Index: webrtc/call/rtc_event_log_unittest.cc |
diff --git a/webrtc/call/rtc_event_log_unittest.cc b/webrtc/call/rtc_event_log_unittest.cc |
index 57d18b7e72e5b783debfc9f9ff2a1b901bc0c88f..e8f0ccf1c2bb26f38bee67163be88beef3d3a071 100644 |
--- a/webrtc/call/rtc_event_log_unittest.cc |
+++ b/webrtc/call/rtc_event_log_unittest.cc |
@@ -20,14 +20,15 @@ |
#include "webrtc/base/buffer.h" |
#include "webrtc/base/checks.h" |
#include "webrtc/base/random.h" |
-#include "webrtc/base/rate_limiter.h" |
#include "webrtc/call.h" |
#include "webrtc/call/rtc_event_log.h" |
#include "webrtc/call/rtc_event_log_parser.h" |
#include "webrtc/call/rtc_event_log_unittest_helper.h" |
#include "webrtc/modules/rtp_rtcp/source/rtcp_packet.h" |
#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/sender_report.h" |
-#include "webrtc/modules/rtp_rtcp/source/rtp_sender.h" |
+#include "webrtc/modules/rtp_rtcp/source/rtp_header_extension.h" |
+#include "webrtc/modules/rtp_rtcp/source/rtp_header_extensions.h" |
+#include "webrtc/modules/rtp_rtcp/source/rtp_packet_to_send.h" |
#include "webrtc/system_wrappers/include/clock.h" |
#include "webrtc/test/test_suite.h" |
#include "webrtc/test/testsupport/fileutils.h" |
@@ -105,55 +106,37 @@ void PrintExpectedEvents(size_t rtp_count, |
* presence of extension number i from kExtensionTypes / kExtensionNames. |
* The least significant bit extension_bitvector has number 0. |
*/ |
-size_t GenerateRtpPacket(uint32_t extensions_bitvector, |
- uint32_t csrcs_count, |
- uint8_t* packet, |
- size_t packet_size, |
- Random* prng) { |
+RtpPacketToSend GenerateRtpPacket(const RtpHeaderExtensionMap* extensions, |
+ uint32_t csrcs_count, |
+ size_t packet_size, |
+ Random* prng) { |
RTC_CHECK_GE(packet_size, 16 + 4 * csrcs_count + 4 * kNumExtensions); |
- Clock* clock = Clock::GetRealTimeClock(); |
- RateLimiter retranmission_rate_limiter(clock, 1000); |
- |
- RTPSender rtp_sender(false, // bool audio |
- clock, // Clock* clock |
- nullptr, // Transport* |
- nullptr, // PacedSender* |
- nullptr, // PacketRouter* |
- nullptr, // SendTimeObserver* |
- nullptr, // BitrateStatisticsObserver* |
- nullptr, // FrameCountObserver* |
- nullptr, // SendSideDelayObserver* |
- nullptr, // RtcEventLog* |
- nullptr, // SendPacketObserver* |
- &retranmission_rate_limiter); |
std::vector<uint32_t> csrcs; |
for (unsigned i = 0; i < csrcs_count; i++) { |
csrcs.push_back(prng->Rand<uint32_t>()); |
} |
- rtp_sender.SetCsrcs(csrcs); |
- rtp_sender.SetSSRC(prng->Rand<uint32_t>()); |
- rtp_sender.SetStartTimestamp(prng->Rand<uint32_t>(), true); |
- rtp_sender.SetSequenceNumber(prng->Rand<uint16_t>()); |
- for (unsigned i = 0; i < kNumExtensions; i++) { |
- if (extensions_bitvector & (1u << i)) { |
- rtp_sender.RegisterRtpHeaderExtension(kExtensionTypes[i], i + 1); |
- } |
- } |
- |
- int8_t payload_type = prng->Rand(0, 127); |
- bool marker_bit = prng->Rand<bool>(); |
- uint32_t capture_timestamp = prng->Rand<uint32_t>(); |
- int64_t capture_time_ms = prng->Rand<uint32_t>(); |
- |
- size_t header_size = rtp_sender.BuildRtpHeader( |
- packet, payload_type, marker_bit, capture_timestamp, capture_time_ms); |
- for (size_t i = header_size; i < packet_size; i++) { |
- packet[i] = prng->Rand<uint8_t>(); |
+ RtpPacketToSend rtp_packet(extensions, packet_size); |
+ rtp_packet.SetPayloadType(prng->Rand(127)); |
+ rtp_packet.SetMarker(prng->Rand<bool>()); |
+ rtp_packet.SetSequenceNumber(prng->Rand<uint16_t>()); |
+ rtp_packet.SetSsrc(prng->Rand<uint32_t>()); |
+ rtp_packet.SetTimestamp(prng->Rand<uint32_t>()); |
+ rtp_packet.SetCsrcs(csrcs); |
+ |
+ rtp_packet.SetExtension<TransmissionOffset>(prng->Rand(0x00ffffff)); |
terelius
2016/08/09 11:43:00
These functions set the extension if and only if t
danilchap
2016/08/09 12:49:23
In this case (calling functions before setting pay
terelius
2016/08/09 13:31:32
So you can only set new extensions before payload,
|
+ rtp_packet.SetExtension<AudioLevel>(prng->Rand<bool>(), prng->Rand(127)); |
+ rtp_packet.SetExtension<AbsoluteSendTime>(prng->Rand<int32_t>()); |
+ rtp_packet.SetExtension<VideoOrientation>(prng->Rand(2)); |
+ rtp_packet.SetExtension<TransportSequenceNumber>(prng->Rand<uint16_t>()); |
+ |
+ size_t payload_size = packet_size - rtp_packet.headers_size(); |
+ uint8_t* payload = rtp_packet.AllocatePayload(payload_size); |
+ for (size_t i = 0; i < payload_size; i++) { |
+ payload[i] = prng->Rand<uint8_t>(); |
} |
- |
- return header_size; |
+ return rtp_packet; |
} |
rtc::Buffer GenerateRtcpPacket(Random* prng) { |
@@ -232,9 +215,8 @@ void LogSessionAndReadBack(size_t rtp_count, |
ASSERT_LE(rtcp_count, rtp_count); |
ASSERT_LE(playout_count, rtp_count); |
ASSERT_LE(bwe_loss_count, rtp_count); |
- std::vector<rtc::Buffer> rtp_packets; |
+ std::vector<RtpPacketToSend> rtp_packets; |
std::vector<rtc::Buffer> rtcp_packets; |
- std::vector<size_t> rtp_header_sizes; |
std::vector<uint32_t> playout_ssrcs; |
std::vector<std::pair<int32_t, uint8_t> > bwe_loss_updates; |
@@ -243,14 +225,18 @@ void LogSessionAndReadBack(size_t rtp_count, |
Random prng(random_seed); |
+ // Initialize rtp header extensions to be used in generated rtp packets. |
+ RtpHeaderExtensionMap extensions; |
+ for (unsigned i = 0; i < kNumExtensions; i++) { |
+ if (extensions_bitvector & (1u << i)) { |
+ extensions.Register(kExtensionTypes[i], i + 1); |
+ } |
+ } |
// Create rtp_count RTP packets containing random data. |
for (size_t i = 0; i < rtp_count; i++) { |
size_t packet_size = prng.Rand(1000, 1100); |
- rtp_packets.push_back(rtc::Buffer(packet_size)); |
- size_t header_size = |
- GenerateRtpPacket(extensions_bitvector, csrcs_count, |
- rtp_packets[i].data(), packet_size, &prng); |
- rtp_header_sizes.push_back(header_size); |
+ rtp_packets.push_back( |
+ GenerateRtpPacket(&extensions, csrcs_count, packet_size, &prng)); |
} |
// Create rtcp_count RTCP packets containing random data. |
for (size_t i = 0; i < rtcp_count; i++) { |
@@ -353,7 +339,7 @@ void LogSessionAndReadBack(size_t rtp_count, |
parsed_log, event_index, |
(i % 2 == 0) ? kIncomingPacket : kOutgoingPacket, |
(i % 3 == 0) ? MediaType::AUDIO : MediaType::VIDEO, |
- rtp_packets[i - 1].data(), rtp_header_sizes[i - 1], |
+ rtp_packets[i - 1].data(), rtp_packets[i - 1].headers_size(), |
rtp_packets[i - 1].size()); |
event_index++; |
if (i * rtcp_count >= rtcp_index * rtp_count) { |
@@ -423,9 +409,8 @@ TEST(RtcEventLogTest, LogEventAndReadBack) { |
// Create one RTP and one RTCP packet containing random data. |
size_t packet_size = prng.Rand(1000, 1100); |
- rtc::Buffer rtp_packet(packet_size); |
- size_t header_size = |
- GenerateRtpPacket(0, 0, rtp_packet.data(), packet_size, &prng); |
+ RtpPacketToSend rtp_packet = |
+ GenerateRtpPacket(nullptr, 0, packet_size, &prng); |
rtc::Buffer rtcp_packet = GenerateRtcpPacket(&prng); |
// Find the name of the current test, in order to use it as a temporary |
@@ -461,9 +446,9 @@ TEST(RtcEventLogTest, LogEventAndReadBack) { |
RtcEventLogTestHelper::VerifyLogStartEvent(parsed_log, 0); |
- RtcEventLogTestHelper::VerifyRtpEvent(parsed_log, 1, kIncomingPacket, |
- MediaType::VIDEO, rtp_packet.data(), |
- header_size, rtp_packet.size()); |
+ RtcEventLogTestHelper::VerifyRtpEvent( |
+ parsed_log, 1, kIncomingPacket, MediaType::VIDEO, rtp_packet.data(), |
+ rtp_packet.headers_size(), rtp_packet.size()); |
RtcEventLogTestHelper::VerifyRtcpEvent(parsed_log, 2, kOutgoingPacket, |
MediaType::VIDEO, rtcp_packet.data(), |