8d609f6b6d
Reason for revert: Re-land, reverting did not fix bug. https://bugs.chromium.org/p/webrtc/issues/detail?id=7465 Original issue's description: > Revert of Implemented the GetSources() in native code. (patchset #11 id:510001 of https://codereview.webrtc.org/2770233003/ ) > > Reason for revert: > Suspected of WebRtcApprtcBrowserTest.MANUAL_WorksOnApprtc breakage, see > > https://bugs.chromium.org/p/webrtc/issues/detail?id=7465 > > Original issue's description: > > Added the GetSources() to the RtpReceiverInterface and implemented > > it for the AudioRtpReceiver. > > > > This method returns a vector of RtpSource(both CSRC source and SSRC > > source) which contains the ID of a source, the timestamp, the source > > type (SSRC or CSRC) and the audio level. > > > > The RtpSource objects are buffered and maintained by the > > RtpReceiver in webrtc/modules/rtp_rtcp/. When the method is called, > > the info of the contributing source will be pulled along the object > > chain: > > AudioRtpReceiver -> VoiceChannel -> WebRtcVoiceMediaChannel -> > > AudioReceiveStream -> voe::Channel -> RtpRtcp module > > > > Spec:https://w3c.github.io/webrtc-pc/archives/20151006/webrtc.html#widl-RTCRtpReceiver-getContributingSources-sequence-RTCRtpContributingSource > > > > BUG=chromium:703122 > > TBR=stefan@webrtc.org, danilchap@webrtc.org > > > > Review-Url: https://codereview.webrtc.org/2770233003 > > Cr-Commit-Position: refs/heads/master@{#17591} > > Committed:292084c376> > TBR=deadbeef@webrtc.org,solenberg@webrtc.org,hbos@webrtc.org,philipel@webrtc.org,stefan@webrtc.org,danilchap@webrtc.org,zhihuang@webrtc.org > # Not skipping CQ checks because original CL landed more than 1 days ago. > BUG=chromium:703122 > > Review-Url: https://codereview.webrtc.org/2809613002 > Cr-Commit-Position: refs/heads/master@{#17616} > Committed:fbcc5cb386TBR=deadbeef@webrtc.org,solenberg@webrtc.org,philipel@webrtc.org,stefan@webrtc.org,danilchap@webrtc.org,zhihuang@webrtc.org,olka@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=chromium:703122 Review-Url: https://codereview.webrtc.org/2810623003 Cr-Commit-Position: refs/heads/master@{#17621}
261 lines
10 KiB
C++
261 lines
10 KiB
C++
/*
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* Copyright (c) 2017 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 <memory>
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#include "webrtc/common_types.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_defines.h"
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#include "webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.h"
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#include "webrtc/test/gtest.h"
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namespace webrtc {
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const uint32_t kTestRate = 64000u;
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const uint8_t kTestPayload[] = {'t', 'e', 's', 't'};
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const uint8_t kPcmuPayloadType = 96;
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const int64_t kGetSourcesTimeoutMs = 10000;
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const int kSourceListsSize = 20;
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class RtpReceiverTest : public ::testing::Test {
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protected:
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RtpReceiverTest()
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: fake_clock_(123456),
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rtp_receiver_(
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RtpReceiver::CreateAudioReceiver(&fake_clock_,
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nullptr,
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nullptr,
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&rtp_payload_registry_)) {
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CodecInst voice_codec = {};
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voice_codec.pltype = kPcmuPayloadType;
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voice_codec.plfreq = 8000;
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voice_codec.rate = kTestRate;
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memcpy(voice_codec.plname, "PCMU", 5);
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rtp_receiver_->RegisterReceivePayload(voice_codec);
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}
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~RtpReceiverTest() {}
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bool FindSourceByIdAndType(const std::vector<RtpSource>& sources,
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uint32_t source_id,
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RtpSourceType type,
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RtpSource* source) {
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for (size_t i = 0; i < sources.size(); ++i) {
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if (sources[i].source_id() == source_id &&
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sources[i].source_type() == type) {
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(*source) = sources[i];
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return true;
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}
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}
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return false;
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}
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SimulatedClock fake_clock_;
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RTPPayloadRegistry rtp_payload_registry_;
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std::unique_ptr<RtpReceiver> rtp_receiver_;
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};
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TEST_F(RtpReceiverTest, GetSources) {
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RTPHeader header;
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header.payloadType = kPcmuPayloadType;
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header.ssrc = 1;
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header.timestamp = fake_clock_.TimeInMilliseconds();
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header.numCSRCs = 2;
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header.arrOfCSRCs[0] = 111;
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header.arrOfCSRCs[1] = 222;
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PayloadUnion payload_specific = {AudioPayload()};
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bool in_order = false;
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RtpSource source(0, 0, RtpSourceType::SSRC);
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EXPECT_TRUE(rtp_receiver_->IncomingRtpPacket(header, kTestPayload, 4,
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payload_specific, in_order));
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auto sources = rtp_receiver_->GetSources();
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// One SSRC source and two CSRC sources.
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ASSERT_EQ(3u, sources.size());
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ASSERT_TRUE(FindSourceByIdAndType(sources, 1u, RtpSourceType::SSRC, &source));
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EXPECT_EQ(fake_clock_.TimeInMilliseconds(), source.timestamp_ms());
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ASSERT_TRUE(
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FindSourceByIdAndType(sources, 222u, RtpSourceType::CSRC, &source));
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EXPECT_EQ(fake_clock_.TimeInMilliseconds(), source.timestamp_ms());
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ASSERT_TRUE(
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FindSourceByIdAndType(sources, 111u, RtpSourceType::CSRC, &source));
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EXPECT_EQ(fake_clock_.TimeInMilliseconds(), source.timestamp_ms());
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// Advance the fake clock and the method is expected to return the
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// contributing source object with same source id and updated timestamp.
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fake_clock_.AdvanceTimeMilliseconds(1);
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EXPECT_TRUE(rtp_receiver_->IncomingRtpPacket(header, kTestPayload, 4,
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payload_specific, in_order));
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sources = rtp_receiver_->GetSources();
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ASSERT_EQ(3u, sources.size());
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ASSERT_TRUE(FindSourceByIdAndType(sources, 1u, RtpSourceType::SSRC, &source));
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EXPECT_EQ(fake_clock_.TimeInMilliseconds(), source.timestamp_ms());
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ASSERT_TRUE(
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FindSourceByIdAndType(sources, 222u, RtpSourceType::CSRC, &source));
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EXPECT_EQ(fake_clock_.TimeInMilliseconds(), source.timestamp_ms());
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ASSERT_TRUE(
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FindSourceByIdAndType(sources, 111u, RtpSourceType::CSRC, &source));
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EXPECT_EQ(fake_clock_.TimeInMilliseconds(), source.timestamp_ms());
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// Test the edge case that the sources are still there just before the
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// timeout.
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int64_t prev_timestamp = fake_clock_.TimeInMilliseconds();
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fake_clock_.AdvanceTimeMilliseconds(kGetSourcesTimeoutMs);
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sources = rtp_receiver_->GetSources();
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ASSERT_EQ(3u, sources.size());
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ASSERT_TRUE(FindSourceByIdAndType(sources, 1u, RtpSourceType::SSRC, &source));
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EXPECT_EQ(prev_timestamp, source.timestamp_ms());
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ASSERT_TRUE(
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FindSourceByIdAndType(sources, 222u, RtpSourceType::CSRC, &source));
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EXPECT_EQ(prev_timestamp, source.timestamp_ms());
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ASSERT_TRUE(
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FindSourceByIdAndType(sources, 111u, RtpSourceType::CSRC, &source));
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EXPECT_EQ(prev_timestamp, source.timestamp_ms());
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// Time out.
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fake_clock_.AdvanceTimeMilliseconds(1);
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sources = rtp_receiver_->GetSources();
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// All the sources should be out of date.
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ASSERT_EQ(0u, sources.size());
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}
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// Test the case that the SSRC is changed.
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TEST_F(RtpReceiverTest, GetSourcesChangeSSRC) {
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int64_t prev_time = -1;
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int64_t cur_time = fake_clock_.TimeInMilliseconds();
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RTPHeader header;
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header.payloadType = kPcmuPayloadType;
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header.ssrc = 1;
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header.timestamp = cur_time;
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PayloadUnion payload_specific = {AudioPayload()};
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bool in_order = false;
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RtpSource source(0, 0, RtpSourceType::SSRC);
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EXPECT_TRUE(rtp_receiver_->IncomingRtpPacket(header, kTestPayload, 4,
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payload_specific, in_order));
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auto sources = rtp_receiver_->GetSources();
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ASSERT_EQ(1u, sources.size());
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EXPECT_EQ(1u, sources[0].source_id());
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EXPECT_EQ(cur_time, sources[0].timestamp_ms());
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// The SSRC is changed and the old SSRC is expected to be returned.
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fake_clock_.AdvanceTimeMilliseconds(100);
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prev_time = cur_time;
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cur_time = fake_clock_.TimeInMilliseconds();
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header.ssrc = 2;
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header.timestamp = cur_time;
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EXPECT_TRUE(rtp_receiver_->IncomingRtpPacket(header, kTestPayload, 4,
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payload_specific, in_order));
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sources = rtp_receiver_->GetSources();
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ASSERT_EQ(2u, sources.size());
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ASSERT_TRUE(FindSourceByIdAndType(sources, 2u, RtpSourceType::SSRC, &source));
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EXPECT_EQ(cur_time, source.timestamp_ms());
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ASSERT_TRUE(FindSourceByIdAndType(sources, 1u, RtpSourceType::SSRC, &source));
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EXPECT_EQ(prev_time, source.timestamp_ms());
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// The SSRC is changed again and happen to be changed back to 1. No
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// duplication is expected.
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fake_clock_.AdvanceTimeMilliseconds(100);
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header.ssrc = 1;
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header.timestamp = cur_time;
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prev_time = cur_time;
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cur_time = fake_clock_.TimeInMilliseconds();
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EXPECT_TRUE(rtp_receiver_->IncomingRtpPacket(header, kTestPayload, 4,
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payload_specific, in_order));
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sources = rtp_receiver_->GetSources();
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ASSERT_EQ(2u, sources.size());
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ASSERT_TRUE(FindSourceByIdAndType(sources, 1u, RtpSourceType::SSRC, &source));
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EXPECT_EQ(cur_time, source.timestamp_ms());
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ASSERT_TRUE(FindSourceByIdAndType(sources, 2u, RtpSourceType::SSRC, &source));
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EXPECT_EQ(prev_time, source.timestamp_ms());
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// Old SSRC source timeout.
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fake_clock_.AdvanceTimeMilliseconds(kGetSourcesTimeoutMs);
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cur_time = fake_clock_.TimeInMilliseconds();
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EXPECT_TRUE(rtp_receiver_->IncomingRtpPacket(header, kTestPayload, 4,
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payload_specific, in_order));
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sources = rtp_receiver_->GetSources();
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ASSERT_EQ(1u, sources.size());
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EXPECT_EQ(1u, sources[0].source_id());
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EXPECT_EQ(cur_time, sources[0].timestamp_ms());
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EXPECT_EQ(RtpSourceType::SSRC, sources[0].source_type());
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}
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TEST_F(RtpReceiverTest, GetSourcesRemoveOutdatedSource) {
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int64_t timestamp = fake_clock_.TimeInMilliseconds();
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bool in_order = false;
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RTPHeader header;
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header.payloadType = kPcmuPayloadType;
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header.timestamp = timestamp;
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PayloadUnion payload_specific = {AudioPayload()};
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header.numCSRCs = 1;
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RtpSource source(0, 0, RtpSourceType::SSRC);
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for (size_t i = 0; i < kSourceListsSize; ++i) {
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header.ssrc = i;
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header.arrOfCSRCs[0] = (i + 1);
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EXPECT_TRUE(rtp_receiver_->IncomingRtpPacket(header, kTestPayload, 4,
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payload_specific, in_order));
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}
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auto sources = rtp_receiver_->GetSources();
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// Expect |kSourceListsSize| SSRC sources and |kSourceListsSize| CSRC sources.
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ASSERT_TRUE(sources.size() == 2 * kSourceListsSize);
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for (size_t i = 0; i < kSourceListsSize; ++i) {
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// The SSRC source IDs are expected to be 19, 18, 17 ... 0
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ASSERT_TRUE(
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FindSourceByIdAndType(sources, i, RtpSourceType::SSRC, &source));
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EXPECT_EQ(timestamp, source.timestamp_ms());
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// The CSRC source IDs are expected to be 20, 19, 18 ... 1
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ASSERT_TRUE(
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FindSourceByIdAndType(sources, (i + 1), RtpSourceType::CSRC, &source));
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EXPECT_EQ(timestamp, source.timestamp_ms());
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}
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fake_clock_.AdvanceTimeMilliseconds(kGetSourcesTimeoutMs);
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for (size_t i = 0; i < kSourceListsSize; ++i) {
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// The SSRC source IDs are expected to be 19, 18, 17 ... 0
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ASSERT_TRUE(
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FindSourceByIdAndType(sources, i, RtpSourceType::SSRC, &source));
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EXPECT_EQ(timestamp, source.timestamp_ms());
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// The CSRC source IDs are expected to be 20, 19, 18 ... 1
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ASSERT_TRUE(
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FindSourceByIdAndType(sources, (i + 1), RtpSourceType::CSRC, &source));
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EXPECT_EQ(timestamp, source.timestamp_ms());
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}
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// Timeout. All the existing objects are out of date and are expected to be
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// removed.
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fake_clock_.AdvanceTimeMilliseconds(1);
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header.ssrc = 111;
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header.arrOfCSRCs[0] = 222;
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EXPECT_TRUE(rtp_receiver_->IncomingRtpPacket(header, kTestPayload, 4,
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payload_specific, in_order));
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auto rtp_receiver_impl = static_cast<RtpReceiverImpl*>(rtp_receiver_.get());
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auto ssrc_sources = rtp_receiver_impl->ssrc_sources_for_testing();
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ASSERT_EQ(1u, ssrc_sources.size());
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EXPECT_EQ(111u, ssrc_sources.begin()->source_id());
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EXPECT_EQ(RtpSourceType::SSRC, ssrc_sources.begin()->source_type());
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EXPECT_EQ(fake_clock_.TimeInMilliseconds(),
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ssrc_sources.begin()->timestamp_ms());
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auto csrc_sources = rtp_receiver_impl->csrc_sources_for_testing();
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ASSERT_EQ(1u, csrc_sources.size());
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EXPECT_EQ(222u, csrc_sources.begin()->source_id());
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EXPECT_EQ(RtpSourceType::CSRC, csrc_sources.begin()->source_type());
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EXPECT_EQ(fake_clock_.TimeInMilliseconds(),
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csrc_sources.begin()->timestamp_ms());
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}
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} // namespace webrtc
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