
The main purpose was the interface-> include rename, but other files were also moved, eliminating the "main" dir. To avoid breaking downstream, the "interface" directories were copied into a new "video_coding/include" dir. The old headers got pragma warnings added about deprecation (a very short deprecation since I plan to remove them as soon downstream is updated). Other files also moved: video_coding/main/source -> video_coding video_coding/main/test -> video_coding/test BUG=webrtc:5095 TESTED=Passing compile-trybots with --clobber flag: git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel --bot=linux_gn_rel --bot=win_x64_gn_rel --bot=mac_x64_gn_rel --bot=android_gn_rel -m tryserver.webrtc R=stefan@webrtc.org, tommi@webrtc.org Review URL: https://codereview.webrtc.org/1417283007 . Cr-Commit-Position: refs/heads/master@{#10694}
494 lines
15 KiB
C++
494 lines
15 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/video_coding/test/rtp_player.h"
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#include <stdio.h>
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#include <map>
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#include "webrtc/base/scoped_ptr.h"
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#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
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#include "webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h"
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#include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h"
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#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
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#include "webrtc/modules/video_coding/internal_defines.h"
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#include "webrtc/modules/video_coding/test/test_util.h"
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#include "webrtc/system_wrappers/include/clock.h"
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#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
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#include "webrtc/test/rtp_file_reader.h"
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#if 1
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# define DEBUG_LOG1(text, arg)
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#else
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# define DEBUG_LOG1(text, arg) (printf(text "\n", arg))
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#endif
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namespace webrtc {
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namespace rtpplayer {
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enum {
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kMaxPacketBufferSize = 4096,
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kDefaultTransmissionTimeOffsetExtensionId = 2
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};
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class RawRtpPacket {
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public:
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RawRtpPacket(const uint8_t* data, size_t length, uint32_t ssrc,
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uint16_t seq_num)
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: data_(new uint8_t[length]),
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length_(length),
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resend_time_ms_(-1),
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ssrc_(ssrc),
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seq_num_(seq_num) {
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assert(data);
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memcpy(data_.get(), data, length_);
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}
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const uint8_t* data() const { return data_.get(); }
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size_t length() const { return length_; }
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int64_t resend_time_ms() const { return resend_time_ms_; }
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void set_resend_time_ms(int64_t timeMs) { resend_time_ms_ = timeMs; }
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uint32_t ssrc() const { return ssrc_; }
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uint16_t seq_num() const { return seq_num_; }
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private:
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rtc::scoped_ptr<uint8_t[]> data_;
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size_t length_;
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int64_t resend_time_ms_;
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uint32_t ssrc_;
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uint16_t seq_num_;
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RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(RawRtpPacket);
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};
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class LostPackets {
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public:
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LostPackets(Clock* clock, int64_t rtt_ms)
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: crit_sect_(CriticalSectionWrapper::CreateCriticalSection()),
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debug_file_(fopen("PacketLossDebug.txt", "w")),
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loss_count_(0),
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packets_(),
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clock_(clock),
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rtt_ms_(rtt_ms) {
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assert(clock);
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}
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~LostPackets() {
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if (debug_file_) {
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fclose(debug_file_);
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debug_file_ = NULL;
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}
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while (!packets_.empty()) {
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delete packets_.back();
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packets_.pop_back();
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}
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}
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void AddPacket(RawRtpPacket* packet) {
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assert(packet);
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printf("Throw: %08x:%u\n", packet->ssrc(), packet->seq_num());
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CriticalSectionScoped cs(crit_sect_.get());
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if (debug_file_) {
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fprintf(debug_file_, "%u Lost packet: %u\n", loss_count_,
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packet->seq_num());
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}
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packets_.push_back(packet);
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loss_count_++;
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}
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void SetResendTime(uint32_t ssrc, int16_t resendSeqNum) {
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int64_t resend_time_ms = clock_->TimeInMilliseconds() + rtt_ms_;
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int64_t now_ms = clock_->TimeInMilliseconds();
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CriticalSectionScoped cs(crit_sect_.get());
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for (RtpPacketIterator it = packets_.begin(); it != packets_.end(); ++it) {
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RawRtpPacket* packet = *it;
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if (ssrc == packet->ssrc() && resendSeqNum == packet->seq_num() &&
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packet->resend_time_ms() + 10 < now_ms) {
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if (debug_file_) {
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fprintf(debug_file_, "Resend %u at %u\n", packet->seq_num(),
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MaskWord64ToUWord32(resend_time_ms));
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}
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packet->set_resend_time_ms(resend_time_ms);
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return;
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}
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}
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// We may get here since the captured stream may itself be missing packets.
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}
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RawRtpPacket* NextPacketToResend(int64_t time_now) {
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CriticalSectionScoped cs(crit_sect_.get());
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for (RtpPacketIterator it = packets_.begin(); it != packets_.end(); ++it) {
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RawRtpPacket* packet = *it;
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if (time_now >= packet->resend_time_ms() &&
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packet->resend_time_ms() != -1) {
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packets_.erase(it);
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return packet;
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}
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}
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return NULL;
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}
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int NumberOfPacketsToResend() const {
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CriticalSectionScoped cs(crit_sect_.get());
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int count = 0;
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for (ConstRtpPacketIterator it = packets_.begin(); it != packets_.end();
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++it) {
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if ((*it)->resend_time_ms() >= 0) {
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count++;
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}
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}
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return count;
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}
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void LogPacketResent(RawRtpPacket* packet) {
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int64_t now_ms = clock_->TimeInMilliseconds();
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CriticalSectionScoped cs(crit_sect_.get());
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if (debug_file_) {
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fprintf(debug_file_, "Resent %u at %u\n", packet->seq_num(),
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MaskWord64ToUWord32(now_ms));
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}
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}
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void Print() const {
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CriticalSectionScoped cs(crit_sect_.get());
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printf("Lost packets: %u\n", loss_count_);
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printf("Packets waiting to be resent: %d\n", NumberOfPacketsToResend());
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printf("Packets still lost: %zd\n", packets_.size());
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printf("Sequence numbers:\n");
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for (ConstRtpPacketIterator it = packets_.begin(); it != packets_.end();
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++it) {
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printf("%u, ", (*it)->seq_num());
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}
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printf("\n");
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}
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private:
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typedef std::vector<RawRtpPacket*> RtpPacketList;
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typedef RtpPacketList::iterator RtpPacketIterator;
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typedef RtpPacketList::const_iterator ConstRtpPacketIterator;
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rtc::scoped_ptr<CriticalSectionWrapper> crit_sect_;
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FILE* debug_file_;
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int loss_count_;
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RtpPacketList packets_;
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Clock* clock_;
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int64_t rtt_ms_;
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RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(LostPackets);
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};
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class SsrcHandlers {
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public:
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SsrcHandlers(PayloadSinkFactoryInterface* payload_sink_factory,
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const PayloadTypes& payload_types)
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: payload_sink_factory_(payload_sink_factory),
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payload_types_(payload_types),
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handlers_() {
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assert(payload_sink_factory);
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}
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~SsrcHandlers() {
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while (!handlers_.empty()) {
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delete handlers_.begin()->second;
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handlers_.erase(handlers_.begin());
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}
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}
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int RegisterSsrc(uint32_t ssrc, LostPackets* lost_packets, Clock* clock) {
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if (handlers_.count(ssrc) > 0) {
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return 0;
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}
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DEBUG_LOG1("Registering handler for ssrc=%08x", ssrc);
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rtc::scoped_ptr<Handler> handler(
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new Handler(ssrc, payload_types_, lost_packets));
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handler->payload_sink_.reset(payload_sink_factory_->Create(handler.get()));
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if (handler->payload_sink_.get() == NULL) {
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return -1;
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}
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RtpRtcp::Configuration configuration;
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configuration.clock = clock;
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configuration.audio = false;
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handler->rtp_module_.reset(RtpReceiver::CreateVideoReceiver(
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configuration.clock, handler->payload_sink_.get(), NULL,
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handler->rtp_payload_registry_.get()));
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if (handler->rtp_module_.get() == NULL) {
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return -1;
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}
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handler->rtp_module_->SetNACKStatus(kNackOff);
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handler->rtp_header_parser_->RegisterRtpHeaderExtension(
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kRtpExtensionTransmissionTimeOffset,
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kDefaultTransmissionTimeOffsetExtensionId);
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for (PayloadTypesIterator it = payload_types_.begin();
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it != payload_types_.end(); ++it) {
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VideoCodec codec;
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memset(&codec, 0, sizeof(codec));
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strncpy(codec.plName, it->name().c_str(), sizeof(codec.plName)-1);
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codec.plType = it->payload_type();
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codec.codecType = it->codec_type();
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if (handler->rtp_module_->RegisterReceivePayload(codec.plName,
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codec.plType,
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90000,
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0,
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codec.maxBitrate) < 0) {
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return -1;
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}
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}
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handlers_[ssrc] = handler.release();
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return 0;
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}
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void IncomingPacket(const uint8_t* data, size_t length) {
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for (HandlerMapIt it = handlers_.begin(); it != handlers_.end(); ++it) {
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if (!it->second->rtp_header_parser_->IsRtcp(data, length)) {
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RTPHeader header;
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it->second->rtp_header_parser_->Parse(data, length, &header);
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PayloadUnion payload_specific;
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it->second->rtp_payload_registry_->GetPayloadSpecifics(
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header.payloadType, &payload_specific);
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it->second->rtp_module_->IncomingRtpPacket(header, data, length,
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payload_specific, true);
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}
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}
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}
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private:
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class Handler : public RtpStreamInterface {
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public:
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Handler(uint32_t ssrc, const PayloadTypes& payload_types,
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LostPackets* lost_packets)
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: rtp_header_parser_(RtpHeaderParser::Create()),
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rtp_payload_registry_(new RTPPayloadRegistry(
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RTPPayloadStrategy::CreateStrategy(false))),
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rtp_module_(),
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payload_sink_(),
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ssrc_(ssrc),
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payload_types_(payload_types),
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lost_packets_(lost_packets) {
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assert(lost_packets);
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}
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virtual ~Handler() {}
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virtual void ResendPackets(const uint16_t* sequence_numbers,
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uint16_t length) {
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assert(sequence_numbers);
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for (uint16_t i = 0; i < length; i++) {
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lost_packets_->SetResendTime(ssrc_, sequence_numbers[i]);
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}
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}
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virtual uint32_t ssrc() const { return ssrc_; }
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virtual const PayloadTypes& payload_types() const {
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return payload_types_;
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}
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rtc::scoped_ptr<RtpHeaderParser> rtp_header_parser_;
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rtc::scoped_ptr<RTPPayloadRegistry> rtp_payload_registry_;
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rtc::scoped_ptr<RtpReceiver> rtp_module_;
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rtc::scoped_ptr<PayloadSinkInterface> payload_sink_;
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private:
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uint32_t ssrc_;
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const PayloadTypes& payload_types_;
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LostPackets* lost_packets_;
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RTC_DISALLOW_COPY_AND_ASSIGN(Handler);
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};
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typedef std::map<uint32_t, Handler*> HandlerMap;
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typedef std::map<uint32_t, Handler*>::iterator HandlerMapIt;
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PayloadSinkFactoryInterface* payload_sink_factory_;
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PayloadTypes payload_types_;
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HandlerMap handlers_;
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RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(SsrcHandlers);
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};
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class RtpPlayerImpl : public RtpPlayerInterface {
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public:
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RtpPlayerImpl(PayloadSinkFactoryInterface* payload_sink_factory,
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const PayloadTypes& payload_types,
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Clock* clock,
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rtc::scoped_ptr<test::RtpFileReader>* packet_source,
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float loss_rate,
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int64_t rtt_ms,
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bool reordering)
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: ssrc_handlers_(payload_sink_factory, payload_types),
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clock_(clock),
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next_rtp_time_(0),
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first_packet_(true),
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first_packet_rtp_time_(0),
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first_packet_time_ms_(0),
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loss_rate_(loss_rate),
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lost_packets_(clock, rtt_ms),
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resend_packet_count_(0),
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no_loss_startup_(100),
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end_of_file_(false),
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reordering_(false),
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reorder_buffer_() {
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assert(clock);
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assert(packet_source);
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assert(packet_source->get());
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packet_source_.swap(*packet_source);
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srand(321);
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}
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virtual ~RtpPlayerImpl() {}
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virtual int NextPacket(int64_t time_now) {
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// Send any packets ready to be resent.
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for (RawRtpPacket* packet = lost_packets_.NextPacketToResend(time_now);
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packet != NULL;
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packet = lost_packets_.NextPacketToResend(time_now)) {
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int ret = SendPacket(packet->data(), packet->length());
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if (ret > 0) {
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printf("Resend: %08x:%u\n", packet->ssrc(), packet->seq_num());
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lost_packets_.LogPacketResent(packet);
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resend_packet_count_++;
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}
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delete packet;
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if (ret < 0) {
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return ret;
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}
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}
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// Send any packets from packet source.
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if (!end_of_file_ && (TimeUntilNextPacket() == 0 || first_packet_)) {
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if (first_packet_) {
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if (!packet_source_->NextPacket(&next_packet_))
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return 0;
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first_packet_rtp_time_ = next_packet_.time_ms;
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first_packet_time_ms_ = clock_->TimeInMilliseconds();
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first_packet_ = false;
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}
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if (reordering_ && reorder_buffer_.get() == NULL) {
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reorder_buffer_.reset(
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new RawRtpPacket(next_packet_.data, next_packet_.length, 0, 0));
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return 0;
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}
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int ret = SendPacket(next_packet_.data, next_packet_.length);
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if (reorder_buffer_.get()) {
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SendPacket(reorder_buffer_->data(), reorder_buffer_->length());
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reorder_buffer_.reset(NULL);
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}
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if (ret < 0) {
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return ret;
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}
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if (!packet_source_->NextPacket(&next_packet_)) {
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end_of_file_ = true;
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return 0;
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}
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else if (next_packet_.length == 0) {
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return 0;
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}
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}
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if (end_of_file_ && lost_packets_.NumberOfPacketsToResend() == 0) {
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return 1;
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}
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return 0;
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}
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virtual uint32_t TimeUntilNextPacket() const {
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int64_t time_left = (next_rtp_time_ - first_packet_rtp_time_) -
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(clock_->TimeInMilliseconds() - first_packet_time_ms_);
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if (time_left < 0) {
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return 0;
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}
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return static_cast<uint32_t>(time_left);
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}
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virtual void Print() const {
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printf("Resent packets: %u\n", resend_packet_count_);
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lost_packets_.Print();
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}
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private:
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int SendPacket(const uint8_t* data, size_t length) {
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assert(data);
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assert(length > 0);
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rtc::scoped_ptr<RtpHeaderParser> rtp_header_parser(
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RtpHeaderParser::Create());
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if (!rtp_header_parser->IsRtcp(data, length)) {
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RTPHeader header;
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if (!rtp_header_parser->Parse(data, length, &header)) {
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return -1;
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}
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uint32_t ssrc = header.ssrc;
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if (ssrc_handlers_.RegisterSsrc(ssrc, &lost_packets_, clock_) < 0) {
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DEBUG_LOG1("Unable to register ssrc: %d", ssrc);
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return -1;
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}
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if (no_loss_startup_ > 0) {
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no_loss_startup_--;
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} else if ((rand() + 1.0)/(RAND_MAX + 1.0) < loss_rate_) {
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uint16_t seq_num = header.sequenceNumber;
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lost_packets_.AddPacket(new RawRtpPacket(data, length, ssrc, seq_num));
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DEBUG_LOG1("Dropped packet: %d!", header.header.sequenceNumber);
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return 0;
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}
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}
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ssrc_handlers_.IncomingPacket(data, length);
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return 1;
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}
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SsrcHandlers ssrc_handlers_;
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Clock* clock_;
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rtc::scoped_ptr<test::RtpFileReader> packet_source_;
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test::RtpPacket next_packet_;
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uint32_t next_rtp_time_;
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bool first_packet_;
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int64_t first_packet_rtp_time_;
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int64_t first_packet_time_ms_;
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float loss_rate_;
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LostPackets lost_packets_;
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uint32_t resend_packet_count_;
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uint32_t no_loss_startup_;
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bool end_of_file_;
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bool reordering_;
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rtc::scoped_ptr<RawRtpPacket> reorder_buffer_;
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RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(RtpPlayerImpl);
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};
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RtpPlayerInterface* Create(const std::string& input_filename,
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PayloadSinkFactoryInterface* payload_sink_factory, Clock* clock,
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const PayloadTypes& payload_types, float loss_rate, int64_t rtt_ms,
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bool reordering) {
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rtc::scoped_ptr<test::RtpFileReader> packet_source(
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test::RtpFileReader::Create(test::RtpFileReader::kRtpDump,
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input_filename));
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if (packet_source.get() == NULL) {
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packet_source.reset(test::RtpFileReader::Create(test::RtpFileReader::kPcap,
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input_filename));
|
|
if (packet_source.get() == NULL) {
|
|
return NULL;
|
|
}
|
|
}
|
|
|
|
rtc::scoped_ptr<RtpPlayerImpl> impl(
|
|
new RtpPlayerImpl(payload_sink_factory, payload_types, clock,
|
|
&packet_source, loss_rate, rtt_ms, reordering));
|
|
return impl.release();
|
|
}
|
|
} // namespace rtpplayer
|
|
} // namespace webrtc
|