec_);
|
|
}
|
|
decoder_reset_ = false;
|
|
}
|
|
// Post processing is not supported if the frame is backed by a texture.
|
|
if (video_frame.native_handle() == NULL) {
|
|
if (pre_render_callback_ != NULL)
|
|
pre_render_callback_->FrameCallback(&video_frame);
|
|
if (effect_filter_) {
|
|
unsigned int length = CalcBufferSize(kI420,
|
|
video_frame.width(),
|
|
video_frame.height());
|
|
scoped_array<uint8_t> video_buffer(new uint8_t[length]);
|
|
ExtractBuffer(video_frame, length, video_buffer.get());
|
|
effect_filter_->Transform(length, video_buffer.get(),
|
|
video_frame.timestamp(), video_frame.width(),
|
|
video_frame.height());
|
|
}
|
|
if (color_enhancement_) {
|
|
VideoProcessingModule::ColorEnhancement(&video_frame);
|
|
}
|
|
}
|
|
|
|
uint32_t arr_ofCSRC[kRtpCsrcSize];
|
|
int32_t no_of_csrcs = vie_receiver_.GetCsrcs(arr_ofCSRC);
|
|
if (no_of_csrcs <= 0) {
|
|
arr_ofCSRC[0] = vie_receiver_.GetRemoteSsrc();
|
|
no_of_csrcs = 1;
|
|
}
|
|
WEBRTC_TRACE(kTraceStream, kTraceVideo, ViEId(engine_id_, channel_id_),
|
|
"%s(timestamp:%u)", __FUNCTION__, video_frame.timestamp());
|
|
DeliverFrame(&video_frame, no_of_csrcs, arr_ofCSRC);
|
|
return 0;
|
|
}
|
|
|
|
int32_t ViEChannel::ReceivedDecodedReferenceFrame(
|
|
const uint64_t picture_id) {
|
|
return rtp_rtcp_->SendRTCPReferencePictureSelection(picture_id);
|
|
}
|
|
|
|
void ViEChannel::IncomingCodecChanged(const VideoCodec& codec) {
|
|
CriticalSectionScoped cs(callback_cs_.get());
|
|
receive_codec_ = codec;
|
|
}
|
|
|
|
int32_t ViEChannel::OnReceiveStatisticsUpdate(const uint32_t bit_rate,
|
|
const uint32_t frame_rate) {
|
|
CriticalSectionScoped cs(callback_cs_.get());
|
|
if (codec_observer_) {
|
|
WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
|
|
"%s: bitrate %u, framerate %u", __FUNCTION__, bit_rate,
|
|
frame_rate);
|
|
codec_observer_->IncomingRate(channel_id_, frame_rate, bit_rate);
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
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(callback_cs_.get());
|
|
if (!codec_observer_)
|
|
return;
|
|
codec_observer_->DecoderTiming(decode_ms,
|
|
max_decode_ms,
|
|
current_delay_ms,
|
|
target_delay_ms,
|
|
jitter_buffer_ms,
|
|
min_playout_delay_ms,
|
|
render_delay_ms);
|
|
}
|
|
|
|
int32_t ViEChannel::RequestKeyFrame() {
|
|
WEBRTC_TRACE(kTraceStream, kTraceVideo, ViEId(engine_id_, channel_id_),
|
|
"%s", __FUNCTION__);
|
|
{
|
|
CriticalSectionScoped cs(callback_cs_.get());
|
|
if (codec_observer_ && do_key_frame_callbackRequest_) {
|
|
codec_observer_->RequestNewKeyFrame(channel_id_);
|
|
}
|
|
}
|
|
return rtp_rtcp_->RequestKeyFrame();
|
|
}
|
|
|
|
int32_t ViEChannel::SliceLossIndicationRequest(
|
|
const uint64_t picture_id) {
|
|
return rtp_rtcp_->SendRTCPSliceLossIndication((uint8_t) picture_id);
|
|
}
|
|
|
|
int32_t ViEChannel::ResendPackets(const uint16_t* sequence_numbers,
|
|
uint16_t length) {
|
|
WEBRTC_TRACE(kTraceStream, kTraceVideo, ViEId(engine_id_, channel_id_),
|
|
"%s(length: %d)", __FUNCTION__, length);
|
|
return rtp_rtcp_->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(uint32_t rtt) {
|
|
vcm_.SetReceiveChannelParameters(rtt);
|
|
}
|
|
|
|
int32_t ViEChannel::StartDecodeThread() {
|
|
// Start the decode thread
|
|
if (decode_thread_) {
|
|
// Already started.
|
|
return 0;
|
|
}
|
|
decode_thread_ = ThreadWrapper::CreateThread(ChannelDecodeThreadFunction,
|
|
this, kHighestPriority,
|
|
"DecodingThread");
|
|
if (!decode_thread_) {
|
|
WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
|
|
"%s: could not create decode thread", __FUNCTION__);
|
|
return -1;
|
|
}
|
|
|
|
unsigned int thread_id;
|
|
if (decode_thread_->Start(thread_id) == false) {
|
|
delete decode_thread_;
|
|
decode_thread_ = NULL;
|
|
WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
|
|
"%s: could not start decode thread", __FUNCTION__);
|
|
return -1;
|
|
}
|
|
|
|
WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
|
|
"%s: decode thread with id %u started", __FUNCTION__);
|
|
return 0;
|
|
}
|
|
|
|
int32_t ViEChannel::StopDecodeThread() {
|
|
if (!decode_thread_) {
|
|
WEBRTC_TRACE(kTraceWarning, kTraceVideo, ViEId(engine_id_, channel_id_),
|
|
"%s: decode thread not running", __FUNCTION__);
|
|
return 0;
|
|
}
|
|
|
|
decode_thread_->SetNotAlive();
|
|
if (decode_thread_->Stop()) {
|
|
delete decode_thread_;
|
|
} else {
|
|
// Couldn't stop the thread, leak instead of crash.
|
|
WEBRTC_TRACE(kTraceWarning, kTraceVideo, ViEId(engine_id_, channel_id_),
|
|
"%s: could not stop decode thread", __FUNCTION__);
|
|
assert(false && "could not stop decode thread");
|
|
}
|
|
decode_thread_ = NULL;
|
|
return 0;
|
|
}
|
|
|
|
int32_t ViEChannel::RegisterExternalEncryption(Encryption* encryption) {
|
|
WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_), "%s",
|
|
__FUNCTION__);
|
|
|
|
CriticalSectionScoped cs(callback_cs_.get());
|
|
if (external_encryption_) {
|
|
WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
|
|
"%s: external encryption already registered", __FUNCTION__);
|
|
return -1;
|
|
}
|
|
|
|
external_encryption_ = encryption;
|
|
|
|
vie_receiver_.RegisterExternalDecryption(encryption);
|
|
vie_sender_.RegisterExternalEncryption(encryption);
|
|
|
|
WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
|
|
"%s", "external encryption object registerd with channel=%d",
|
|
channel_id_);
|
|
return 0;
|
|
}
|
|
|
|
int32_t ViEChannel::DeRegisterExternalEncryption() {
|
|
WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_), "%s",
|
|
__FUNCTION__);
|
|
|
|
CriticalSectionScoped cs(callback_cs_.get());
|
|
if (!external_encryption_) {
|
|
WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
|
|
"%s: external encryption is not registered", __FUNCTION__);
|
|
return -1;
|
|
}
|
|
|
|
external_transport_ = NULL;
|
|
vie_receiver_.DeregisterExternalDecryption();
|
|
vie_sender_.DeregisterExternalEncryption();
|
|
WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
|
|
"%s external encryption object de-registerd with channel=%d",
|
|
__FUNCTION__, channel_id_);
|
|
return 0;
|
|
}
|
|
|
|
int32_t ViEChannel::SetVoiceChannel(int32_t ve_channel_id,
|
|
VoEVideoSync* ve_sync_interface) {
|
|
WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
|
|
"%s, audio channel %d, video channel %d", __FUNCTION__,
|
|
ve_channel_id, channel_id_);
|
|
|
|
if (ve_sync_interface) {
|
|
// Register lip sync
|
|
module_process_thread_.RegisterModule(&vie_sync_);
|
|
} else {
|
|
module_process_thread_.DeRegisterModule(&vie_sync_);
|
|
}
|
|
return vie_sync_.ConfigureSync(ve_channel_id,
|
|
ve_sync_interface,
|
|
rtp_rtcp_.get(),
|
|
vie_receiver_.GetRtpReceiver());
|
|
}
|
|
|
|
int32_t ViEChannel::VoiceChannel() {
|
|
return vie_sync_.VoiceChannel();
|
|
}
|
|
|
|
int32_t ViEChannel::RegisterEffectFilter(ViEEffectFilter* effect_filter) {
|
|
CriticalSectionScoped cs(callback_cs_.get());
|
|
if (!effect_filter) {
|
|
if (!effect_filter_) {
|
|
WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
|
|
"%s: no effect filter added for channel %d",
|
|
__FUNCTION__, channel_id_);
|
|
return -1;
|
|
}
|
|
WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
|
|
"%s: deregister effect filter for device %d", __FUNCTION__,
|
|
channel_id_);
|
|
} else {
|
|
WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
|
|
"%s: register effect filter for device %d", __FUNCTION__,
|
|
channel_id_);
|
|
if (effect_filter_) {
|
|
WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
|
|
"%s: effect filter already added for channel %d",
|
|
__FUNCTION__, channel_id_);
|
|
return -1;
|
|
}
|
|
}
|
|
effect_filter_ = effect_filter;
|
|
return 0;
|
|
}
|
|
|
|
void ViEChannel::RegisterPreRenderCallback(
|
|
I420FrameCallback* pre_render_callback) {
|
|
CriticalSectionScoped cs(callback_cs_.get());
|
|
pre_render_callback_ = pre_render_callback;
|
|
}
|
|
|
|
void ViEChannel::RegisterPreDecodeImageCallback(
|
|
EncodedImageCallback* pre_decode_callback) {
|
|
CriticalSectionScoped cs(callback_cs_.get());
|
|
vcm_.RegisterPreDecodeImageCallback(pre_decode_callback);
|
|
}
|
|
|
|
void ViEChannel::OnApplicationDataReceived(const int32_t id,
|
|
const uint8_t sub_type,
|
|
const uint32_t name,
|
|
const uint16_t length,
|
|
const uint8_t* data) {
|
|
if (channel_id_ != ChannelId(id)) {
|
|
WEBRTC_TRACE(kTraceStream, kTraceVideo, ViEId(engine_id_, channel_id_),
|
|
"%s, incorrect id", __FUNCTION__, id);
|
|
return;
|
|
}
|
|
CriticalSectionScoped cs(callback_cs_.get());
|
|
{
|
|
if (rtcp_observer_) {
|
|
rtcp_observer_->OnApplicationDataReceived(
|
|
channel_id_, sub_type, name, reinterpret_cast<const char*>(data),
|
|
length);
|
|
}
|
|
}
|
|
}
|
|
|
|
int32_t ViEChannel::OnInitializeDecoder(
|
|
const int32_t id,
|
|
const int8_t payload_type,
|
|
const char payload_name[RTP_PAYLOAD_NAME_SIZE],
|
|
const int frequency,
|
|
const uint8_t channels,
|
|
const uint32_t rate) {
|
|
WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
|
|
"%s: payload_type %d, payload_name %s", __FUNCTION__,
|
|
payload_type, payload_name);
|
|
vcm_.ResetDecoder();
|
|
|
|
CriticalSectionScoped cs(callback_cs_.get());
|
|
decoder_reset_ = true;
|
|
return 0;
|
|
}
|
|
|
|
void ViEChannel::OnIncomingSSRCChanged(const int32_t id, const uint32_t ssrc) {
|
|
if (channel_id_ != ChannelId(id)) {
|
|
assert(false);
|
|
WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
|
|
"%s, incorrect id", __FUNCTION__, id);
|
|
return;
|
|
}
|
|
|
|
WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
|
|
"%s: %u", __FUNCTION__, ssrc);
|
|
|
|
rtp_rtcp_->SetRemoteSSRC(ssrc);
|
|
|
|
CriticalSectionScoped cs(callback_cs_.get());
|
|
{
|
|
if (rtp_observer_) {
|
|
rtp_observer_->IncomingSSRCChanged(channel_id_, ssrc);
|
|
}
|
|
}
|
|
}
|
|
|
|
void ViEChannel::OnIncomingCSRCChanged(const int32_t id,
|
|
const uint32_t CSRC,
|
|
const bool added) {
|
|
WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
|
|
"%s: %u added: %d", __FUNCTION__, CSRC, added);
|
|
|
|
if (channel_id_ != ChannelId(id)) {
|
|
assert(false);
|
|
WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
|
|
"%s, incorrect id", __FUNCTION__, id);
|
|
return;
|
|
}
|
|
|
|
WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
|
|
"%s: %u", __FUNCTION__, CSRC);
|
|
|
|
CriticalSectionScoped cs(callback_cs_.get());
|
|
{
|
|
if (rtp_observer_) {
|
|
rtp_observer_->IncomingCSRCChanged(channel_id_, CSRC, added);
|
|
}
|
|
}
|
|
}
|
|
|
|
void ViEChannel::ResetStatistics(uint32_t ssrc) {
|
|
StreamStatistician* statistician =
|
|
vie_receiver_.GetReceiveStatistics()->GetStatistician(ssrc);
|
|
if (statistician)
|
|
statistician->ResetStatistics();
|
|
}
|
|
|
|
} // namespace webrtc
|