/*
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* Copyright (c) 2013 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 <algorithm>
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#include <sstream>
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#include <string>
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#include "testing/gtest/include/gtest/gtest.h"
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#include "webrtc/base/checks.h"
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#include "webrtc/base/scoped_ptr.h"
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#include "webrtc/base/thread_annotations.h"
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#include "webrtc/call.h"
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#include "webrtc/call/transport_adapter.h"
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#include "webrtc/common.h"
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#include "webrtc/config.h"
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#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
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#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
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#include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
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#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
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#include "webrtc/system_wrappers/include/rtp_to_ntp.h"
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#include "webrtc/test/call_test.h"
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#include "webrtc/test/direct_transport.h"
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#include "webrtc/test/encoder_settings.h"
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#include "webrtc/test/fake_audio_device.h"
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#include "webrtc/test/fake_decoder.h"
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#include "webrtc/test/fake_encoder.h"
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#include "webrtc/test/frame_generator.h"
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#include "webrtc/test/frame_generator_capturer.h"
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#include "webrtc/test/rtp_rtcp_observer.h"
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#include "webrtc/test/testsupport/fileutils.h"
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#include "webrtc/test/testsupport/perf_test.h"
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#include "webrtc/voice_engine/include/voe_base.h"
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#include "webrtc/voice_engine/include/voe_codec.h"
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#include "webrtc/voice_engine/include/voe_network.h"
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#include "webrtc/voice_engine/include/voe_rtp_rtcp.h"
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#include "webrtc/voice_engine/include/voe_video_sync.h"
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namespace webrtc {
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class CallPerfTest : public test::CallTest {
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protected:
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void TestAudioVideoSync(bool fec, bool create_audio_first);
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void TestCpuOveruse(LoadObserver::Load tested_load, int encode_delay_ms);
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void TestMinTransmitBitrate(bool pad_to_min_bitrate);
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void TestCaptureNtpTime(const FakeNetworkPipe::Config& net_config,
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int threshold_ms,
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int start_time_ms,
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int run_time_ms);
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};
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class SyncRtcpObserver : public test::RtpRtcpObserver {
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public:
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SyncRtcpObserver() : test::RtpRtcpObserver(CallPerfTest::kLongTimeoutMs) {}
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Action OnSendRtcp(const uint8_t* packet, size_t length) override {
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RTCPUtility::RTCPParserV2 parser(packet, length, true);
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EXPECT_TRUE(parser.IsValid());
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for (RTCPUtility::RTCPPacketTypes packet_type = parser.Begin();
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packet_type != RTCPUtility::RTCPPacketTypes::kInvalid;
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packet_type = parser.Iterate()) {
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if (packet_type == RTCPUtility::RTCPPacketTypes::kSr) {
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const RTCPUtility::RTCPPacket& packet = parser.Packet();
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RtcpMeasurement ntp_rtp_pair(
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packet.SR.NTPMostSignificant,
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packet.SR.NTPLeastSignificant,
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packet.SR.RTPTimestamp);
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StoreNtpRtpPair(ntp_rtp_pair);
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}
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}
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return SEND_PACKET;
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}
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int64_t RtpTimestampToNtp(uint32_t timestamp) const {
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rtc::CritScope lock(&crit_);
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int64_t timestamp_in_ms = -1;
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if (ntp_rtp_pairs_.size() == 2) {
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// TODO(stefan): We can't EXPECT_TRUE on this call due to a bug in the
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// RTCP sender where it sends RTCP SR before any RTP packets, which leads
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// to a bogus NTP/RTP mapping.
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RtpToNtpMs(timestamp, ntp_rtp_pairs_, ×tamp_in_ms);
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return timestamp_in_ms;
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}
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return -1;
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}
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private:
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void StoreNtpRtpPair(RtcpMeasurement ntp_rtp_pair) {
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rtc::CritScope lock(&crit_);
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for (RtcpList::iterator it = ntp_rtp_pairs_.begin();
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it != ntp_rtp_pairs_.end();
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++it) {
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if (ntp_rtp_pair.ntp_secs == it->ntp_secs &&
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ntp_rtp_pair.ntp_frac == it->ntp_frac) {
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// This RTCP has already been added to the list.
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return;
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}
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}
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// We need two RTCP SR reports to map between RTP and NTP. More than two
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// will not improve the mapping.
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if (ntp_rtp_pairs_.size() == 2) {
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ntp_rtp_pairs_.pop_back();
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}
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ntp_rtp_pairs_.push_front(ntp_rtp_pair);
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}
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mutable rtc::CriticalSection crit_;
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RtcpList ntp_rtp_pairs_ GUARDED_BY(crit_);
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};
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class VideoRtcpAndSyncObserver : public SyncRtcpObserver, public VideoRenderer {
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static const int kInSyncThresholdMs = 50;
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static const int kStartupTimeMs = 2000;
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static const int kMinRunTimeMs = 30000;
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public:
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VideoRtcpAndSyncObserver(Clock* clock,
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int voe_channel,
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VoEVideoSync* voe_sync,
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SyncRtcpObserver* audio_observer)
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: clock_(clock),
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voe_channel_(voe_channel),
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voe_sync_(voe_sync),
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audio_observer_(audio_observer),
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creation_time_ms_(clock_->TimeInMilliseconds()),
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first_time_in_sync_(-1) {}
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void RenderFrame(const VideoFrame& video_frame,
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int time_to_render_ms) override {
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int64_t now_ms = clock_->TimeInMilliseconds();
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uint32_t playout_timestamp = 0;
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if (voe_sync_->GetPlayoutTimestamp(voe_channel_, playout_timestamp) != 0)
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return;
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int64_t latest_audio_ntp =
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audio_observer_->RtpTimestampToNtp(playout_timestamp);
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int64_t latest_video_ntp = RtpTimestampToNtp(video_frame.timestamp());
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if (latest_audio_ntp < 0 || latest_video_ntp < 0)
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return;
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int time_until_render_ms =
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std::max(0, static_cast<int>(video_frame.render_time_ms() - now_ms));
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latest_video_ntp += time_until_render_ms;
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int64_t stream_offset = latest_audio_ntp - latest_video_ntp;
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std::stringstream ss;
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ss << stream_offset;
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webrtc::test::PrintResult("stream_offset",
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"",
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"synchronization",
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ss.str(),
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"ms",
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false);
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int64_t time_since_creation = now_ms - creation_time_ms_;
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// During the first couple of seconds audio and video can falsely be
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// estimated as being synchronized. We don't want to trigger on those.
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if (time_since_creation < kStartupTimeMs)
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return;
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if (std::abs(latest_audio_ntp - latest_video_ntp) < kInSyncThresholdMs) {
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if (first_time_in_sync_ == -1) {
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first_time_in_sync_ = now_ms;
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webrtc::test::PrintResult("sync_convergence_time",
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"",
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"synchronization",
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time_since_creation,
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"ms",
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false);
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}
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if (time_since_creation > kMinRunTimeMs)
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observation_complete_.Set();
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}
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}
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bool IsTextureSupported() const override { return false; }
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private:
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Clock* const clock_;
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const int voe_channel_;
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VoEVideoSync* const voe_sync_;
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SyncRtcpObserver* const audio_observer_;
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const int64_t creation_time_ms_;
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int64_t first_time_in_sync_;
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};
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void CallPerfTest::TestAudioVideoSync(bool fec, bool create_audio_first) {
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const char* kSyncGroup = "av_sync";
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const uint32_t kAudioSendSsrc = 1234;
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const uint32_t kAudioRecvSsrc = 5678;
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class AudioPacketReceiver : public PacketReceiver {
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public:
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AudioPacketReceiver(int channel, VoENetwork* voe_network)
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: channel_(channel),
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voe_network_(voe_network),
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parser_(RtpHeaderParser::Create()) {}
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DeliveryStatus DeliverPacket(MediaType media_type,
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const uint8_t* packet,
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size_t length,
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const PacketTime& packet_time) override {
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EXPECT_TRUE(media_type == MediaType::ANY ||
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media_type == MediaType::AUDIO);
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int ret;
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if (parser_->IsRtcp(packet, length)) {
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ret = voe_network_->ReceivedRTCPPacket(channel_, packet, length);
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} else {
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ret = voe_network_->ReceivedRTPPacket(channel_, packet, length,
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PacketTime());
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}
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return ret == 0 ? DELIVERY_OK : DELIVERY_PACKET_ERROR;
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}
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private:
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int channel_;
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VoENetwork* voe_network_;
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rtc::scoped_ptr<RtpHeaderParser> parser_;
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};
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VoiceEngine* voice_engine = VoiceEngine::Create();
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VoEBase* voe_base = VoEBase::GetInterface(voice_engine);
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VoECodec* voe_codec = VoECodec::GetInterface(voice_engine);
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VoENetwork* voe_network = VoENetwork::GetInterface(voice_engine);
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VoEVideoSync* voe_sync = VoEVideoSync::GetInterface(voice_engine);
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const std::string audio_filename =
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test::ResourcePath("voice_engine/audio_long16", "pcm");
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ASSERT_STRNE("", audio_filename.c_str());
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test::FakeAudioDevice fake_audio_device(Clock::GetRealTimeClock(),
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audio_filename);
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EXPECT_EQ(0, voe_base->Init(&fake_audio_device, nullptr));
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Config voe_config;
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voe_config.Set<VoicePacing>(new VoicePacing(true));
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int send_channel_id = voe_base->CreateChannel(voe_config);
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int recv_channel_id = voe_base->CreateChannel();
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SyncRtcpObserver audio_observer;
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AudioState::Config send_audio_state_config;
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send_audio_state_config.voice_engine = voice_engine;
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Call::Config sender_config;
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sender_config.audio_state = AudioState::Create(send_audio_state_config);
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Call::Config receiver_config;
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receiver_config.audio_state = sender_config.audio_state;
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CreateCalls(sender_config, receiver_config);
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AudioPacketReceiver voe_send_packet_receiver(send_channel_id, voe_network);
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AudioPacketReceiver voe_recv_packet_receiver(recv_channel_id, voe_network);
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FakeNetworkPipe::Config net_config;
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net_config.queue_delay_ms = 500;
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net_config.loss_percent = 5;
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test::PacketTransport audio_send_transport(
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nullptr, &audio_observer, test::PacketTransport::kSender, net_config);
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audio_send_transport.SetReceiver(&voe_recv_packet_receiver);
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test::PacketTransport audio_receive_transport(
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nullptr, &audio_observer, test::PacketTransport::kReceiver, net_config);
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audio_receive_transport.SetReceiver(&voe_send_packet_receiver);
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internal::TransportAdapter send_transport_adapter(&audio_send_transport);
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send_transport_adapter.Enable();
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EXPECT_EQ(0, voe_network->RegisterExternalTransport(send_channel_id,
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send_transport_adapter));
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internal::TransportAdapter recv_transport_adapter(&audio_receive_transport);
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recv_transport_adapter.Enable();
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EXPECT_EQ(0, voe_network->RegisterExternalTransport(recv_channel_id,
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recv_transport_adapter));
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VideoRtcpAndSyncObserver observer(Clock::GetRealTimeClock(), recv_channel_id,
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voe_sync, &audio_observer);
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test::PacketTransport sync_send_transport(sender_call_.get(), &observer,
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test::PacketTransport::kSender,
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FakeNetworkPipe::Config());
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sync_send_transport.SetReceiver(receiver_call_->Receiver());
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test::PacketTransport sync_receive_transport(receiver_call_.get(), &observer,
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test::PacketTransport::kReceiver,
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FakeNetworkPipe::Config());
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sync_receive_transport.SetReceiver(sender_call_->Receiver());
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test::FakeDecoder fake_decoder;
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CreateSendConfig(1, 0, &sync_send_transport);
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CreateMatchingReceiveConfigs(&sync_receive_transport);
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AudioSendStream::Config audio_send_config(&audio_send_transport);
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audio_send_config.voe_channel_id = send_channel_id;
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audio_send_config.rtp.ssrc = kAudioSendSsrc;
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AudioSendStream* audio_send_stream =
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sender_call_->CreateAudioSendStream(audio_send_config);
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CodecInst isac = {103, "ISAC", 16000, 480, 1, 32000};
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EXPECT_EQ(0, voe_codec->SetSendCodec(send_channel_id, isac));
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video_send_config_.rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
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if (fec) {
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video_send_config_.rtp.fec.red_payload_type = kRedPayloadType;
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video_send_config_.rtp.fec.ulpfec_payload_type = kUlpfecPayloadType;
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video_receive_configs_[0].rtp.fec.red_payload_type = kRedPayloadType;
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video_receive_configs_[0].rtp.fec.ulpfec_payload_type = kUlpfecPayloadType;
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}
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video_receive_configs_[0].rtp.nack.rtp_history_ms = 1000;
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video_receive_configs_[0].renderer = &observer;
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video_receive_configs_[0].sync_group = kSyncGroup;
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AudioReceiveStream::Config audio_recv_config;
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audio_recv_config.rtp.remote_ssrc = kAudioSendSsrc;
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audio_recv_config.rtp.local_ssrc = kAudioRecvSsrc;
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audio_recv_config.voe_channel_id = recv_channel_id;
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audio_recv_config.sync_group = kSyncGroup;
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AudioReceiveStream* audio_receive_stream;
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if (create_audio_first) {
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audio_receive_stream =
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receiver_call_->CreateAudioReceiveStream(audio_recv_config);
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CreateVideoStreams();
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} else {
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CreateVideoStreams();
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audio_receive_stream =
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receiver_call_->CreateAudioReceiveStream(audio_recv_config);
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}
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CreateFrameGeneratorCapturer();
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Start();
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fake_audio_device.Start();
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EXPECT_EQ(0, voe_base->StartPlayout(recv_channel_id));
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EXPECT_EQ(0, voe_base->StartReceive(recv_channel_id));
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EXPECT_EQ(0, voe_base->StartSend(send_channel_id));
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EXPECT_TRUE(observer.Wait())
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<< "Timed out while waiting for audio and video to be synchronized.";
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EXPECT_EQ(0, voe_base->StopSend(send_channel_id));
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EXPECT_EQ(0, voe_base->StopReceive(recv_channel_id));
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EXPECT_EQ(0, voe_base->StopPlayout(recv_channel_id));
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fake_audio_device.Stop();
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Stop();
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sync_send_transport.StopSending();
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sync_receive_transport.StopSending();
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audio_send_transport.StopSending();
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audio_receive_transport.StopSending();
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DestroyStreams();
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sender_call_->DestroyAudioSendStream(audio_send_stream);
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receiver_call_->DestroyAudioReceiveStream(audio_receive_stream);
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voe_base->DeleteChannel(send_channel_id);
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voe_base->DeleteChannel(recv_channel_id);
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voe_base->Release();
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voe_codec->Release();
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voe_network->Release();
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voe_sync->Release();
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DestroyCalls();
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VoiceEngine::Delete(voice_engine);
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}
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TEST_F(CallPerfTest, PlaysOutAudioAndVideoInSyncWithAudioCreatedFirst) {
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TestAudioVideoSync(false, true);
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}
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TEST_F(CallPerfTest, PlaysOutAudioAndVideoInSyncWithVideoCreatedFirst) {
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TestAudioVideoSync(false, false);
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}
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TEST_F(CallPerfTest, PlaysOutAudioAndVideoInSyncWithFec) {
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TestAudioVideoSync(true, false);
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}
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void CallPerfTest::TestCaptureNtpTime(const FakeNetworkPipe::Config& net_config,
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int threshold_ms,
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int start_time_ms,
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int run_time_ms) {
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class CaptureNtpTimeObserver : public test::EndToEndTest,
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public VideoRenderer {
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public:
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CaptureNtpTimeObserver(const FakeNetworkPipe::Config& net_config,
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int threshold_ms,
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int start_time_ms,
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int run_time_ms)
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: EndToEndTest(kLongTimeoutMs),
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net_config_(net_config),
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clock_(Clock::GetRealTimeClock()),
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threshold_ms_(threshold_ms),
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start_time_ms_(start_time_ms),
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run_time_ms_(run_time_ms),
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creation_time_ms_(clock_->TimeInMilliseconds()),
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capturer_(nullptr),
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rtp_start_timestamp_set_(false),
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rtp_start_timestamp_(0) {}
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private:
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test::PacketTransport* CreateSendTransport(Call* sender_call) override {
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return new test::PacketTransport(
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sender_call, this, test::PacketTransport::kSender, net_config_);
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}
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test::PacketTransport* CreateReceiveTransport() override {
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return new test::PacketTransport(
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nullptr, this, test::PacketTransport::kReceiver, net_config_);
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}
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void RenderFrame(const VideoFrame& video_frame,
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int time_to_render_ms) override {
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rtc::CritScope lock(&crit_);
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if (video_frame.ntp_time_ms() <= 0) {
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// Haven't got enough RTCP SR in order to calculate the capture ntp
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// time.
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return;
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}
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int64_t now_ms = clock_->TimeInMilliseconds();
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int64_t time_since_creation = now_ms - creation_time_ms_;
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if (time_since_creation < start_time_ms_) {
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// Wait for |start_time_ms_| before start measuring.
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return;
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}
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if (time_since_creation > run_time_ms_) {
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observation_complete_.Set();
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}
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FrameCaptureTimeList::iterator iter =
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capture_time_list_.find(video_frame.timestamp());
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EXPECT_TRUE(iter != capture_time_list_.end());
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// The real capture time has been wrapped to uint32_t before converted
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// to rtp timestamp in the sender side. So here we convert the estimated
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// capture time to a uint32_t 90k timestamp also for comparing.
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uint32_t estimated_capture_timestamp =
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90 * static_cast<uint32_t>(video_frame.ntp_time_ms());
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uint32_t real_capture_timestamp = iter->second;
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int time_offset_ms = real_capture_timestamp - estimated_capture_timestamp;
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time_offset_ms = time_offset_ms / 90;
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std::stringstream ss;
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ss << time_offset_ms;
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webrtc::test::PrintResult(
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"capture_ntp_time", "", "real - estimated", ss.str(), "ms", true);
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EXPECT_TRUE(std::abs(time_offset_ms) < threshold_ms_);
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}
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bool IsTextureSupported() const override { return false; }
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virtual Action OnSendRtp(const uint8_t* packet, size_t length) {
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rtc::CritScope lock(&crit_);
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RTPHeader header;
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EXPECT_TRUE(parser_->Parse(packet, length, &header));
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if (!rtp_start_timestamp_set_) {
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// Calculate the rtp timestamp offset in order to calculate the real
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// capture time.
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uint32_t first_capture_timestamp =
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90 * static_cast<uint32_t>(capturer_->first_frame_capture_time());
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rtp_start_timestamp_ = header.timestamp - first_capture_timestamp;
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rtp_start_timestamp_set_ = true;
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}
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uint32_t capture_timestamp = header.timestamp - rtp_start_timestamp_;
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capture_time_list_.insert(
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capture_time_list_.end(),
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std::make_pair(header.timestamp, capture_timestamp));
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return SEND_PACKET;
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}
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void OnFrameGeneratorCapturerCreated(
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test::FrameGeneratorCapturer* frame_generator_capturer) override {
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capturer_ = frame_generator_capturer;
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}
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void ModifyVideoConfigs(
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VideoSendStream::Config* send_config,
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std::vector<VideoReceiveStream::Config>* receive_configs,
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VideoEncoderConfig* encoder_config) override {
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(*receive_configs)[0].renderer = this;
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// Enable the receiver side rtt calculation.
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(*receive_configs)[0].rtp.rtcp_xr.receiver_reference_time_report = true;
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}
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void PerformTest() override {
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EXPECT_TRUE(Wait()) << "Timed out while waiting for "
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"estimated capture NTP time to be "
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"within bounds.";
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}
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rtc::CriticalSection crit_;
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const FakeNetworkPipe::Config net_config_;
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Clock* const clock_;
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int threshold_ms_;
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int start_time_ms_;
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int run_time_ms_;
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int64_t creation_time_ms_;
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test::FrameGeneratorCapturer* capturer_;
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bool rtp_start_timestamp_set_;
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uint32_t rtp_start_timestamp_;
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typedef std::map<uint32_t, uint32_t> FrameCaptureTimeList;
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FrameCaptureTimeList capture_time_list_ GUARDED_BY(&crit_);
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} test(net_config, threshold_ms, start_time_ms, run_time_ms);
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RunBaseTest(&test);
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}
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TEST_F(CallPerfTest, CaptureNtpTimeWithNetworkDelay) {
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FakeNetworkPipe::Config net_config;
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net_config.queue_delay_ms = 100;
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// TODO(wu): lower the threshold as the calculation/estimatation becomes more
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// accurate.
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const int kThresholdMs = 100;
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const int kStartTimeMs = 10000;
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const int kRunTimeMs = 20000;
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TestCaptureNtpTime(net_config, kThresholdMs, kStartTimeMs, kRunTimeMs);
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}
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TEST_F(CallPerfTest, CaptureNtpTimeWithNetworkJitter) {
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FakeNetworkPipe::Config net_config;
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net_config.queue_delay_ms = 100;
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net_config.delay_standard_deviation_ms = 10;
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// TODO(wu): lower the threshold as the calculation/estimatation becomes more
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// accurate.
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const int kThresholdMs = 100;
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const int kStartTimeMs = 10000;
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const int kRunTimeMs = 20000;
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TestCaptureNtpTime(net_config, kThresholdMs, kStartTimeMs, kRunTimeMs);
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}
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void CallPerfTest::TestCpuOveruse(LoadObserver::Load tested_load,
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int encode_delay_ms) {
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class LoadObserver : public test::SendTest, public webrtc::LoadObserver {
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public:
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LoadObserver(LoadObserver::Load tested_load, int encode_delay_ms)
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: SendTest(kLongTimeoutMs),
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tested_load_(tested_load),
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encoder_(Clock::GetRealTimeClock(), encode_delay_ms) {}
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void OnLoadUpdate(Load load) override {
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if (load == tested_load_)
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observation_complete_.Set();
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}
|
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void ModifyVideoConfigs(
|
VideoSendStream::Config* send_config,
|
std::vector<VideoReceiveStream::Config>* receive_configs,
|
VideoEncoderConfig* encoder_config) override {
|
send_config->overuse_callback = this;
|
send_config->encoder_settings.encoder = &encoder_;
|
}
|
|
void PerformTest() override {
|
EXPECT_TRUE(Wait()) << "Timed out before receiving an overuse callback.";
|
}
|
|
LoadObserver::Load tested_load_;
|
test::DelayedEncoder encoder_;
|
} test(tested_load, encode_delay_ms);
|
|
RunBaseTest(&test);
|
}
|
|
TEST_F(CallPerfTest, ReceivesCpuUnderuse) {
|
const int kEncodeDelayMs = 2;
|
TestCpuOveruse(LoadObserver::kUnderuse, kEncodeDelayMs);
|
}
|
|
TEST_F(CallPerfTest, ReceivesCpuOveruse) {
|
const int kEncodeDelayMs = 35;
|
TestCpuOveruse(LoadObserver::kOveruse, kEncodeDelayMs);
|
}
|
|
void CallPerfTest::TestMinTransmitBitrate(bool pad_to_min_bitrate) {
|
static const int kMaxEncodeBitrateKbps = 30;
|
static const int kMinTransmitBitrateBps = 150000;
|
static const int kMinAcceptableTransmitBitrate = 130;
|
static const int kMaxAcceptableTransmitBitrate = 170;
|
static const int kNumBitrateObservationsInRange = 100;
|
static const int kAcceptableBitrateErrorMargin = 15; // +- 7
|
class BitrateObserver : public test::EndToEndTest {
|
public:
|
explicit BitrateObserver(bool using_min_transmit_bitrate)
|
: EndToEndTest(kLongTimeoutMs),
|
send_stream_(nullptr),
|
pad_to_min_bitrate_(using_min_transmit_bitrate),
|
num_bitrate_observations_in_range_(0) {}
|
|
private:
|
// TODO(holmer): Run this with a timer instead of once per packet.
|
Action OnSendRtp(const uint8_t* packet, size_t length) override {
|
VideoSendStream::Stats stats = send_stream_->GetStats();
|
if (stats.substreams.size() > 0) {
|
RTC_DCHECK_EQ(1u, stats.substreams.size());
|
int bitrate_kbps =
|
stats.substreams.begin()->second.total_bitrate_bps / 1000;
|
if (bitrate_kbps > 0) {
|
test::PrintResult(
|
"bitrate_stats_",
|
(pad_to_min_bitrate_ ? "min_transmit_bitrate"
|
: "without_min_transmit_bitrate"),
|
"bitrate_kbps",
|
static_cast<size_t>(bitrate_kbps),
|
"kbps",
|
false);
|
if (pad_to_min_bitrate_) {
|
if (bitrate_kbps > kMinAcceptableTransmitBitrate &&
|
bitrate_kbps < kMaxAcceptableTransmitBitrate) {
|
++num_bitrate_observations_in_range_;
|
}
|
} else {
|
// Expect bitrate stats to roughly match the max encode bitrate.
|
if (bitrate_kbps > (kMaxEncodeBitrateKbps -
|
kAcceptableBitrateErrorMargin / 2) &&
|
bitrate_kbps < (kMaxEncodeBitrateKbps +
|
kAcceptableBitrateErrorMargin / 2)) {
|
++num_bitrate_observations_in_range_;
|
}
|
}
|
if (num_bitrate_observations_in_range_ ==
|
kNumBitrateObservationsInRange)
|
observation_complete_.Set();
|
}
|
}
|
return SEND_PACKET;
|
}
|
|
void OnVideoStreamsCreated(
|
VideoSendStream* send_stream,
|
const std::vector<VideoReceiveStream*>& receive_streams) override {
|
send_stream_ = send_stream;
|
}
|
|
void ModifyVideoConfigs(
|
VideoSendStream::Config* send_config,
|
std::vector<VideoReceiveStream::Config>* receive_configs,
|
VideoEncoderConfig* encoder_config) override {
|
if (pad_to_min_bitrate_) {
|
encoder_config->min_transmit_bitrate_bps = kMinTransmitBitrateBps;
|
} else {
|
RTC_DCHECK_EQ(0, encoder_config->min_transmit_bitrate_bps);
|
}
|
}
|
|
void PerformTest() override {
|
EXPECT_TRUE(Wait()) << "Timeout while waiting for send-bitrate stats.";
|
}
|
|
VideoSendStream* send_stream_;
|
const bool pad_to_min_bitrate_;
|
int num_bitrate_observations_in_range_;
|
} test(pad_to_min_bitrate);
|
|
fake_encoder_.SetMaxBitrate(kMaxEncodeBitrateKbps);
|
RunBaseTest(&test);
|
}
|
|
TEST_F(CallPerfTest, PadsToMinTransmitBitrate) { TestMinTransmitBitrate(true); }
|
|
TEST_F(CallPerfTest, NoPadWithoutMinTransmitBitrate) {
|
TestMinTransmitBitrate(false);
|
}
|
|
TEST_F(CallPerfTest, KeepsHighBitrateWhenReconfiguringSender) {
|
static const uint32_t kInitialBitrateKbps = 400;
|
static const uint32_t kReconfigureThresholdKbps = 600;
|
static const uint32_t kPermittedReconfiguredBitrateDiffKbps = 100;
|
|
class BitrateObserver : public test::EndToEndTest, public test::FakeEncoder {
|
public:
|
BitrateObserver()
|
: EndToEndTest(kDefaultTimeoutMs),
|
FakeEncoder(Clock::GetRealTimeClock()),
|
time_to_reconfigure_(false, false),
|
encoder_inits_(0),
|
last_set_bitrate_(0),
|
send_stream_(nullptr) {}
|
|
int32_t InitEncode(const VideoCodec* config,
|
int32_t number_of_cores,
|
size_t max_payload_size) override {
|
if (encoder_inits_ == 0) {
|
EXPECT_EQ(kInitialBitrateKbps, config->startBitrate)
|
<< "Encoder not initialized at expected bitrate.";
|
}
|
++encoder_inits_;
|
if (encoder_inits_ == 2) {
|
EXPECT_GE(last_set_bitrate_, kReconfigureThresholdKbps);
|
EXPECT_NEAR(config->startBitrate,
|
last_set_bitrate_,
|
kPermittedReconfiguredBitrateDiffKbps)
|
<< "Encoder reconfigured with bitrate too far away from last set.";
|
observation_complete_.Set();
|
}
|
return FakeEncoder::InitEncode(config, number_of_cores, max_payload_size);
|
}
|
|
int32_t SetRates(uint32_t new_target_bitrate_kbps,
|
uint32_t framerate) override {
|
last_set_bitrate_ = new_target_bitrate_kbps;
|
if (encoder_inits_ == 1 &&
|
new_target_bitrate_kbps > kReconfigureThresholdKbps) {
|
time_to_reconfigure_.Set();
|
}
|
return FakeEncoder::SetRates(new_target_bitrate_kbps, framerate);
|
}
|
|
Call::Config GetSenderCallConfig() override {
|
Call::Config config = EndToEndTest::GetSenderCallConfig();
|
config.bitrate_config.start_bitrate_bps = kInitialBitrateKbps * 1000;
|
return config;
|
}
|
|
void ModifyVideoConfigs(
|
VideoSendStream::Config* send_config,
|
std::vector<VideoReceiveStream::Config>* receive_configs,
|
VideoEncoderConfig* encoder_config) override {
|
send_config->encoder_settings.encoder = this;
|
encoder_config->streams[0].min_bitrate_bps = 50000;
|
encoder_config->streams[0].target_bitrate_bps =
|
encoder_config->streams[0].max_bitrate_bps = 2000000;
|
|
encoder_config_ = *encoder_config;
|
}
|
|
void OnVideoStreamsCreated(
|
VideoSendStream* send_stream,
|
const std::vector<VideoReceiveStream*>& receive_streams) override {
|
send_stream_ = send_stream;
|
}
|
|
void PerformTest() override {
|
ASSERT_TRUE(time_to_reconfigure_.Wait(kDefaultTimeoutMs))
|
<< "Timed out before receiving an initial high bitrate.";
|
encoder_config_.streams[0].width *= 2;
|
encoder_config_.streams[0].height *= 2;
|
EXPECT_TRUE(send_stream_->ReconfigureVideoEncoder(encoder_config_));
|
EXPECT_TRUE(Wait())
|
<< "Timed out while waiting for a couple of high bitrate estimates "
|
"after reconfiguring the send stream.";
|
}
|
|
private:
|
rtc::Event time_to_reconfigure_;
|
int encoder_inits_;
|
uint32_t last_set_bitrate_;
|
VideoSendStream* send_stream_;
|
VideoEncoderConfig encoder_config_;
|
} test;
|
|
RunBaseTest(&test);
|
}
|
|
} // namespace webrtc
|
