Peter Boström d6f1a38165 Remove ViEChannel simulcast lock.
Since the number of streams is now known on construction we can
initialize all RTP modules on construction. They are internally locked
so we don't nede a simulcast lock anymore.

BUG=1695
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/52639004 .

Cr-Commit-Position: refs/heads/master@{#9577}
2015-07-14 14:08:14 +00:00

190 lines
6.0 KiB
C++

/*
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "webrtc/modules/rtp_rtcp/test/testAPI/test_api.h"
#include <algorithm>
#include <vector>
using namespace webrtc;
namespace webrtc {
void LoopBackTransport::SetSendModule(RtpRtcp* rtp_rtcp_module,
RTPPayloadRegistry* payload_registry,
RtpReceiver* receiver,
ReceiveStatistics* receive_statistics) {
rtp_rtcp_module_ = rtp_rtcp_module;
rtp_payload_registry_ = payload_registry;
rtp_receiver_ = receiver;
receive_statistics_ = receive_statistics;
}
void LoopBackTransport::DropEveryNthPacket(int n) {
packet_loss_ = n;
}
int LoopBackTransport::SendPacket(int channel, const void* data, size_t len) {
count_++;
if (packet_loss_ > 0) {
if ((count_ % packet_loss_) == 0) {
return len;
}
}
RTPHeader header;
rtc::scoped_ptr<RtpHeaderParser> parser(RtpHeaderParser::Create());
if (!parser->Parse(static_cast<const uint8_t*>(data), len, &header)) {
return -1;
}
PayloadUnion payload_specific;
if (!rtp_payload_registry_->GetPayloadSpecifics(header.payloadType,
&payload_specific)) {
return -1;
}
receive_statistics_->IncomingPacket(header, len, false);
if (!rtp_receiver_->IncomingRtpPacket(header,
static_cast<const uint8_t*>(data), len,
payload_specific, true)) {
return -1;
}
return len;
}
int LoopBackTransport::SendRTCPPacket(int channel,
const void* data,
size_t len) {
if (rtp_rtcp_module_->IncomingRtcpPacket((const uint8_t*)data, len) < 0) {
return -1;
}
return static_cast<int>(len);
}
int32_t TestRtpReceiver::OnReceivedPayloadData(
const uint8_t* payload_data,
const size_t payload_size,
const webrtc::WebRtcRTPHeader* rtp_header) {
EXPECT_LE(payload_size, sizeof(payload_data_));
memcpy(payload_data_, payload_data, payload_size);
memcpy(&rtp_header_, rtp_header, sizeof(rtp_header_));
payload_size_ = payload_size;
return 0;
}
} // namespace webrtc
class RtpRtcpAPITest : public ::testing::Test {
protected:
RtpRtcpAPITest() : fake_clock_(123456) {
test_csrcs_.push_back(1234);
test_csrcs_.push_back(2345);
test_id = 123;
test_ssrc_ = 3456;
test_timestamp_ = 4567;
test_sequence_number_ = 2345;
}
~RtpRtcpAPITest() {}
void SetUp() override {
RtpRtcp::Configuration configuration;
configuration.id = test_id;
configuration.audio = true;
configuration.clock = &fake_clock_;
module_.reset(RtpRtcp::CreateRtpRtcp(configuration));
rtp_payload_registry_.reset(new RTPPayloadRegistry(
RTPPayloadStrategy::CreateStrategy(true)));
rtp_receiver_.reset(RtpReceiver::CreateAudioReceiver(
test_id, &fake_clock_, NULL, NULL, NULL, rtp_payload_registry_.get()));
}
int test_id;
rtc::scoped_ptr<RTPPayloadRegistry> rtp_payload_registry_;
rtc::scoped_ptr<RtpReceiver> rtp_receiver_;
rtc::scoped_ptr<RtpRtcp> module_;
uint32_t test_ssrc_;
uint32_t test_timestamp_;
uint16_t test_sequence_number_;
std::vector<uint32_t> test_csrcs_;
SimulatedClock fake_clock_;
};
TEST_F(RtpRtcpAPITest, Basic) {
module_->SetSequenceNumber(test_sequence_number_);
EXPECT_EQ(test_sequence_number_, module_->SequenceNumber());
module_->SetStartTimestamp(test_timestamp_);
EXPECT_EQ(test_timestamp_, module_->StartTimestamp());
EXPECT_FALSE(module_->Sending());
EXPECT_EQ(0, module_->SetSendingStatus(true));
EXPECT_TRUE(module_->Sending());
}
TEST_F(RtpRtcpAPITest, MTU) {
EXPECT_EQ(0, module_->SetMaxTransferUnit(1234));
EXPECT_EQ(1234 - 20 - 8, module_->MaxPayloadLength());
EXPECT_EQ(0, module_->SetTransportOverhead(true, true, 12));
EXPECT_EQ(1234 - 20 - 20 - 20 - 12, module_->MaxPayloadLength());
EXPECT_EQ(0, module_->SetTransportOverhead(false, false, 0));
EXPECT_EQ(1234 - 20 - 8, module_->MaxPayloadLength());
}
TEST_F(RtpRtcpAPITest, SSRC) {
module_->SetSSRC(test_ssrc_);
EXPECT_EQ(test_ssrc_, module_->SSRC());
}
TEST_F(RtpRtcpAPITest, RTCP) {
EXPECT_EQ(kRtcpOff, module_->RTCP());
module_->SetRTCPStatus(kRtcpCompound);
EXPECT_EQ(kRtcpCompound, module_->RTCP());
EXPECT_EQ(0, module_->SetCNAME("john.doe@test.test"));
EXPECT_FALSE(module_->TMMBR());
module_->SetTMMBRStatus(true);
EXPECT_TRUE(module_->TMMBR());
module_->SetTMMBRStatus(false);
EXPECT_FALSE(module_->TMMBR());
EXPECT_EQ(kNackOff, rtp_receiver_->NACK());
rtp_receiver_->SetNACKStatus(kNackRtcp);
EXPECT_EQ(kNackRtcp, rtp_receiver_->NACK());
}
TEST_F(RtpRtcpAPITest, RtxSender) {
module_->SetRtxSendStatus(kRtxRetransmitted);
EXPECT_EQ(kRtxRetransmitted, module_->RtxSendStatus());
module_->SetRtxSendStatus(kRtxOff);
EXPECT_EQ(kRtxOff, module_->RtxSendStatus());
module_->SetRtxSendStatus(kRtxRetransmitted);
EXPECT_EQ(kRtxRetransmitted, module_->RtxSendStatus());
}
TEST_F(RtpRtcpAPITest, RtxReceiver) {
const uint32_t kRtxSsrc = 1;
const int kRtxPayloadType = 119;
const int kPayloadType = 100;
EXPECT_FALSE(rtp_payload_registry_->RtxEnabled());
rtp_payload_registry_->SetRtxSsrc(kRtxSsrc);
rtp_payload_registry_->SetRtxPayloadType(kRtxPayloadType, kPayloadType);
EXPECT_TRUE(rtp_payload_registry_->RtxEnabled());
RTPHeader rtx_header;
rtx_header.ssrc = kRtxSsrc;
rtx_header.payloadType = kRtxPayloadType;
EXPECT_TRUE(rtp_payload_registry_->IsRtx(rtx_header));
rtx_header.ssrc = 0;
EXPECT_FALSE(rtp_payload_registry_->IsRtx(rtx_header));
rtx_header.ssrc = kRtxSsrc;
rtx_header.payloadType = 0;
EXPECT_TRUE(rtp_payload_registry_->IsRtx(rtx_header));
}