0200f70792
Should not affect webrtc standalone. For chromium, disabling helps mitigate viewing performance problems. BUG=chromium:441440 R=mflodman@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/41909004 Cr-Commit-Position: refs/heads/master@{#8375} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8375 4adac7df-926f-26a2-2b94-8c16560cd09d
1732 lines
57 KiB
C++
1732 lines
57 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/modules/rtp_rtcp/source/rtp_sender.h"
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#include <stdlib.h> // srand
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#include "webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h"
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#include "webrtc/modules/rtp_rtcp/source/rtp_sender_video.h"
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#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
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#include "webrtc/system_wrappers/interface/logging.h"
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#include "webrtc/system_wrappers/interface/tick_util.h"
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#include "webrtc/system_wrappers/interface/trace_event.h"
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namespace webrtc {
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// Max in the RFC 3550 is 255 bytes, we limit it to be modulus 32 for SRTP.
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const size_t kMaxPaddingLength = 224;
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const int kSendSideDelayWindowMs = 1000;
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namespace {
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const char* FrameTypeToString(FrameType frame_type) {
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switch (frame_type) {
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case kFrameEmpty: return "empty";
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case kAudioFrameSpeech: return "audio_speech";
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case kAudioFrameCN: return "audio_cn";
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case kVideoFrameKey: return "video_key";
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case kVideoFrameDelta: return "video_delta";
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}
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return "";
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}
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} // namespace
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class BitrateAggregator {
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public:
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explicit BitrateAggregator(BitrateStatisticsObserver* bitrate_callback)
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: callback_(bitrate_callback),
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total_bitrate_observer_(*this),
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retransmit_bitrate_observer_(*this),
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ssrc_(0) {}
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void OnStatsUpdated() const {
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if (callback_)
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callback_->Notify(total_bitrate_observer_.statistics(),
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retransmit_bitrate_observer_.statistics(),
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ssrc_);
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}
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Bitrate::Observer* total_bitrate_observer() {
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return &total_bitrate_observer_;
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}
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Bitrate::Observer* retransmit_bitrate_observer() {
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return &retransmit_bitrate_observer_;
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}
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void set_ssrc(uint32_t ssrc) { ssrc_ = ssrc; }
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private:
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// We assume that these observers are called on the same thread, which is
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// true for RtpSender as they are called on the Process thread.
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class BitrateObserver : public Bitrate::Observer {
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public:
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explicit BitrateObserver(const BitrateAggregator& aggregator)
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: aggregator_(aggregator) {}
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// Implements Bitrate::Observer.
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virtual void BitrateUpdated(const BitrateStatistics& stats) OVERRIDE {
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statistics_ = stats;
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aggregator_.OnStatsUpdated();
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}
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BitrateStatistics statistics() const { return statistics_; }
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private:
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BitrateStatistics statistics_;
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const BitrateAggregator& aggregator_;
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};
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BitrateStatisticsObserver* const callback_;
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BitrateObserver total_bitrate_observer_;
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BitrateObserver retransmit_bitrate_observer_;
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uint32_t ssrc_;
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};
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RTPSender::RTPSender(int32_t id,
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bool audio,
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Clock* clock,
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Transport* transport,
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RtpAudioFeedback* audio_feedback,
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PacedSender* paced_sender,
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BitrateStatisticsObserver* bitrate_callback,
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FrameCountObserver* frame_count_observer,
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SendSideDelayObserver* send_side_delay_observer)
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: clock_(clock),
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// TODO(holmer): Remove this conversion when we remove the use of
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// TickTime.
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clock_delta_ms_(clock_->TimeInMilliseconds() -
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TickTime::MillisecondTimestamp()),
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bitrates_(new BitrateAggregator(bitrate_callback)),
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total_bitrate_sent_(clock, bitrates_->total_bitrate_observer()),
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id_(id),
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audio_configured_(audio),
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audio_(audio ? new RTPSenderAudio(id, clock, this, audio_feedback)
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: nullptr),
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video_(audio ? nullptr : new RTPSenderVideo(clock, this)),
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paced_sender_(paced_sender),
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last_capture_time_ms_sent_(0),
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send_critsect_(CriticalSectionWrapper::CreateCriticalSection()),
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transport_(transport),
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sending_media_(true), // Default to sending media.
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max_payload_length_(IP_PACKET_SIZE - 28), // Default is IP-v4/UDP.
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packet_over_head_(28),
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payload_type_(-1),
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payload_type_map_(),
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rtp_header_extension_map_(),
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transmission_time_offset_(0),
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absolute_send_time_(0),
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// NACK.
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nack_byte_count_times_(),
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nack_byte_count_(),
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nack_bitrate_(clock, bitrates_->retransmit_bitrate_observer()),
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packet_history_(clock),
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// Statistics
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statistics_crit_(CriticalSectionWrapper::CreateCriticalSection()),
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rtp_stats_callback_(NULL),
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frame_count_observer_(frame_count_observer),
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send_side_delay_observer_(send_side_delay_observer),
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// RTP variables
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start_timestamp_forced_(false),
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start_timestamp_(0),
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ssrc_db_(*SSRCDatabase::GetSSRCDatabase()),
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remote_ssrc_(0),
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sequence_number_forced_(false),
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ssrc_forced_(false),
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timestamp_(0),
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capture_time_ms_(0),
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last_timestamp_time_ms_(0),
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media_has_been_sent_(false),
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last_packet_marker_bit_(false),
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csrcs_(),
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rtx_(kRtxOff),
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payload_type_rtx_(-1),
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target_bitrate_critsect_(CriticalSectionWrapper::CreateCriticalSection()),
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target_bitrate_(0) {
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memset(nack_byte_count_times_, 0, sizeof(nack_byte_count_times_));
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memset(nack_byte_count_, 0, sizeof(nack_byte_count_));
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// We need to seed the random generator.
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srand(static_cast<uint32_t>(clock_->TimeInMilliseconds()));
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ssrc_ = ssrc_db_.CreateSSRC(); // Can't be 0.
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ssrc_rtx_ = ssrc_db_.CreateSSRC(); // Can't be 0.
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bitrates_->set_ssrc(ssrc_);
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// Random start, 16 bits. Can't be 0.
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sequence_number_rtx_ = static_cast<uint16_t>(rand() + 1) & 0x7FFF;
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sequence_number_ = static_cast<uint16_t>(rand() + 1) & 0x7FFF;
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}
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RTPSender::~RTPSender() {
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if (remote_ssrc_ != 0) {
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ssrc_db_.ReturnSSRC(remote_ssrc_);
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}
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ssrc_db_.ReturnSSRC(ssrc_);
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SSRCDatabase::ReturnSSRCDatabase();
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while (!payload_type_map_.empty()) {
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std::map<int8_t, RtpUtility::Payload*>::iterator it =
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payload_type_map_.begin();
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delete it->second;
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payload_type_map_.erase(it);
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}
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}
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void RTPSender::SetTargetBitrate(uint32_t bitrate) {
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CriticalSectionScoped cs(target_bitrate_critsect_.get());
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target_bitrate_ = bitrate;
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}
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uint32_t RTPSender::GetTargetBitrate() {
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CriticalSectionScoped cs(target_bitrate_critsect_.get());
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return target_bitrate_;
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}
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uint16_t RTPSender::ActualSendBitrateKbit() const {
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return (uint16_t)(total_bitrate_sent_.BitrateNow() / 1000);
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}
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uint32_t RTPSender::VideoBitrateSent() const {
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if (video_) {
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return video_->VideoBitrateSent();
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}
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return 0;
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}
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uint32_t RTPSender::FecOverheadRate() const {
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if (video_) {
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return video_->FecOverheadRate();
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}
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return 0;
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}
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uint32_t RTPSender::NackOverheadRate() const {
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return nack_bitrate_.BitrateLast();
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}
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bool RTPSender::GetSendSideDelay(int* avg_send_delay_ms,
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int* max_send_delay_ms) const {
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CriticalSectionScoped lock(statistics_crit_.get());
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SendDelayMap::const_iterator it = send_delays_.upper_bound(
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clock_->TimeInMilliseconds() - kSendSideDelayWindowMs);
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if (it == send_delays_.end())
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return false;
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int num_delays = 0;
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for (; it != send_delays_.end(); ++it) {
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*max_send_delay_ms = std::max(*max_send_delay_ms, it->second);
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*avg_send_delay_ms += it->second;
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++num_delays;
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}
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*avg_send_delay_ms = (*avg_send_delay_ms + num_delays / 2) / num_delays;
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return true;
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}
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int32_t RTPSender::SetTransmissionTimeOffset(int32_t transmission_time_offset) {
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if (transmission_time_offset > (0x800000 - 1) ||
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transmission_time_offset < -(0x800000 - 1)) { // Word24.
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return -1;
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}
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CriticalSectionScoped cs(send_critsect_.get());
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transmission_time_offset_ = transmission_time_offset;
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return 0;
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}
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int32_t RTPSender::SetAbsoluteSendTime(uint32_t absolute_send_time) {
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if (absolute_send_time > 0xffffff) { // UWord24.
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return -1;
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}
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CriticalSectionScoped cs(send_critsect_.get());
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absolute_send_time_ = absolute_send_time;
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return 0;
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}
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int32_t RTPSender::RegisterRtpHeaderExtension(RTPExtensionType type,
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uint8_t id) {
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CriticalSectionScoped cs(send_critsect_.get());
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return rtp_header_extension_map_.Register(type, id);
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}
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int32_t RTPSender::DeregisterRtpHeaderExtension(RTPExtensionType type) {
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CriticalSectionScoped cs(send_critsect_.get());
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return rtp_header_extension_map_.Deregister(type);
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}
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size_t RTPSender::RtpHeaderExtensionTotalLength() const {
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CriticalSectionScoped cs(send_critsect_.get());
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return rtp_header_extension_map_.GetTotalLengthInBytes();
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}
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int32_t RTPSender::RegisterPayload(
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const char payload_name[RTP_PAYLOAD_NAME_SIZE],
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int8_t payload_number,
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uint32_t frequency,
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uint8_t channels,
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uint32_t rate) {
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assert(payload_name);
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CriticalSectionScoped cs(send_critsect_.get());
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std::map<int8_t, RtpUtility::Payload*>::iterator it =
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payload_type_map_.find(payload_number);
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if (payload_type_map_.end() != it) {
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// We already use this payload type.
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RtpUtility::Payload* payload = it->second;
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assert(payload);
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// Check if it's the same as we already have.
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if (RtpUtility::StringCompare(
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payload->name, payload_name, RTP_PAYLOAD_NAME_SIZE - 1)) {
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if (audio_configured_ && payload->audio &&
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payload->typeSpecific.Audio.frequency == frequency &&
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(payload->typeSpecific.Audio.rate == rate ||
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payload->typeSpecific.Audio.rate == 0 || rate == 0)) {
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payload->typeSpecific.Audio.rate = rate;
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// Ensure that we update the rate if new or old is zero.
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return 0;
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}
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if (!audio_configured_ && !payload->audio) {
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return 0;
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}
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}
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return -1;
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}
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int32_t ret_val = -1;
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RtpUtility::Payload* payload = NULL;
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if (audio_configured_) {
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ret_val = audio_->RegisterAudioPayload(payload_name, payload_number,
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frequency, channels, rate, payload);
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} else {
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ret_val = video_->RegisterVideoPayload(payload_name, payload_number, rate,
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payload);
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}
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if (payload) {
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payload_type_map_[payload_number] = payload;
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}
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return ret_val;
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}
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int32_t RTPSender::DeRegisterSendPayload(int8_t payload_type) {
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CriticalSectionScoped lock(send_critsect_.get());
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std::map<int8_t, RtpUtility::Payload*>::iterator it =
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payload_type_map_.find(payload_type);
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if (payload_type_map_.end() == it) {
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return -1;
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}
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RtpUtility::Payload* payload = it->second;
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delete payload;
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payload_type_map_.erase(it);
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return 0;
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}
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void RTPSender::SetSendPayloadType(int8_t payload_type) {
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CriticalSectionScoped cs(send_critsect_.get());
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payload_type_ = payload_type;
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}
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int8_t RTPSender::SendPayloadType() const {
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CriticalSectionScoped cs(send_critsect_.get());
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return payload_type_;
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}
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int RTPSender::SendPayloadFrequency() const {
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return audio_ != NULL ? audio_->AudioFrequency() : kVideoPayloadTypeFrequency;
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}
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int32_t RTPSender::SetMaxPayloadLength(size_t max_payload_length,
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uint16_t packet_over_head) {
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// Sanity check.
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if (max_payload_length < 100 || max_payload_length > IP_PACKET_SIZE) {
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LOG(LS_ERROR) << "Invalid max payload length: " << max_payload_length;
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return -1;
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}
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CriticalSectionScoped cs(send_critsect_.get());
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max_payload_length_ = max_payload_length;
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packet_over_head_ = packet_over_head;
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return 0;
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}
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size_t RTPSender::MaxDataPayloadLength() const {
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int rtx;
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{
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CriticalSectionScoped rtx_lock(send_critsect_.get());
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rtx = rtx_;
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}
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if (audio_configured_) {
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return max_payload_length_ - RTPHeaderLength();
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} else {
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return max_payload_length_ - RTPHeaderLength() // RTP overhead.
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- video_->FECPacketOverhead() // FEC/ULP/RED overhead.
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- ((rtx) ? 2 : 0); // RTX overhead.
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}
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}
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size_t RTPSender::MaxPayloadLength() const {
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return max_payload_length_;
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}
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uint16_t RTPSender::PacketOverHead() const { return packet_over_head_; }
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void RTPSender::SetRtxStatus(int mode) {
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CriticalSectionScoped cs(send_critsect_.get());
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rtx_ = mode;
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}
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int RTPSender::RtxStatus() const {
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CriticalSectionScoped cs(send_critsect_.get());
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return rtx_;
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}
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void RTPSender::SetRtxSsrc(uint32_t ssrc) {
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CriticalSectionScoped cs(send_critsect_.get());
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ssrc_rtx_ = ssrc;
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}
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uint32_t RTPSender::RtxSsrc() const {
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CriticalSectionScoped cs(send_critsect_.get());
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return ssrc_rtx_;
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}
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void RTPSender::SetRtxPayloadType(int payload_type) {
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CriticalSectionScoped cs(send_critsect_.get());
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payload_type_rtx_ = payload_type;
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}
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int32_t RTPSender::CheckPayloadType(int8_t payload_type,
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RtpVideoCodecTypes* video_type) {
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CriticalSectionScoped cs(send_critsect_.get());
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if (payload_type < 0) {
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LOG(LS_ERROR) << "Invalid payload_type " << payload_type;
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return -1;
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}
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if (audio_configured_) {
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int8_t red_pl_type = -1;
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if (audio_->RED(red_pl_type) == 0) {
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// We have configured RED.
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if (red_pl_type == payload_type) {
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// And it's a match...
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return 0;
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}
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}
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}
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if (payload_type_ == payload_type) {
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if (!audio_configured_) {
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*video_type = video_->VideoCodecType();
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}
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return 0;
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}
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std::map<int8_t, RtpUtility::Payload*>::iterator it =
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payload_type_map_.find(payload_type);
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if (it == payload_type_map_.end()) {
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LOG(LS_WARNING) << "Payload type " << payload_type << " not registered.";
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return -1;
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}
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SetSendPayloadType(payload_type);
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RtpUtility::Payload* payload = it->second;
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assert(payload);
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if (!payload->audio && !audio_configured_) {
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video_->SetVideoCodecType(payload->typeSpecific.Video.videoCodecType);
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*video_type = payload->typeSpecific.Video.videoCodecType;
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video_->SetMaxConfiguredBitrateVideo(payload->typeSpecific.Video.maxRate);
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}
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return 0;
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}
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int32_t RTPSender::SendOutgoingData(FrameType frame_type,
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int8_t payload_type,
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uint32_t capture_timestamp,
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int64_t capture_time_ms,
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const uint8_t* payload_data,
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size_t payload_size,
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const RTPFragmentationHeader* fragmentation,
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VideoCodecInformation* codec_info,
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const RTPVideoTypeHeader* rtp_type_hdr) {
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uint32_t ssrc;
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{
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// Drop this packet if we're not sending media packets.
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CriticalSectionScoped cs(send_critsect_.get());
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ssrc = ssrc_;
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if (!sending_media_) {
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return 0;
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}
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}
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RtpVideoCodecTypes video_type = kRtpVideoGeneric;
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if (CheckPayloadType(payload_type, &video_type) != 0) {
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LOG(LS_ERROR) << "Don't send data with unknown payload type.";
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return -1;
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}
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uint32_t ret_val;
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if (audio_configured_) {
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TRACE_EVENT_ASYNC_STEP1("webrtc", "Audio", capture_timestamp,
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"Send", "type", FrameTypeToString(frame_type));
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assert(frame_type == kAudioFrameSpeech || frame_type == kAudioFrameCN ||
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frame_type == kFrameEmpty);
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ret_val = audio_->SendAudio(frame_type, payload_type, capture_timestamp,
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payload_data, payload_size, fragmentation);
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} else {
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TRACE_EVENT_ASYNC_STEP1("webrtc", "Video", capture_time_ms,
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"Send", "type", FrameTypeToString(frame_type));
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assert(frame_type != kAudioFrameSpeech && frame_type != kAudioFrameCN);
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if (frame_type == kFrameEmpty)
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return 0;
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ret_val = video_->SendVideo(video_type, frame_type, payload_type,
|
|
capture_timestamp, capture_time_ms,
|
|
payload_data, payload_size,
|
|
fragmentation, codec_info,
|
|
rtp_type_hdr);
|
|
|
|
}
|
|
|
|
CriticalSectionScoped cs(statistics_crit_.get());
|
|
// Note: This is currently only counting for video.
|
|
if (frame_type == kVideoFrameKey) {
|
|
++frame_counts_.key_frames;
|
|
} else if (frame_type == kVideoFrameDelta) {
|
|
++frame_counts_.delta_frames;
|
|
}
|
|
if (frame_count_observer_) {
|
|
frame_count_observer_->FrameCountUpdated(frame_counts_, ssrc);
|
|
}
|
|
|
|
return ret_val;
|
|
}
|
|
|
|
size_t RTPSender::TrySendRedundantPayloads(size_t bytes_to_send) {
|
|
{
|
|
CriticalSectionScoped cs(send_critsect_.get());
|
|
if ((rtx_ & kRtxRedundantPayloads) == 0)
|
|
return 0;
|
|
}
|
|
|
|
uint8_t buffer[IP_PACKET_SIZE];
|
|
int bytes_left = static_cast<int>(bytes_to_send);
|
|
while (bytes_left > 0) {
|
|
size_t length = bytes_left;
|
|
int64_t capture_time_ms;
|
|
if (!packet_history_.GetBestFittingPacket(buffer, &length,
|
|
&capture_time_ms)) {
|
|
break;
|
|
}
|
|
if (!PrepareAndSendPacket(buffer, length, capture_time_ms, true, false))
|
|
break;
|
|
RtpUtility::RtpHeaderParser rtp_parser(buffer, length);
|
|
RTPHeader rtp_header;
|
|
rtp_parser.Parse(rtp_header);
|
|
bytes_left -= static_cast<int>(length - rtp_header.headerLength);
|
|
}
|
|
return bytes_to_send - bytes_left;
|
|
}
|
|
|
|
size_t RTPSender::BuildPaddingPacket(uint8_t* packet, size_t header_length) {
|
|
size_t padding_bytes_in_packet = kMaxPaddingLength;
|
|
packet[0] |= 0x20; // Set padding bit.
|
|
int32_t *data =
|
|
reinterpret_cast<int32_t *>(&(packet[header_length]));
|
|
|
|
// Fill data buffer with random data.
|
|
for (size_t j = 0; j < (padding_bytes_in_packet >> 2); ++j) {
|
|
data[j] = rand(); // NOLINT
|
|
}
|
|
// Set number of padding bytes in the last byte of the packet.
|
|
packet[header_length + padding_bytes_in_packet - 1] =
|
|
static_cast<uint8_t>(padding_bytes_in_packet);
|
|
return padding_bytes_in_packet;
|
|
}
|
|
|
|
size_t RTPSender::TrySendPadData(size_t bytes) {
|
|
int64_t capture_time_ms;
|
|
uint32_t timestamp;
|
|
{
|
|
CriticalSectionScoped cs(send_critsect_.get());
|
|
timestamp = timestamp_;
|
|
capture_time_ms = capture_time_ms_;
|
|
if (last_timestamp_time_ms_ > 0) {
|
|
timestamp +=
|
|
(clock_->TimeInMilliseconds() - last_timestamp_time_ms_) * 90;
|
|
capture_time_ms +=
|
|
(clock_->TimeInMilliseconds() - last_timestamp_time_ms_);
|
|
}
|
|
}
|
|
return SendPadData(timestamp, capture_time_ms, bytes);
|
|
}
|
|
|
|
size_t RTPSender::SendPadData(uint32_t timestamp,
|
|
int64_t capture_time_ms,
|
|
size_t bytes) {
|
|
size_t padding_bytes_in_packet = 0;
|
|
size_t bytes_sent = 0;
|
|
for (; bytes > 0; bytes -= padding_bytes_in_packet) {
|
|
// Always send full padding packets.
|
|
if (bytes < kMaxPaddingLength)
|
|
bytes = kMaxPaddingLength;
|
|
|
|
uint32_t ssrc;
|
|
uint16_t sequence_number;
|
|
int payload_type;
|
|
bool over_rtx;
|
|
{
|
|
CriticalSectionScoped cs(send_critsect_.get());
|
|
// Only send padding packets following the last packet of a frame,
|
|
// indicated by the marker bit.
|
|
if (rtx_ == kRtxOff) {
|
|
// Without RTX we can't send padding in the middle of frames.
|
|
if (!last_packet_marker_bit_)
|
|
return 0;
|
|
ssrc = ssrc_;
|
|
sequence_number = sequence_number_;
|
|
++sequence_number_;
|
|
payload_type = payload_type_;
|
|
over_rtx = false;
|
|
} else {
|
|
// Without abs-send-time a media packet must be sent before padding so
|
|
// that the timestamps used for estimation are correct.
|
|
if (!media_has_been_sent_ && !rtp_header_extension_map_.IsRegistered(
|
|
kRtpExtensionAbsoluteSendTime))
|
|
return 0;
|
|
ssrc = ssrc_rtx_;
|
|
sequence_number = sequence_number_rtx_;
|
|
++sequence_number_rtx_;
|
|
payload_type = ((rtx_ & kRtxRedundantPayloads) > 0) ? payload_type_rtx_
|
|
: payload_type_;
|
|
over_rtx = true;
|
|
}
|
|
}
|
|
|
|
uint8_t padding_packet[IP_PACKET_SIZE];
|
|
size_t header_length =
|
|
CreateRtpHeader(padding_packet, payload_type, ssrc, false, timestamp,
|
|
sequence_number, std::vector<uint32_t>());
|
|
assert(header_length != static_cast<size_t>(-1));
|
|
padding_bytes_in_packet = BuildPaddingPacket(padding_packet, header_length);
|
|
assert(padding_bytes_in_packet <= bytes);
|
|
size_t length = padding_bytes_in_packet + header_length;
|
|
int64_t now_ms = clock_->TimeInMilliseconds();
|
|
|
|
RtpUtility::RtpHeaderParser rtp_parser(padding_packet, length);
|
|
RTPHeader rtp_header;
|
|
rtp_parser.Parse(rtp_header);
|
|
|
|
if (capture_time_ms > 0) {
|
|
UpdateTransmissionTimeOffset(
|
|
padding_packet, length, rtp_header, now_ms - capture_time_ms);
|
|
}
|
|
|
|
UpdateAbsoluteSendTime(padding_packet, length, rtp_header, now_ms);
|
|
if (!SendPacketToNetwork(padding_packet, length))
|
|
break;
|
|
bytes_sent += padding_bytes_in_packet;
|
|
UpdateRtpStats(padding_packet, length, rtp_header, over_rtx, false);
|
|
}
|
|
|
|
return bytes_sent;
|
|
}
|
|
|
|
void RTPSender::SetStorePacketsStatus(bool enable, uint16_t number_to_store) {
|
|
packet_history_.SetStorePacketsStatus(enable, number_to_store);
|
|
}
|
|
|
|
bool RTPSender::StorePackets() const {
|
|
return packet_history_.StorePackets();
|
|
}
|
|
|
|
int32_t RTPSender::ReSendPacket(uint16_t packet_id, int64_t min_resend_time) {
|
|
size_t length = IP_PACKET_SIZE;
|
|
uint8_t data_buffer[IP_PACKET_SIZE];
|
|
int64_t capture_time_ms;
|
|
if (!packet_history_.GetPacketAndSetSendTime(packet_id, min_resend_time, true,
|
|
data_buffer, &length,
|
|
&capture_time_ms)) {
|
|
// Packet not found.
|
|
return 0;
|
|
}
|
|
|
|
if (paced_sender_) {
|
|
RtpUtility::RtpHeaderParser rtp_parser(data_buffer, length);
|
|
RTPHeader header;
|
|
if (!rtp_parser.Parse(header)) {
|
|
assert(false);
|
|
return -1;
|
|
}
|
|
// Convert from TickTime to Clock since capture_time_ms is based on
|
|
// TickTime.
|
|
int64_t corrected_capture_tims_ms = capture_time_ms + clock_delta_ms_;
|
|
if (!paced_sender_->SendPacket(
|
|
PacedSender::kHighPriority, header.ssrc, header.sequenceNumber,
|
|
corrected_capture_tims_ms, length - header.headerLength, true)) {
|
|
// We can't send the packet right now.
|
|
// We will be called when it is time.
|
|
return length;
|
|
}
|
|
}
|
|
int rtx = kRtxOff;
|
|
{
|
|
CriticalSectionScoped lock(send_critsect_.get());
|
|
rtx = rtx_;
|
|
}
|
|
return PrepareAndSendPacket(data_buffer, length, capture_time_ms,
|
|
(rtx & kRtxRetransmitted) > 0, true) ?
|
|
static_cast<int32_t>(length) : -1;
|
|
}
|
|
|
|
bool RTPSender::SendPacketToNetwork(const uint8_t *packet, size_t size) {
|
|
int bytes_sent = -1;
|
|
if (transport_) {
|
|
bytes_sent = transport_->SendPacket(id_, packet, size);
|
|
}
|
|
TRACE_EVENT_INSTANT2(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"),
|
|
"RTPSender::SendPacketToNetwork", "size", size, "sent",
|
|
bytes_sent);
|
|
// TODO(pwestin): Add a separate bitrate for sent bitrate after pacer.
|
|
if (bytes_sent <= 0) {
|
|
LOG(LS_WARNING) << "Transport failed to send packet";
|
|
return false;
|
|
}
|
|
return true;
|
|
}
|
|
|
|
int RTPSender::SelectiveRetransmissions() const {
|
|
if (!video_)
|
|
return -1;
|
|
return video_->SelectiveRetransmissions();
|
|
}
|
|
|
|
int RTPSender::SetSelectiveRetransmissions(uint8_t settings) {
|
|
if (!video_)
|
|
return -1;
|
|
return video_->SetSelectiveRetransmissions(settings);
|
|
}
|
|
|
|
void RTPSender::OnReceivedNACK(const std::list<uint16_t>& nack_sequence_numbers,
|
|
int64_t avg_rtt) {
|
|
TRACE_EVENT2(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"),
|
|
"RTPSender::OnReceivedNACK", "num_seqnum",
|
|
nack_sequence_numbers.size(), "avg_rtt", avg_rtt);
|
|
const int64_t now = clock_->TimeInMilliseconds();
|
|
uint32_t bytes_re_sent = 0;
|
|
uint32_t target_bitrate = GetTargetBitrate();
|
|
|
|
// Enough bandwidth to send NACK?
|
|
if (!ProcessNACKBitRate(now)) {
|
|
LOG(LS_INFO) << "NACK bitrate reached. Skip sending NACK response. Target "
|
|
<< target_bitrate;
|
|
return;
|
|
}
|
|
|
|
for (std::list<uint16_t>::const_iterator it = nack_sequence_numbers.begin();
|
|
it != nack_sequence_numbers.end(); ++it) {
|
|
const int32_t bytes_sent = ReSendPacket(*it, 5 + avg_rtt);
|
|
if (bytes_sent > 0) {
|
|
bytes_re_sent += bytes_sent;
|
|
} else if (bytes_sent == 0) {
|
|
// The packet has previously been resent.
|
|
// Try resending next packet in the list.
|
|
continue;
|
|
} else {
|
|
// Failed to send one Sequence number. Give up the rest in this nack.
|
|
LOG(LS_WARNING) << "Failed resending RTP packet " << *it
|
|
<< ", Discard rest of packets";
|
|
break;
|
|
}
|
|
// Delay bandwidth estimate (RTT * BW).
|
|
if (target_bitrate != 0 && avg_rtt) {
|
|
// kbits/s * ms = bits => bits/8 = bytes
|
|
size_t target_bytes =
|
|
(static_cast<size_t>(target_bitrate / 1000) * avg_rtt) >> 3;
|
|
if (bytes_re_sent > target_bytes) {
|
|
break; // Ignore the rest of the packets in the list.
|
|
}
|
|
}
|
|
}
|
|
if (bytes_re_sent > 0) {
|
|
UpdateNACKBitRate(bytes_re_sent, now);
|
|
}
|
|
}
|
|
|
|
bool RTPSender::ProcessNACKBitRate(uint32_t now) {
|
|
uint32_t num = 0;
|
|
size_t byte_count = 0;
|
|
const uint32_t kAvgIntervalMs = 1000;
|
|
uint32_t target_bitrate = GetTargetBitrate();
|
|
|
|
CriticalSectionScoped cs(send_critsect_.get());
|
|
|
|
if (target_bitrate == 0) {
|
|
return true;
|
|
}
|
|
for (num = 0; num < NACK_BYTECOUNT_SIZE; ++num) {
|
|
if ((now - nack_byte_count_times_[num]) > kAvgIntervalMs) {
|
|
// Don't use data older than 1sec.
|
|
break;
|
|
} else {
|
|
byte_count += nack_byte_count_[num];
|
|
}
|
|
}
|
|
uint32_t time_interval = kAvgIntervalMs;
|
|
if (num == NACK_BYTECOUNT_SIZE) {
|
|
// More than NACK_BYTECOUNT_SIZE nack messages has been received
|
|
// during the last msg_interval.
|
|
if (nack_byte_count_times_[num - 1] <= now) {
|
|
time_interval = now - nack_byte_count_times_[num - 1];
|
|
}
|
|
}
|
|
return (byte_count * 8) < (target_bitrate / 1000 * time_interval);
|
|
}
|
|
|
|
void RTPSender::UpdateNACKBitRate(uint32_t bytes, int64_t now) {
|
|
CriticalSectionScoped cs(send_critsect_.get());
|
|
if (bytes == 0)
|
|
return;
|
|
nack_bitrate_.Update(bytes);
|
|
// Save bitrate statistics.
|
|
// Shift all but first time.
|
|
for (int i = NACK_BYTECOUNT_SIZE - 2; i >= 0; i--) {
|
|
nack_byte_count_[i + 1] = nack_byte_count_[i];
|
|
nack_byte_count_times_[i + 1] = nack_byte_count_times_[i];
|
|
}
|
|
nack_byte_count_[0] = bytes;
|
|
nack_byte_count_times_[0] = now;
|
|
}
|
|
|
|
// Called from pacer when we can send the packet.
|
|
bool RTPSender::TimeToSendPacket(uint16_t sequence_number,
|
|
int64_t capture_time_ms,
|
|
bool retransmission) {
|
|
size_t length = IP_PACKET_SIZE;
|
|
uint8_t data_buffer[IP_PACKET_SIZE];
|
|
int64_t stored_time_ms;
|
|
|
|
if (!packet_history_.GetPacketAndSetSendTime(sequence_number,
|
|
0,
|
|
retransmission,
|
|
data_buffer,
|
|
&length,
|
|
&stored_time_ms)) {
|
|
// Packet cannot be found. Allow sending to continue.
|
|
return true;
|
|
}
|
|
if (!retransmission && capture_time_ms > 0) {
|
|
UpdateDelayStatistics(capture_time_ms, clock_->TimeInMilliseconds());
|
|
}
|
|
int rtx;
|
|
{
|
|
CriticalSectionScoped lock(send_critsect_.get());
|
|
rtx = rtx_;
|
|
}
|
|
return PrepareAndSendPacket(data_buffer,
|
|
length,
|
|
capture_time_ms,
|
|
retransmission && (rtx & kRtxRetransmitted) > 0,
|
|
retransmission);
|
|
}
|
|
|
|
bool RTPSender::PrepareAndSendPacket(uint8_t* buffer,
|
|
size_t length,
|
|
int64_t capture_time_ms,
|
|
bool send_over_rtx,
|
|
bool is_retransmit) {
|
|
uint8_t *buffer_to_send_ptr = buffer;
|
|
|
|
RtpUtility::RtpHeaderParser rtp_parser(buffer, length);
|
|
RTPHeader rtp_header;
|
|
rtp_parser.Parse(rtp_header);
|
|
if (!is_retransmit && rtp_header.markerBit) {
|
|
TRACE_EVENT_ASYNC_END0(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"), "PacedSend",
|
|
capture_time_ms);
|
|
}
|
|
|
|
TRACE_EVENT_INSTANT2(
|
|
TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"), "PrepareAndSendPacket",
|
|
"timestamp", rtp_header.timestamp, "seqnum", rtp_header.sequenceNumber);
|
|
|
|
uint8_t data_buffer_rtx[IP_PACKET_SIZE];
|
|
if (send_over_rtx) {
|
|
BuildRtxPacket(buffer, &length, data_buffer_rtx);
|
|
buffer_to_send_ptr = data_buffer_rtx;
|
|
}
|
|
|
|
int64_t now_ms = clock_->TimeInMilliseconds();
|
|
int64_t diff_ms = now_ms - capture_time_ms;
|
|
UpdateTransmissionTimeOffset(buffer_to_send_ptr, length, rtp_header,
|
|
diff_ms);
|
|
UpdateAbsoluteSendTime(buffer_to_send_ptr, length, rtp_header, now_ms);
|
|
bool ret = SendPacketToNetwork(buffer_to_send_ptr, length);
|
|
if (ret) {
|
|
CriticalSectionScoped lock(send_critsect_.get());
|
|
media_has_been_sent_ = true;
|
|
}
|
|
UpdateRtpStats(buffer_to_send_ptr, length, rtp_header, send_over_rtx,
|
|
is_retransmit);
|
|
return ret;
|
|
}
|
|
|
|
void RTPSender::UpdateRtpStats(const uint8_t* buffer,
|
|
size_t packet_length,
|
|
const RTPHeader& header,
|
|
bool is_rtx,
|
|
bool is_retransmit) {
|
|
StreamDataCounters* counters;
|
|
// Get ssrc before taking statistics_crit_ to avoid possible deadlock.
|
|
uint32_t ssrc = is_rtx ? RtxSsrc() : SSRC();
|
|
|
|
CriticalSectionScoped lock(statistics_crit_.get());
|
|
if (is_rtx) {
|
|
counters = &rtx_rtp_stats_;
|
|
} else {
|
|
counters = &rtp_stats_;
|
|
}
|
|
|
|
total_bitrate_sent_.Update(packet_length);
|
|
|
|
if (counters->first_packet_time_ms == -1) {
|
|
counters->first_packet_time_ms = clock_->TimeInMilliseconds();
|
|
}
|
|
if (IsFecPacket(buffer, header)) {
|
|
counters->fec.AddPacket(packet_length, header);
|
|
}
|
|
if (is_retransmit) {
|
|
counters->retransmitted.AddPacket(packet_length, header);
|
|
}
|
|
counters->transmitted.AddPacket(packet_length, header);
|
|
|
|
if (rtp_stats_callback_) {
|
|
rtp_stats_callback_->DataCountersUpdated(*counters, ssrc);
|
|
}
|
|
}
|
|
|
|
bool RTPSender::IsFecPacket(const uint8_t* buffer,
|
|
const RTPHeader& header) const {
|
|
if (!video_) {
|
|
return false;
|
|
}
|
|
bool fec_enabled;
|
|
uint8_t pt_red;
|
|
uint8_t pt_fec;
|
|
video_->GenericFECStatus(fec_enabled, pt_red, pt_fec);
|
|
return fec_enabled &&
|
|
header.payloadType == pt_red &&
|
|
buffer[header.headerLength] == pt_fec;
|
|
}
|
|
|
|
size_t RTPSender::TimeToSendPadding(size_t bytes) {
|
|
{
|
|
CriticalSectionScoped cs(send_critsect_.get());
|
|
if (!sending_media_) return 0;
|
|
}
|
|
if (bytes == 0)
|
|
return 0;
|
|
size_t bytes_sent = TrySendRedundantPayloads(bytes);
|
|
if (bytes_sent < bytes)
|
|
bytes_sent += TrySendPadData(bytes - bytes_sent);
|
|
return bytes_sent;
|
|
}
|
|
|
|
// TODO(pwestin): send in the RtpHeaderParser to avoid parsing it again.
|
|
int32_t RTPSender::SendToNetwork(
|
|
uint8_t *buffer, size_t payload_length, size_t rtp_header_length,
|
|
int64_t capture_time_ms, StorageType storage,
|
|
PacedSender::Priority priority) {
|
|
RtpUtility::RtpHeaderParser rtp_parser(buffer,
|
|
payload_length + rtp_header_length);
|
|
RTPHeader rtp_header;
|
|
rtp_parser.Parse(rtp_header);
|
|
|
|
int64_t now_ms = clock_->TimeInMilliseconds();
|
|
|
|
// |capture_time_ms| <= 0 is considered invalid.
|
|
// TODO(holmer): This should be changed all over Video Engine so that negative
|
|
// time is consider invalid, while 0 is considered a valid time.
|
|
if (capture_time_ms > 0) {
|
|
UpdateTransmissionTimeOffset(buffer, payload_length + rtp_header_length,
|
|
rtp_header, now_ms - capture_time_ms);
|
|
}
|
|
|
|
UpdateAbsoluteSendTime(buffer, payload_length + rtp_header_length,
|
|
rtp_header, now_ms);
|
|
|
|
// Used for NACK and to spread out the transmission of packets.
|
|
if (packet_history_.PutRTPPacket(buffer, rtp_header_length + payload_length,
|
|
max_payload_length_, capture_time_ms,
|
|
storage) != 0) {
|
|
return -1;
|
|
}
|
|
|
|
if (paced_sender_ && storage != kDontStore) {
|
|
// Correct offset between implementations of millisecond time stamps in
|
|
// TickTime and Clock.
|
|
int64_t corrected_time_ms = capture_time_ms + clock_delta_ms_;
|
|
if (!paced_sender_->SendPacket(priority, rtp_header.ssrc,
|
|
rtp_header.sequenceNumber, corrected_time_ms,
|
|
payload_length, false)) {
|
|
if (last_capture_time_ms_sent_ == 0 ||
|
|
corrected_time_ms > last_capture_time_ms_sent_) {
|
|
last_capture_time_ms_sent_ = corrected_time_ms;
|
|
TRACE_EVENT_ASYNC_BEGIN1(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"),
|
|
"PacedSend", corrected_time_ms,
|
|
"capture_time_ms", corrected_time_ms);
|
|
}
|
|
// We can't send the packet right now.
|
|
// We will be called when it is time.
|
|
return 0;
|
|
}
|
|
}
|
|
if (capture_time_ms > 0) {
|
|
UpdateDelayStatistics(capture_time_ms, now_ms);
|
|
}
|
|
|
|
size_t length = payload_length + rtp_header_length;
|
|
bool sent = SendPacketToNetwork(buffer, length);
|
|
|
|
if (storage != kDontStore) {
|
|
// Mark the packet as sent in the history even if send failed. Dropping a
|
|
// packet here should be treated as any other packet drop so we should be
|
|
// ready for a retransmission.
|
|
packet_history_.SetSent(rtp_header.sequenceNumber);
|
|
}
|
|
if (!sent)
|
|
return -1;
|
|
|
|
{
|
|
CriticalSectionScoped lock(send_critsect_.get());
|
|
media_has_been_sent_ = true;
|
|
}
|
|
UpdateRtpStats(buffer, length, rtp_header, false, false);
|
|
return 0;
|
|
}
|
|
|
|
void RTPSender::UpdateDelayStatistics(int64_t capture_time_ms, int64_t now_ms) {
|
|
uint32_t ssrc;
|
|
int avg_delay_ms = 0;
|
|
int max_delay_ms = 0;
|
|
{
|
|
CriticalSectionScoped lock(send_critsect_.get());
|
|
ssrc = ssrc_;
|
|
}
|
|
{
|
|
CriticalSectionScoped cs(statistics_crit_.get());
|
|
// TODO(holmer): Compute this iteratively instead.
|
|
send_delays_[now_ms] = now_ms - capture_time_ms;
|
|
send_delays_.erase(send_delays_.begin(),
|
|
send_delays_.lower_bound(now_ms -
|
|
kSendSideDelayWindowMs));
|
|
}
|
|
if (send_side_delay_observer_ &&
|
|
GetSendSideDelay(&avg_delay_ms, &max_delay_ms)) {
|
|
send_side_delay_observer_->SendSideDelayUpdated(avg_delay_ms,
|
|
max_delay_ms, ssrc);
|
|
}
|
|
}
|
|
|
|
void RTPSender::ProcessBitrate() {
|
|
CriticalSectionScoped cs(send_critsect_.get());
|
|
total_bitrate_sent_.Process();
|
|
nack_bitrate_.Process();
|
|
if (audio_configured_) {
|
|
return;
|
|
}
|
|
video_->ProcessBitrate();
|
|
}
|
|
|
|
size_t RTPSender::RTPHeaderLength() const {
|
|
CriticalSectionScoped lock(send_critsect_.get());
|
|
size_t rtp_header_length = 12;
|
|
rtp_header_length += sizeof(uint32_t) * csrcs_.size();
|
|
rtp_header_length += RtpHeaderExtensionTotalLength();
|
|
return rtp_header_length;
|
|
}
|
|
|
|
uint16_t RTPSender::IncrementSequenceNumber() {
|
|
CriticalSectionScoped cs(send_critsect_.get());
|
|
return sequence_number_++;
|
|
}
|
|
|
|
void RTPSender::ResetDataCounters() {
|
|
uint32_t ssrc;
|
|
uint32_t ssrc_rtx;
|
|
{
|
|
CriticalSectionScoped ssrc_lock(send_critsect_.get());
|
|
ssrc = ssrc_;
|
|
ssrc_rtx = ssrc_rtx_;
|
|
}
|
|
CriticalSectionScoped lock(statistics_crit_.get());
|
|
rtp_stats_ = StreamDataCounters();
|
|
rtx_rtp_stats_ = StreamDataCounters();
|
|
if (rtp_stats_callback_) {
|
|
rtp_stats_callback_->DataCountersUpdated(rtp_stats_, ssrc);
|
|
rtp_stats_callback_->DataCountersUpdated(rtx_rtp_stats_, ssrc_rtx);
|
|
}
|
|
}
|
|
|
|
void RTPSender::GetDataCounters(StreamDataCounters* rtp_stats,
|
|
StreamDataCounters* rtx_stats) const {
|
|
CriticalSectionScoped lock(statistics_crit_.get());
|
|
*rtp_stats = rtp_stats_;
|
|
*rtx_stats = rtx_rtp_stats_;
|
|
}
|
|
|
|
size_t RTPSender::CreateRtpHeader(uint8_t* header,
|
|
int8_t payload_type,
|
|
uint32_t ssrc,
|
|
bool marker_bit,
|
|
uint32_t timestamp,
|
|
uint16_t sequence_number,
|
|
const std::vector<uint32_t>& csrcs) const {
|
|
header[0] = 0x80; // version 2.
|
|
header[1] = static_cast<uint8_t>(payload_type);
|
|
if (marker_bit) {
|
|
header[1] |= kRtpMarkerBitMask; // Marker bit is set.
|
|
}
|
|
RtpUtility::AssignUWord16ToBuffer(header + 2, sequence_number);
|
|
RtpUtility::AssignUWord32ToBuffer(header + 4, timestamp);
|
|
RtpUtility::AssignUWord32ToBuffer(header + 8, ssrc);
|
|
int32_t rtp_header_length = 12;
|
|
|
|
if (csrcs.size() > 0) {
|
|
uint8_t *ptr = &header[rtp_header_length];
|
|
for (size_t i = 0; i < csrcs.size(); ++i) {
|
|
RtpUtility::AssignUWord32ToBuffer(ptr, csrcs[i]);
|
|
ptr += 4;
|
|
}
|
|
header[0] = (header[0] & 0xf0) | csrcs.size();
|
|
|
|
// Update length of header.
|
|
rtp_header_length += sizeof(uint32_t) * csrcs.size();
|
|
}
|
|
|
|
uint16_t len = BuildRTPHeaderExtension(header + rtp_header_length);
|
|
if (len > 0) {
|
|
header[0] |= 0x10; // Set extension bit.
|
|
rtp_header_length += len;
|
|
}
|
|
return rtp_header_length;
|
|
}
|
|
|
|
int32_t RTPSender::BuildRTPheader(uint8_t* data_buffer,
|
|
int8_t payload_type,
|
|
bool marker_bit,
|
|
uint32_t capture_timestamp,
|
|
int64_t capture_time_ms,
|
|
bool timestamp_provided,
|
|
bool inc_sequence_number) {
|
|
assert(payload_type >= 0);
|
|
CriticalSectionScoped cs(send_critsect_.get());
|
|
|
|
if (timestamp_provided) {
|
|
timestamp_ = start_timestamp_ + capture_timestamp;
|
|
} else {
|
|
// Make a unique time stamp.
|
|
// We can't inc by the actual time, since then we increase the risk of back
|
|
// timing.
|
|
timestamp_++;
|
|
}
|
|
last_timestamp_time_ms_ = clock_->TimeInMilliseconds();
|
|
uint32_t sequence_number = sequence_number_++;
|
|
capture_time_ms_ = capture_time_ms;
|
|
last_packet_marker_bit_ = marker_bit;
|
|
return CreateRtpHeader(data_buffer, payload_type, ssrc_, marker_bit,
|
|
timestamp_, sequence_number, csrcs_);
|
|
}
|
|
|
|
uint16_t RTPSender::BuildRTPHeaderExtension(uint8_t* data_buffer) const {
|
|
if (rtp_header_extension_map_.Size() <= 0) {
|
|
return 0;
|
|
}
|
|
// RTP header extension, RFC 3550.
|
|
// 0 1 2 3
|
|
// 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
|
|
// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|
|
// | defined by profile | length |
|
|
// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|
|
// | header extension |
|
|
// | .... |
|
|
//
|
|
const uint32_t kPosLength = 2;
|
|
const uint32_t kHeaderLength = kRtpOneByteHeaderLength;
|
|
|
|
// Add extension ID (0xBEDE).
|
|
RtpUtility::AssignUWord16ToBuffer(data_buffer, kRtpOneByteHeaderExtensionId);
|
|
|
|
// Add extensions.
|
|
uint16_t total_block_length = 0;
|
|
|
|
RTPExtensionType type = rtp_header_extension_map_.First();
|
|
while (type != kRtpExtensionNone) {
|
|
uint8_t block_length = 0;
|
|
switch (type) {
|
|
case kRtpExtensionTransmissionTimeOffset:
|
|
block_length = BuildTransmissionTimeOffsetExtension(
|
|
data_buffer + kHeaderLength + total_block_length);
|
|
break;
|
|
case kRtpExtensionAudioLevel:
|
|
block_length = BuildAudioLevelExtension(
|
|
data_buffer + kHeaderLength + total_block_length);
|
|
break;
|
|
case kRtpExtensionAbsoluteSendTime:
|
|
block_length = BuildAbsoluteSendTimeExtension(
|
|
data_buffer + kHeaderLength + total_block_length);
|
|
break;
|
|
default:
|
|
assert(false);
|
|
}
|
|
total_block_length += block_length;
|
|
type = rtp_header_extension_map_.Next(type);
|
|
}
|
|
if (total_block_length == 0) {
|
|
// No extension added.
|
|
return 0;
|
|
}
|
|
// Set header length (in number of Word32, header excluded).
|
|
assert(total_block_length % 4 == 0);
|
|
RtpUtility::AssignUWord16ToBuffer(data_buffer + kPosLength,
|
|
total_block_length / 4);
|
|
// Total added length.
|
|
return kHeaderLength + total_block_length;
|
|
}
|
|
|
|
uint8_t RTPSender::BuildTransmissionTimeOffsetExtension(
|
|
uint8_t* data_buffer) const {
|
|
// From RFC 5450: Transmission Time Offsets in RTP Streams.
|
|
//
|
|
// The transmission time is signaled to the receiver in-band using the
|
|
// general mechanism for RTP header extensions [RFC5285]. The payload
|
|
// of this extension (the transmitted value) is a 24-bit signed integer.
|
|
// When added to the RTP timestamp of the packet, it represents the
|
|
// "effective" RTP transmission time of the packet, on the RTP
|
|
// timescale.
|
|
//
|
|
// The form of the transmission offset extension block:
|
|
//
|
|
// 0 1 2 3
|
|
// 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
|
|
// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|
|
// | ID | len=2 | transmission offset |
|
|
// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|
|
|
|
// Get id defined by user.
|
|
uint8_t id;
|
|
if (rtp_header_extension_map_.GetId(kRtpExtensionTransmissionTimeOffset,
|
|
&id) != 0) {
|
|
// Not registered.
|
|
return 0;
|
|
}
|
|
size_t pos = 0;
|
|
const uint8_t len = 2;
|
|
data_buffer[pos++] = (id << 4) + len;
|
|
RtpUtility::AssignUWord24ToBuffer(data_buffer + pos,
|
|
transmission_time_offset_);
|
|
pos += 3;
|
|
assert(pos == kTransmissionTimeOffsetLength);
|
|
return kTransmissionTimeOffsetLength;
|
|
}
|
|
|
|
uint8_t RTPSender::BuildAudioLevelExtension(uint8_t* data_buffer) const {
|
|
// An RTP Header Extension for Client-to-Mixer Audio Level Indication
|
|
//
|
|
// https://datatracker.ietf.org/doc/draft-lennox-avt-rtp-audio-level-exthdr/
|
|
//
|
|
// The form of the audio level extension block:
|
|
//
|
|
// 0 1 2 3
|
|
// 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
|
|
// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|
|
// | ID | len=0 |V| level | 0x00 | 0x00 |
|
|
// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|
|
//
|
|
// Note that we always include 2 pad bytes, which will result in legal and
|
|
// correctly parsed RTP, but may be a bit wasteful if more short extensions
|
|
// are implemented. Right now the pad bytes would anyway be required at end
|
|
// of the extension block, so it makes no difference.
|
|
|
|
// Get id defined by user.
|
|
uint8_t id;
|
|
if (rtp_header_extension_map_.GetId(kRtpExtensionAudioLevel, &id) != 0) {
|
|
// Not registered.
|
|
return 0;
|
|
}
|
|
size_t pos = 0;
|
|
const uint8_t len = 0;
|
|
data_buffer[pos++] = (id << 4) + len;
|
|
data_buffer[pos++] = (1 << 7) + 0; // Voice, 0 dBov.
|
|
data_buffer[pos++] = 0; // Padding.
|
|
data_buffer[pos++] = 0; // Padding.
|
|
// kAudioLevelLength is including pad bytes.
|
|
assert(pos == kAudioLevelLength);
|
|
return kAudioLevelLength;
|
|
}
|
|
|
|
uint8_t RTPSender::BuildAbsoluteSendTimeExtension(uint8_t* data_buffer) const {
|
|
// Absolute send time in RTP streams.
|
|
//
|
|
// The absolute send time is signaled to the receiver in-band using the
|
|
// general mechanism for RTP header extensions [RFC5285]. The payload
|
|
// of this extension (the transmitted value) is a 24-bit unsigned integer
|
|
// containing the sender's current time in seconds as a fixed point number
|
|
// with 18 bits fractional part.
|
|
//
|
|
// The form of the absolute send time extension block:
|
|
//
|
|
// 0 1 2 3
|
|
// 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
|
|
// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|
|
// | ID | len=2 | absolute send time |
|
|
// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|
|
|
|
// Get id defined by user.
|
|
uint8_t id;
|
|
if (rtp_header_extension_map_.GetId(kRtpExtensionAbsoluteSendTime,
|
|
&id) != 0) {
|
|
// Not registered.
|
|
return 0;
|
|
}
|
|
size_t pos = 0;
|
|
const uint8_t len = 2;
|
|
data_buffer[pos++] = (id << 4) + len;
|
|
RtpUtility::AssignUWord24ToBuffer(data_buffer + pos, absolute_send_time_);
|
|
pos += 3;
|
|
assert(pos == kAbsoluteSendTimeLength);
|
|
return kAbsoluteSendTimeLength;
|
|
}
|
|
|
|
void RTPSender::UpdateTransmissionTimeOffset(uint8_t* rtp_packet,
|
|
size_t rtp_packet_length,
|
|
const RTPHeader& rtp_header,
|
|
int64_t time_diff_ms) const {
|
|
CriticalSectionScoped cs(send_critsect_.get());
|
|
// Get id.
|
|
uint8_t id = 0;
|
|
if (rtp_header_extension_map_.GetId(kRtpExtensionTransmissionTimeOffset,
|
|
&id) != 0) {
|
|
// Not registered.
|
|
return;
|
|
}
|
|
// Get length until start of header extension block.
|
|
int extension_block_pos =
|
|
rtp_header_extension_map_.GetLengthUntilBlockStartInBytes(
|
|
kRtpExtensionTransmissionTimeOffset);
|
|
if (extension_block_pos < 0) {
|
|
LOG(LS_WARNING)
|
|
<< "Failed to update transmission time offset, not registered.";
|
|
return;
|
|
}
|
|
size_t block_pos = 12 + rtp_header.numCSRCs + extension_block_pos;
|
|
if (rtp_packet_length < block_pos + kTransmissionTimeOffsetLength ||
|
|
rtp_header.headerLength <
|
|
block_pos + kTransmissionTimeOffsetLength) {
|
|
LOG(LS_WARNING)
|
|
<< "Failed to update transmission time offset, invalid length.";
|
|
return;
|
|
}
|
|
// Verify that header contains extension.
|
|
if (!((rtp_packet[12 + rtp_header.numCSRCs] == 0xBE) &&
|
|
(rtp_packet[12 + rtp_header.numCSRCs + 1] == 0xDE))) {
|
|
LOG(LS_WARNING) << "Failed to update transmission time offset, hdr "
|
|
"extension not found.";
|
|
return;
|
|
}
|
|
// Verify first byte in block.
|
|
const uint8_t first_block_byte = (id << 4) + 2;
|
|
if (rtp_packet[block_pos] != first_block_byte) {
|
|
LOG(LS_WARNING) << "Failed to update transmission time offset.";
|
|
return;
|
|
}
|
|
// Update transmission offset field (converting to a 90 kHz timestamp).
|
|
RtpUtility::AssignUWord24ToBuffer(rtp_packet + block_pos + 1,
|
|
time_diff_ms * 90); // RTP timestamp.
|
|
}
|
|
|
|
bool RTPSender::UpdateAudioLevel(uint8_t* rtp_packet,
|
|
size_t rtp_packet_length,
|
|
const RTPHeader& rtp_header,
|
|
bool is_voiced,
|
|
uint8_t dBov) const {
|
|
CriticalSectionScoped cs(send_critsect_.get());
|
|
|
|
// Get id.
|
|
uint8_t id = 0;
|
|
if (rtp_header_extension_map_.GetId(kRtpExtensionAudioLevel, &id) != 0) {
|
|
// Not registered.
|
|
return false;
|
|
}
|
|
// Get length until start of header extension block.
|
|
int extension_block_pos =
|
|
rtp_header_extension_map_.GetLengthUntilBlockStartInBytes(
|
|
kRtpExtensionAudioLevel);
|
|
if (extension_block_pos < 0) {
|
|
// The feature is not enabled.
|
|
return false;
|
|
}
|
|
size_t block_pos = 12 + rtp_header.numCSRCs + extension_block_pos;
|
|
if (rtp_packet_length < block_pos + kAudioLevelLength ||
|
|
rtp_header.headerLength < block_pos + kAudioLevelLength) {
|
|
LOG(LS_WARNING) << "Failed to update audio level, invalid length.";
|
|
return false;
|
|
}
|
|
// Verify that header contains extension.
|
|
if (!((rtp_packet[12 + rtp_header.numCSRCs] == 0xBE) &&
|
|
(rtp_packet[12 + rtp_header.numCSRCs + 1] == 0xDE))) {
|
|
LOG(LS_WARNING) << "Failed to update audio level, hdr extension not found.";
|
|
return false;
|
|
}
|
|
// Verify first byte in block.
|
|
const uint8_t first_block_byte = (id << 4) + 0;
|
|
if (rtp_packet[block_pos] != first_block_byte) {
|
|
LOG(LS_WARNING) << "Failed to update audio level.";
|
|
return false;
|
|
}
|
|
rtp_packet[block_pos + 1] = (is_voiced ? 0x80 : 0x00) + (dBov & 0x7f);
|
|
return true;
|
|
}
|
|
|
|
void RTPSender::UpdateAbsoluteSendTime(uint8_t* rtp_packet,
|
|
size_t rtp_packet_length,
|
|
const RTPHeader& rtp_header,
|
|
int64_t now_ms) const {
|
|
CriticalSectionScoped cs(send_critsect_.get());
|
|
|
|
// Get id.
|
|
uint8_t id = 0;
|
|
if (rtp_header_extension_map_.GetId(kRtpExtensionAbsoluteSendTime,
|
|
&id) != 0) {
|
|
// Not registered.
|
|
return;
|
|
}
|
|
// Get length until start of header extension block.
|
|
int extension_block_pos =
|
|
rtp_header_extension_map_.GetLengthUntilBlockStartInBytes(
|
|
kRtpExtensionAbsoluteSendTime);
|
|
if (extension_block_pos < 0) {
|
|
// The feature is not enabled.
|
|
return;
|
|
}
|
|
size_t block_pos = 12 + rtp_header.numCSRCs + extension_block_pos;
|
|
if (rtp_packet_length < block_pos + kAbsoluteSendTimeLength ||
|
|
rtp_header.headerLength < block_pos + kAbsoluteSendTimeLength) {
|
|
LOG(LS_WARNING) << "Failed to update absolute send time, invalid length.";
|
|
return;
|
|
}
|
|
// Verify that header contains extension.
|
|
if (!((rtp_packet[12 + rtp_header.numCSRCs] == 0xBE) &&
|
|
(rtp_packet[12 + rtp_header.numCSRCs + 1] == 0xDE))) {
|
|
LOG(LS_WARNING)
|
|
<< "Failed to update absolute send time, hdr extension not found.";
|
|
return;
|
|
}
|
|
// Verify first byte in block.
|
|
const uint8_t first_block_byte = (id << 4) + 2;
|
|
if (rtp_packet[block_pos] != first_block_byte) {
|
|
LOG(LS_WARNING) << "Failed to update absolute send time.";
|
|
return;
|
|
}
|
|
// Update absolute send time field (convert ms to 24-bit unsigned with 18 bit
|
|
// fractional part).
|
|
RtpUtility::AssignUWord24ToBuffer(rtp_packet + block_pos + 1,
|
|
((now_ms << 18) / 1000) & 0x00ffffff);
|
|
}
|
|
|
|
void RTPSender::SetSendingStatus(bool enabled) {
|
|
if (enabled) {
|
|
uint32_t frequency_hz = SendPayloadFrequency();
|
|
uint32_t RTPtime = RtpUtility::GetCurrentRTP(clock_, frequency_hz);
|
|
|
|
// Will be ignored if it's already configured via API.
|
|
SetStartTimestamp(RTPtime, false);
|
|
} else {
|
|
CriticalSectionScoped lock(send_critsect_.get());
|
|
if (!ssrc_forced_) {
|
|
// Generate a new SSRC.
|
|
ssrc_db_.ReturnSSRC(ssrc_);
|
|
ssrc_ = ssrc_db_.CreateSSRC(); // Can't be 0.
|
|
bitrates_->set_ssrc(ssrc_);
|
|
}
|
|
// Don't initialize seq number if SSRC passed externally.
|
|
if (!sequence_number_forced_ && !ssrc_forced_) {
|
|
// Generate a new sequence number.
|
|
sequence_number_ =
|
|
rand() / (RAND_MAX / MAX_INIT_RTP_SEQ_NUMBER); // NOLINT
|
|
}
|
|
}
|
|
}
|
|
|
|
void RTPSender::SetSendingMediaStatus(bool enabled) {
|
|
CriticalSectionScoped cs(send_critsect_.get());
|
|
sending_media_ = enabled;
|
|
}
|
|
|
|
bool RTPSender::SendingMedia() const {
|
|
CriticalSectionScoped cs(send_critsect_.get());
|
|
return sending_media_;
|
|
}
|
|
|
|
uint32_t RTPSender::Timestamp() const {
|
|
CriticalSectionScoped cs(send_critsect_.get());
|
|
return timestamp_;
|
|
}
|
|
|
|
void RTPSender::SetStartTimestamp(uint32_t timestamp, bool force) {
|
|
CriticalSectionScoped cs(send_critsect_.get());
|
|
if (force) {
|
|
start_timestamp_forced_ = true;
|
|
start_timestamp_ = timestamp;
|
|
} else {
|
|
if (!start_timestamp_forced_) {
|
|
start_timestamp_ = timestamp;
|
|
}
|
|
}
|
|
}
|
|
|
|
uint32_t RTPSender::StartTimestamp() const {
|
|
CriticalSectionScoped cs(send_critsect_.get());
|
|
return start_timestamp_;
|
|
}
|
|
|
|
uint32_t RTPSender::GenerateNewSSRC() {
|
|
// If configured via API, return 0.
|
|
CriticalSectionScoped cs(send_critsect_.get());
|
|
|
|
if (ssrc_forced_) {
|
|
return 0;
|
|
}
|
|
ssrc_ = ssrc_db_.CreateSSRC(); // Can't be 0.
|
|
bitrates_->set_ssrc(ssrc_);
|
|
return ssrc_;
|
|
}
|
|
|
|
void RTPSender::SetSSRC(uint32_t ssrc) {
|
|
// This is configured via the API.
|
|
CriticalSectionScoped cs(send_critsect_.get());
|
|
|
|
if (ssrc_ == ssrc && ssrc_forced_) {
|
|
return; // Since it's same ssrc, don't reset anything.
|
|
}
|
|
ssrc_forced_ = true;
|
|
ssrc_db_.ReturnSSRC(ssrc_);
|
|
ssrc_db_.RegisterSSRC(ssrc);
|
|
ssrc_ = ssrc;
|
|
bitrates_->set_ssrc(ssrc_);
|
|
if (!sequence_number_forced_) {
|
|
sequence_number_ =
|
|
rand() / (RAND_MAX / MAX_INIT_RTP_SEQ_NUMBER); // NOLINT
|
|
}
|
|
}
|
|
|
|
uint32_t RTPSender::SSRC() const {
|
|
CriticalSectionScoped cs(send_critsect_.get());
|
|
return ssrc_;
|
|
}
|
|
|
|
void RTPSender::SetCsrcs(const std::vector<uint32_t>& csrcs) {
|
|
assert(csrcs.size() <= kRtpCsrcSize);
|
|
CriticalSectionScoped cs(send_critsect_.get());
|
|
csrcs_ = csrcs;
|
|
}
|
|
|
|
void RTPSender::SetSequenceNumber(uint16_t seq) {
|
|
CriticalSectionScoped cs(send_critsect_.get());
|
|
sequence_number_forced_ = true;
|
|
sequence_number_ = seq;
|
|
}
|
|
|
|
uint16_t RTPSender::SequenceNumber() const {
|
|
CriticalSectionScoped cs(send_critsect_.get());
|
|
return sequence_number_;
|
|
}
|
|
|
|
// Audio.
|
|
int32_t RTPSender::SendTelephoneEvent(uint8_t key,
|
|
uint16_t time_ms,
|
|
uint8_t level) {
|
|
if (!audio_configured_) {
|
|
return -1;
|
|
}
|
|
return audio_->SendTelephoneEvent(key, time_ms, level);
|
|
}
|
|
|
|
int32_t RTPSender::SetAudioPacketSize(uint16_t packet_size_samples) {
|
|
if (!audio_configured_) {
|
|
return -1;
|
|
}
|
|
return audio_->SetAudioPacketSize(packet_size_samples);
|
|
}
|
|
|
|
int32_t RTPSender::SetAudioLevel(uint8_t level_d_bov) {
|
|
return audio_->SetAudioLevel(level_d_bov);
|
|
}
|
|
|
|
int32_t RTPSender::SetRED(int8_t payload_type) {
|
|
if (!audio_configured_) {
|
|
return -1;
|
|
}
|
|
return audio_->SetRED(payload_type);
|
|
}
|
|
|
|
int32_t RTPSender::RED(int8_t *payload_type) const {
|
|
if (!audio_configured_) {
|
|
return -1;
|
|
}
|
|
return audio_->RED(*payload_type);
|
|
}
|
|
|
|
// Video
|
|
VideoCodecInformation *RTPSender::CodecInformationVideo() {
|
|
if (audio_configured_) {
|
|
return NULL;
|
|
}
|
|
return video_->CodecInformationVideo();
|
|
}
|
|
|
|
RtpVideoCodecTypes RTPSender::VideoCodecType() const {
|
|
assert(!audio_configured_ && "Sender is an audio stream!");
|
|
return video_->VideoCodecType();
|
|
}
|
|
|
|
uint32_t RTPSender::MaxConfiguredBitrateVideo() const {
|
|
if (audio_configured_) {
|
|
return 0;
|
|
}
|
|
return video_->MaxConfiguredBitrateVideo();
|
|
}
|
|
|
|
int32_t RTPSender::SendRTPIntraRequest() {
|
|
if (audio_configured_) {
|
|
return -1;
|
|
}
|
|
return video_->SendRTPIntraRequest();
|
|
}
|
|
|
|
int32_t RTPSender::SetGenericFECStatus(bool enable,
|
|
uint8_t payload_type_red,
|
|
uint8_t payload_type_fec) {
|
|
if (audio_configured_) {
|
|
return -1;
|
|
}
|
|
return video_->SetGenericFECStatus(enable, payload_type_red,
|
|
payload_type_fec);
|
|
}
|
|
|
|
int32_t RTPSender::GenericFECStatus(
|
|
bool *enable, uint8_t *payload_type_red,
|
|
uint8_t *payload_type_fec) const {
|
|
if (audio_configured_) {
|
|
return -1;
|
|
}
|
|
return video_->GenericFECStatus(
|
|
*enable, *payload_type_red, *payload_type_fec);
|
|
}
|
|
|
|
int32_t RTPSender::SetFecParameters(
|
|
const FecProtectionParams *delta_params,
|
|
const FecProtectionParams *key_params) {
|
|
if (audio_configured_) {
|
|
return -1;
|
|
}
|
|
return video_->SetFecParameters(delta_params, key_params);
|
|
}
|
|
|
|
void RTPSender::BuildRtxPacket(uint8_t* buffer, size_t* length,
|
|
uint8_t* buffer_rtx) {
|
|
CriticalSectionScoped cs(send_critsect_.get());
|
|
uint8_t* data_buffer_rtx = buffer_rtx;
|
|
// Add RTX header.
|
|
RtpUtility::RtpHeaderParser rtp_parser(
|
|
reinterpret_cast<const uint8_t*>(buffer), *length);
|
|
|
|
RTPHeader rtp_header;
|
|
rtp_parser.Parse(rtp_header);
|
|
|
|
// Add original RTP header.
|
|
memcpy(data_buffer_rtx, buffer, rtp_header.headerLength);
|
|
|
|
// Replace payload type, if a specific type is set for RTX.
|
|
if (payload_type_rtx_ != -1) {
|
|
data_buffer_rtx[1] = static_cast<uint8_t>(payload_type_rtx_);
|
|
if (rtp_header.markerBit)
|
|
data_buffer_rtx[1] |= kRtpMarkerBitMask;
|
|
}
|
|
|
|
// Replace sequence number.
|
|
uint8_t *ptr = data_buffer_rtx + 2;
|
|
RtpUtility::AssignUWord16ToBuffer(ptr, sequence_number_rtx_++);
|
|
|
|
// Replace SSRC.
|
|
ptr += 6;
|
|
RtpUtility::AssignUWord32ToBuffer(ptr, ssrc_rtx_);
|
|
|
|
// Add OSN (original sequence number).
|
|
ptr = data_buffer_rtx + rtp_header.headerLength;
|
|
RtpUtility::AssignUWord16ToBuffer(ptr, rtp_header.sequenceNumber);
|
|
ptr += 2;
|
|
|
|
// Add original payload data.
|
|
memcpy(ptr, buffer + rtp_header.headerLength,
|
|
*length - rtp_header.headerLength);
|
|
*length += 2;
|
|
}
|
|
|
|
void RTPSender::RegisterRtpStatisticsCallback(
|
|
StreamDataCountersCallback* callback) {
|
|
CriticalSectionScoped cs(statistics_crit_.get());
|
|
rtp_stats_callback_ = callback;
|
|
}
|
|
|
|
StreamDataCountersCallback* RTPSender::GetRtpStatisticsCallback() const {
|
|
CriticalSectionScoped cs(statistics_crit_.get());
|
|
return rtp_stats_callback_;
|
|
}
|
|
|
|
uint32_t RTPSender::BitrateSent() const {
|
|
return total_bitrate_sent_.BitrateLast();
|
|
}
|
|
|
|
void RTPSender::SetRtpState(const RtpState& rtp_state) {
|
|
SetStartTimestamp(rtp_state.start_timestamp, true);
|
|
CriticalSectionScoped lock(send_critsect_.get());
|
|
sequence_number_ = rtp_state.sequence_number;
|
|
sequence_number_forced_ = true;
|
|
timestamp_ = rtp_state.timestamp;
|
|
capture_time_ms_ = rtp_state.capture_time_ms;
|
|
last_timestamp_time_ms_ = rtp_state.last_timestamp_time_ms;
|
|
media_has_been_sent_ = rtp_state.media_has_been_sent;
|
|
}
|
|
|
|
RtpState RTPSender::GetRtpState() const {
|
|
CriticalSectionScoped lock(send_critsect_.get());
|
|
|
|
RtpState state;
|
|
state.sequence_number = sequence_number_;
|
|
state.start_timestamp = start_timestamp_;
|
|
state.timestamp = timestamp_;
|
|
state.capture_time_ms = capture_time_ms_;
|
|
state.last_timestamp_time_ms = last_timestamp_time_ms_;
|
|
state.media_has_been_sent = media_has_been_sent_;
|
|
|
|
return state;
|
|
}
|
|
|
|
void RTPSender::SetRtxRtpState(const RtpState& rtp_state) {
|
|
CriticalSectionScoped lock(send_critsect_.get());
|
|
sequence_number_rtx_ = rtp_state.sequence_number;
|
|
}
|
|
|
|
RtpState RTPSender::GetRtxRtpState() const {
|
|
CriticalSectionScoped lock(send_critsect_.get());
|
|
|
|
RtpState state;
|
|
state.sequence_number = sequence_number_rtx_;
|
|
state.start_timestamp = start_timestamp_;
|
|
|
|
return state;
|
|
}
|
|
|
|
} // namespace webrtc
|