Generate random rtp packets with RtpPacketToSend instead of RtpSender
in rtc event log unittest R=stefan@webrtc.org, terelius@webrtc.org Review URL: https://codereview.webrtc.org/2228493002 . Cr-Commit-Position: refs/heads/master@{#13705}
This commit is contained in:
Danil Chapovalov
parent
116fd61599
commit
bcdad0f3a8
@ -20,14 +20,15 @@
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#include "webrtc/base/buffer.h"
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#include "webrtc/base/checks.h"
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#include "webrtc/base/random.h"
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#include "webrtc/base/rate_limiter.h"
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#include "webrtc/call.h"
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#include "webrtc/call/rtc_event_log.h"
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#include "webrtc/call/rtc_event_log_parser.h"
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#include "webrtc/call/rtc_event_log_unittest_helper.h"
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#include "webrtc/modules/rtp_rtcp/source/rtcp_packet.h"
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#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/sender_report.h"
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#include "webrtc/modules/rtp_rtcp/source/rtp_sender.h"
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#include "webrtc/modules/rtp_rtcp/source/rtp_header_extension.h"
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#include "webrtc/modules/rtp_rtcp/source/rtp_header_extensions.h"
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#include "webrtc/modules/rtp_rtcp/source/rtp_packet_to_send.h"
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#include "webrtc/system_wrappers/include/clock.h"
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#include "webrtc/test/test_suite.h"
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#include "webrtc/test/testsupport/fileutils.h"
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@ -105,55 +106,37 @@ void PrintExpectedEvents(size_t rtp_count,
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* presence of extension number i from kExtensionTypes / kExtensionNames.
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* The least significant bit extension_bitvector has number 0.
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*/
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size_t GenerateRtpPacket(uint32_t extensions_bitvector,
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uint32_t csrcs_count,
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uint8_t* packet,
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size_t packet_size,
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Random* prng) {
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RtpPacketToSend GenerateRtpPacket(const RtpHeaderExtensionMap* extensions,
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uint32_t csrcs_count,
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size_t packet_size,
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Random* prng) {
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RTC_CHECK_GE(packet_size, 16 + 4 * csrcs_count + 4 * kNumExtensions);
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Clock* clock = Clock::GetRealTimeClock();
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RateLimiter retranmission_rate_limiter(clock, 1000);
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RTPSender rtp_sender(false, // bool audio
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clock, // Clock* clock
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nullptr, // Transport*
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nullptr, // PacedSender*
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nullptr, // PacketRouter*
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nullptr, // SendTimeObserver*
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nullptr, // BitrateStatisticsObserver*
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nullptr, // FrameCountObserver*
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nullptr, // SendSideDelayObserver*
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nullptr, // RtcEventLog*
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nullptr, // SendPacketObserver*
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&retranmission_rate_limiter);
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std::vector<uint32_t> csrcs;
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for (unsigned i = 0; i < csrcs_count; i++) {
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csrcs.push_back(prng->Rand<uint32_t>());
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}
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rtp_sender.SetCsrcs(csrcs);
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rtp_sender.SetSSRC(prng->Rand<uint32_t>());
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rtp_sender.SetStartTimestamp(prng->Rand<uint32_t>(), true);
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rtp_sender.SetSequenceNumber(prng->Rand<uint16_t>());
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for (unsigned i = 0; i < kNumExtensions; i++) {
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if (extensions_bitvector & (1u << i)) {
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rtp_sender.RegisterRtpHeaderExtension(kExtensionTypes[i], i + 1);
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}
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RtpPacketToSend rtp_packet(extensions, packet_size);
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rtp_packet.SetPayloadType(prng->Rand(127));
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rtp_packet.SetMarker(prng->Rand<bool>());
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rtp_packet.SetSequenceNumber(prng->Rand<uint16_t>());
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rtp_packet.SetSsrc(prng->Rand<uint32_t>());
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rtp_packet.SetTimestamp(prng->Rand<uint32_t>());
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rtp_packet.SetCsrcs(csrcs);
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rtp_packet.SetExtension<TransmissionOffset>(prng->Rand(0x00ffffff));
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rtp_packet.SetExtension<AudioLevel>(prng->Rand<bool>(), prng->Rand(127));
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rtp_packet.SetExtension<AbsoluteSendTime>(prng->Rand<int32_t>());
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rtp_packet.SetExtension<VideoOrientation>(prng->Rand(2));
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rtp_packet.SetExtension<TransportSequenceNumber>(prng->Rand<uint16_t>());
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size_t payload_size = packet_size - rtp_packet.headers_size();
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uint8_t* payload = rtp_packet.AllocatePayload(payload_size);
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for (size_t i = 0; i < payload_size; i++) {
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payload[i] = prng->Rand<uint8_t>();
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}
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int8_t payload_type = prng->Rand(0, 127);
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bool marker_bit = prng->Rand<bool>();
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uint32_t capture_timestamp = prng->Rand<uint32_t>();
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int64_t capture_time_ms = prng->Rand<uint32_t>();
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size_t header_size = rtp_sender.BuildRtpHeader(
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packet, payload_type, marker_bit, capture_timestamp, capture_time_ms);
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for (size_t i = header_size; i < packet_size; i++) {
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packet[i] = prng->Rand<uint8_t>();
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}
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return header_size;
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return rtp_packet;
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}
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rtc::Buffer GenerateRtcpPacket(Random* prng) {
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@ -232,9 +215,8 @@ void LogSessionAndReadBack(size_t rtp_count,
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ASSERT_LE(rtcp_count, rtp_count);
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ASSERT_LE(playout_count, rtp_count);
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ASSERT_LE(bwe_loss_count, rtp_count);
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std::vector<rtc::Buffer> rtp_packets;
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std::vector<RtpPacketToSend> rtp_packets;
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std::vector<rtc::Buffer> rtcp_packets;
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std::vector<size_t> rtp_header_sizes;
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std::vector<uint32_t> playout_ssrcs;
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std::vector<std::pair<int32_t, uint8_t> > bwe_loss_updates;
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@ -243,14 +225,18 @@ void LogSessionAndReadBack(size_t rtp_count,
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Random prng(random_seed);
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// Initialize rtp header extensions to be used in generated rtp packets.
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RtpHeaderExtensionMap extensions;
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for (unsigned i = 0; i < kNumExtensions; i++) {
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if (extensions_bitvector & (1u << i)) {
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extensions.Register(kExtensionTypes[i], i + 1);
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}
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}
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// Create rtp_count RTP packets containing random data.
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for (size_t i = 0; i < rtp_count; i++) {
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size_t packet_size = prng.Rand(1000, 1100);
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rtp_packets.push_back(rtc::Buffer(packet_size));
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size_t header_size =
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GenerateRtpPacket(extensions_bitvector, csrcs_count,
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rtp_packets[i].data(), packet_size, &prng);
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rtp_header_sizes.push_back(header_size);
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rtp_packets.push_back(
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GenerateRtpPacket(&extensions, csrcs_count, packet_size, &prng));
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}
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// Create rtcp_count RTCP packets containing random data.
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for (size_t i = 0; i < rtcp_count; i++) {
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@ -353,7 +339,7 @@ void LogSessionAndReadBack(size_t rtp_count,
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parsed_log, event_index,
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(i % 2 == 0) ? kIncomingPacket : kOutgoingPacket,
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(i % 3 == 0) ? MediaType::AUDIO : MediaType::VIDEO,
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rtp_packets[i - 1].data(), rtp_header_sizes[i - 1],
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rtp_packets[i - 1].data(), rtp_packets[i - 1].headers_size(),
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rtp_packets[i - 1].size());
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event_index++;
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if (i * rtcp_count >= rtcp_index * rtp_count) {
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@ -423,9 +409,8 @@ TEST(RtcEventLogTest, LogEventAndReadBack) {
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// Create one RTP and one RTCP packet containing random data.
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size_t packet_size = prng.Rand(1000, 1100);
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rtc::Buffer rtp_packet(packet_size);
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size_t header_size =
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GenerateRtpPacket(0, 0, rtp_packet.data(), packet_size, &prng);
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RtpPacketToSend rtp_packet =
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GenerateRtpPacket(nullptr, 0, packet_size, &prng);
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rtc::Buffer rtcp_packet = GenerateRtcpPacket(&prng);
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// Find the name of the current test, in order to use it as a temporary
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@ -461,9 +446,9 @@ TEST(RtcEventLogTest, LogEventAndReadBack) {
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RtcEventLogTestHelper::VerifyLogStartEvent(parsed_log, 0);
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RtcEventLogTestHelper::VerifyRtpEvent(parsed_log, 1, kIncomingPacket,
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MediaType::VIDEO, rtp_packet.data(),
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header_size, rtp_packet.size());
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RtcEventLogTestHelper::VerifyRtpEvent(
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parsed_log, 1, kIncomingPacket, MediaType::VIDEO, rtp_packet.data(),
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rtp_packet.headers_size(), rtp_packet.size());
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RtcEventLogTestHelper::VerifyRtcpEvent(parsed_log, 2, kOutgoingPacket,
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MediaType::VIDEO, rtcp_packet.data(),
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