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webrtc / src / webrtc / b6a2c8fa134f71061ff415b2ee3d32f88f99f712 / . / video / video_quality_test.cc
blob: 86d61d6dfd1131ed82dfe5e0290feac33dc7512f [file] [log] [blame]
/*
* Copyright (c) 2015 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/video/video_quality_test.h"
#include <stdio.h>
#include <algorithm>
#include <deque>
#include <map>
#include <set>
#include <sstream>
#include <string>
#include <vector>
#include "webrtc/base/checks.h"
#include "webrtc/base/cpu_time.h"
#include "webrtc/base/event.h"
#include "webrtc/base/format_macros.h"
#include "webrtc/base/memory_usage.h"
#include "webrtc/base/optional.h"
#include "webrtc/base/platform_file.h"
#include "webrtc/base/timeutils.h"
#include "webrtc/call/call.h"
#include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
#include "webrtc/logging/rtc_event_log/rtc_event_log.h"
#include "webrtc/modules/audio_mixer/audio_mixer_impl.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_format.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
#include "webrtc/modules/video_coding/codecs/h264/include/h264.h"
#include "webrtc/modules/video_coding/codecs/vp8/include/vp8.h"
#include "webrtc/modules/video_coding/codecs/vp8/include/vp8_common_types.h"
#include "webrtc/modules/video_coding/codecs/vp9/include/vp9.h"
#include "webrtc/system_wrappers/include/cpu_info.h"
#include "webrtc/system_wrappers/include/field_trial.h"
#include "webrtc/test/gtest.h"
#include "webrtc/test/layer_filtering_transport.h"
#include "webrtc/test/run_loop.h"
#include "webrtc/test/statistics.h"
#include "webrtc/test/testsupport/fileutils.h"
#include "webrtc/test/vcm_capturer.h"
#include "webrtc/test/video_renderer.h"
#include "webrtc/voice_engine/include/voe_base.h"
namespace {
constexpr int kSendStatsPollingIntervalMs = 1000;
constexpr int kPayloadTypeH264 = 122;
constexpr int kPayloadTypeVP8 = 123;
constexpr int kPayloadTypeVP9 = 124;
constexpr size_t kMaxComparisons = 10;
constexpr char kSyncGroup[] = "av_sync";
constexpr int kOpusMinBitrateBps = 6000;
constexpr int kOpusBitrateFbBps = 32000;
constexpr int kFramesSentInQuickTest = 1;
constexpr uint32_t kThumbnailSendSsrcStart = 0xE0000;
constexpr uint32_t kThumbnailRtxSsrcStart = 0xF0000;
struct VoiceEngineState {
VoiceEngineState()
: voice_engine(nullptr),
base(nullptr),
send_channel_id(-1),
receive_channel_id(-1) {}
webrtc::VoiceEngine* voice_engine;
webrtc::VoEBase* base;
int send_channel_id;
int receive_channel_id;
};
void CreateVoiceEngine(VoiceEngineState* voe,
rtc::scoped_refptr<webrtc::AudioDecoderFactory>
decoder_factory) {
voe->voice_engine = webrtc::VoiceEngine::Create();
voe->base = webrtc::VoEBase::GetInterface(voe->voice_engine);
EXPECT_EQ(0, voe->base->Init(nullptr, nullptr, decoder_factory));
webrtc::VoEBase::ChannelConfig config;
config.enable_voice_pacing = true;
voe->send_channel_id = voe->base->CreateChannel(config);
EXPECT_GE(voe->send_channel_id, 0);
voe->receive_channel_id = voe->base->CreateChannel();
EXPECT_GE(voe->receive_channel_id, 0);
}
void DestroyVoiceEngine(VoiceEngineState* voe) {
voe->base->DeleteChannel(voe->send_channel_id);
voe->send_channel_id = -1;
voe->base->DeleteChannel(voe->receive_channel_id);
voe->receive_channel_id = -1;
voe->base->Release();
voe->base = nullptr;
webrtc::VoiceEngine::Delete(voe->voice_engine);
voe->voice_engine = nullptr;
}
class VideoStreamFactory
: public webrtc::VideoEncoderConfig::VideoStreamFactoryInterface {
public:
explicit VideoStreamFactory(const std::vector<webrtc::VideoStream>& streams)
: streams_(streams) {}
private:
std::vector<webrtc::VideoStream> CreateEncoderStreams(
int width,
int height,
const webrtc::VideoEncoderConfig& encoder_config) override {
// The highest layer must match the incoming resolution.
std::vector<webrtc::VideoStream> streams = streams_;
streams[streams_.size() - 1].height = height;
streams[streams_.size() - 1].width = width;
return streams;
}
std::vector<webrtc::VideoStream> streams_;
};
bool IsFlexfec(int payload_type) {
return payload_type == webrtc::VideoQualityTest::kFlexfecPayloadType;
}
} // namespace
namespace webrtc {
class VideoAnalyzer : public PacketReceiver,
public Transport,
public rtc::VideoSinkInterface<VideoFrame> {
public:
VideoAnalyzer(test::LayerFilteringTransport* transport,
const std::string& test_label,
double avg_psnr_threshold,
double avg_ssim_threshold,
int duration_frames,
FILE* graph_data_output_file,
const std::string& graph_title,
uint32_t ssrc_to_analyze,
uint32_t rtx_ssrc_to_analyze,
size_t selected_stream,
int selected_sl,
int selected_tl,
bool is_quick_test_enabled)
: transport_(transport),
receiver_(nullptr),
send_stream_(nullptr),
receive_stream_(nullptr),
captured_frame_forwarder_(this),
test_label_(test_label),
graph_data_output_file_(graph_data_output_file),
graph_title_(graph_title),
ssrc_to_analyze_(ssrc_to_analyze),
rtx_ssrc_to_analyze_(rtx_ssrc_to_analyze),
selected_stream_(selected_stream),
selected_sl_(selected_sl),
selected_tl_(selected_tl),
pre_encode_proxy_(this),
encode_timing_proxy_(this),
frames_to_process_(duration_frames),
frames_recorded_(0),
frames_processed_(0),
dropped_frames_(0),
dropped_frames_before_first_encode_(0),
dropped_frames_before_rendering_(0),
last_render_time_(0),
rtp_timestamp_delta_(0),
total_media_bytes_(0),
first_sending_time_(0),
last_sending_time_(0),
cpu_time_(0),
wallclock_time_(0),
avg_psnr_threshold_(avg_psnr_threshold),
avg_ssim_threshold_(avg_ssim_threshold),
is_quick_test_enabled_(is_quick_test_enabled),
stats_polling_thread_(&PollStatsThread, this, "StatsPoller"),
comparison_available_event_(false, false),
done_(true, false) {
// Create thread pool for CPU-expensive PSNR/SSIM calculations.
// Try to use about as many threads as cores, but leave kMinCoresLeft alone,
// so that we don't accidentally starve "real" worker threads (codec etc).
// Also, don't allocate more than kMaxComparisonThreads, even if there are
// spare cores.
uint32_t num_cores = CpuInfo::DetectNumberOfCores();
RTC_DCHECK_GE(num_cores, 1);
static const uint32_t kMinCoresLeft = 4;
static const uint32_t kMaxComparisonThreads = 8;
if (num_cores <= kMinCoresLeft) {
num_cores = 1;
} else {
num_cores -= kMinCoresLeft;
num_cores = std::min(num_cores, kMaxComparisonThreads);
}
for (uint32_t i = 0; i < num_cores; ++i) {
rtc::PlatformThread* thread =
new rtc::PlatformThread(&FrameComparisonThread, this, "Analyzer");
thread->Start();
comparison_thread_pool_.push_back(thread);
}
}
~VideoAnalyzer() {
for (rtc::PlatformThread* thread : comparison_thread_pool_) {
thread->Stop();
delete thread;
}
}
virtual void SetReceiver(PacketReceiver* receiver) { receiver_ = receiver; }
void SetSendStream(VideoSendStream* stream) {
rtc::CritScope lock(&crit_);
RTC_DCHECK(!send_stream_);
send_stream_ = stream;
}
void SetReceiveStream(VideoReceiveStream* stream) {
rtc::CritScope lock(&crit_);
RTC_DCHECK(!receive_stream_);
receive_stream_ = stream;
}
rtc::VideoSinkInterface<VideoFrame>* InputInterface() {
return &captured_frame_forwarder_;
}
rtc::VideoSourceInterface<VideoFrame>* OutputInterface() {
return &captured_frame_forwarder_;
}
DeliveryStatus DeliverPacket(MediaType media_type,
const uint8_t* packet,
size_t length,
const PacketTime& packet_time) override {
// Ignore timestamps of RTCP packets. They're not synchronized with
// RTP packet timestamps and so they would confuse wrap_handler_.
if (RtpHeaderParser::IsRtcp(packet, length)) {
return receiver_->DeliverPacket(media_type, packet, length, packet_time);
}
RtpUtility::RtpHeaderParser parser(packet, length);
RTPHeader header;
parser.Parse(&header);
if (!IsFlexfec(header.payloadType) &&
(header.ssrc == ssrc_to_analyze_ ||
header.ssrc == rtx_ssrc_to_analyze_)) {
// Ignore FlexFEC timestamps, to avoid collisions with media timestamps.
// (FlexFEC and media are sent on different SSRCs, which have different
// timestamps spaces.)
// Also ignore packets from wrong SSRC, but include retransmits.
rtc::CritScope lock(&crit_);
int64_t timestamp =
wrap_handler_.Unwrap(header.timestamp - rtp_timestamp_delta_);
recv_times_[timestamp] =
Clock::GetRealTimeClock()->CurrentNtpInMilliseconds();
}
return receiver_->DeliverPacket(media_type, packet, length, packet_time);
}
void MeasuredEncodeTiming(int64_t ntp_time_ms, int encode_time_ms) {
rtc::CritScope crit(&comparison_lock_);
samples_encode_time_ms_[ntp_time_ms] = encode_time_ms;
}
void PreEncodeOnFrame(const VideoFrame& video_frame) {
rtc::CritScope lock(&crit_);
if (!first_encoded_timestamp_) {
while (frames_.front().timestamp() != video_frame.timestamp()) {
++dropped_frames_before_first_encode_;
frames_.pop_front();
RTC_CHECK(!frames_.empty());
}
first_encoded_timestamp_ =
rtc::Optional<uint32_t>(video_frame.timestamp());
}
}
void PostEncodeFrameCallback(const EncodedFrame& encoded_frame) {
rtc::CritScope lock(&crit_);
if (!first_sent_timestamp_ &&
encoded_frame.stream_id_ == selected_stream_) {
first_sent_timestamp_ = rtc::Optional<uint32_t>(encoded_frame.timestamp_);
}
}
bool SendRtp(const uint8_t* packet,
size_t length,
const PacketOptions& options) override {
RtpUtility::RtpHeaderParser parser(packet, length);
RTPHeader header;
parser.Parse(&header);
int64_t current_time =
Clock::GetRealTimeClock()->CurrentNtpInMilliseconds();
bool result = transport_->SendRtp(packet, length, options);
{
rtc::CritScope lock(&crit_);
if (rtp_timestamp_delta_ == 0 && header.ssrc == ssrc_to_analyze_) {
RTC_CHECK(static_cast<bool>(first_sent_timestamp_));
rtp_timestamp_delta_ = header.timestamp - *first_sent_timestamp_;
}
if (!IsFlexfec(header.payloadType) && header.ssrc == ssrc_to_analyze_) {
// Ignore FlexFEC timestamps, to avoid collisions with media timestamps.
// (FlexFEC and media are sent on different SSRCs, which have different
// timestamps spaces.)
// Also ignore packets from wrong SSRC and retransmits.
int64_t timestamp =
wrap_handler_.Unwrap(header.timestamp - rtp_timestamp_delta_);
send_times_[timestamp] = current_time;
if (IsInSelectedSpatialAndTemporalLayer(packet, length, header)) {
encoded_frame_sizes_[timestamp] +=
length - (header.headerLength + header.paddingLength);
total_media_bytes_ +=
length - (header.headerLength + header.paddingLength);
}
if (first_sending_time_ == 0)
first_sending_time_ = current_time;
last_sending_time_ = current_time;
}
}
return result;
}
bool SendRtcp(const uint8_t* packet, size_t length) override {
return transport_->SendRtcp(packet, length);
}
void OnFrame(const VideoFrame& video_frame) override {
int64_t render_time_ms =
Clock::GetRealTimeClock()->CurrentNtpInMilliseconds();
rtc::CritScope lock(&crit_);
StartExcludingCpuThreadTime();
int64_t send_timestamp =
wrap_handler_.Unwrap(video_frame.timestamp() - rtp_timestamp_delta_);
while (wrap_handler_.Unwrap(frames_.front().timestamp()) < send_timestamp) {
if (!last_rendered_frame_) {
// No previous frame rendered, this one was dropped after sending but
// before rendering.
++dropped_frames_before_rendering_;
} else {
AddFrameComparison(frames_.front(), *last_rendered_frame_, true,
render_time_ms);
}
frames_.pop_front();
RTC_DCHECK(!frames_.empty());
}
VideoFrame reference_frame = frames_.front();
frames_.pop_front();
int64_t reference_timestamp =
wrap_handler_.Unwrap(reference_frame.timestamp());
if (send_timestamp == reference_timestamp - 1) {
// TODO(ivica): Make this work for > 2 streams.
// Look at RTPSender::BuildRTPHeader.
++send_timestamp;
}
ASSERT_EQ(reference_timestamp, send_timestamp);
AddFrameComparison(reference_frame, video_frame, false, render_time_ms);
last_rendered_frame_ = rtc::Optional<VideoFrame>(video_frame);
StopExcludingCpuThreadTime();
}
void Wait() {
// Frame comparisons can be very expensive. Wait for test to be done, but
// at time-out check if frames_processed is going up. If so, give it more
// time, otherwise fail. Hopefully this will reduce test flakiness.
stats_polling_thread_.Start();
int last_frames_processed = -1;
int iteration = 0;
while (!done_.Wait(VideoQualityTest::kDefaultTimeoutMs)) {
int frames_processed;
{
rtc::CritScope crit(&comparison_lock_);
frames_processed = frames_processed_;
}
// Print some output so test infrastructure won't think we've crashed.
const char* kKeepAliveMessages[3] = {
"Uh, I'm-I'm not quite dead, sir.",
"Uh, I-I think uh, I could pull through, sir.",
"Actually, I think I'm all right to come with you--"};
printf("- %s\n", kKeepAliveMessages[iteration++ % 3]);
if (last_frames_processed == -1) {
last_frames_processed = frames_processed;
continue;
}
if (frames_processed == last_frames_processed) {
EXPECT_GT(frames_processed, last_frames_processed)
<< "Analyzer stalled while waiting for test to finish.";
done_.Set();
break;
}
last_frames_processed = frames_processed;
}
if (iteration > 0)
printf("- Farewell, sweet Concorde!\n");
stats_polling_thread_.Stop();
}
rtc::VideoSinkInterface<VideoFrame>* pre_encode_proxy() {
return &pre_encode_proxy_;
}
EncodedFrameObserver* encode_timing_proxy() { return &encode_timing_proxy_; }
void StartMeasuringCpuProcessTime() {
rtc::CritScope lock(&cpu_measurement_lock_);
cpu_time_ -= rtc::GetProcessCpuTimeNanos();
wallclock_time_ -= rtc::SystemTimeNanos();
}
void StopMeasuringCpuProcessTime() {
rtc::CritScope lock(&cpu_measurement_lock_);
cpu_time_ += rtc::GetProcessCpuTimeNanos();
wallclock_time_ += rtc::SystemTimeNanos();
}
void StartExcludingCpuThreadTime() {
rtc::CritScope lock(&cpu_measurement_lock_);
cpu_time_ += rtc::GetThreadCpuTimeNanos();
}
void StopExcludingCpuThreadTime() {
rtc::CritScope lock(&cpu_measurement_lock_);
cpu_time_ -= rtc::GetThreadCpuTimeNanos();
}
double GetCpuUsagePercent() {
rtc::CritScope lock(&cpu_measurement_lock_);
return static_cast<double>(cpu_time_) / wallclock_time_ * 100.0;
}
test::LayerFilteringTransport* const transport_;
PacketReceiver* receiver_;
private:
struct FrameComparison {
FrameComparison()
: dropped(false),
input_time_ms(0),
send_time_ms(0),
recv_time_ms(0),
render_time_ms(0),
encoded_frame_size(0) {}
FrameComparison(const VideoFrame& reference,
const VideoFrame& render,
bool dropped,
int64_t input_time_ms,
int64_t send_time_ms,
int64_t recv_time_ms,
int64_t render_time_ms,
size_t encoded_frame_size)
: reference(reference),
render(render),
dropped(dropped),
input_time_ms(input_time_ms),
send_time_ms(send_time_ms),
recv_time_ms(recv_time_ms),
render_time_ms(render_time_ms),
encoded_frame_size(encoded_frame_size) {}
FrameComparison(bool dropped,
int64_t input_time_ms,
int64_t send_time_ms,
int64_t recv_time_ms,
int64_t render_time_ms,
size_t encoded_frame_size)
: dropped(dropped),
input_time_ms(input_time_ms),
send_time_ms(send_time_ms),
recv_time_ms(recv_time_ms),
render_time_ms(render_time_ms),
encoded_frame_size(encoded_frame_size) {}
rtc::Optional<VideoFrame> reference;
rtc::Optional<VideoFrame> render;
bool dropped;
int64_t input_time_ms;
int64_t send_time_ms;
int64_t recv_time_ms;
int64_t render_time_ms;
size_t encoded_frame_size;
};
struct Sample {
Sample(int dropped,
int64_t input_time_ms,
int64_t send_time_ms,
int64_t recv_time_ms,
int64_t render_time_ms,
size_t encoded_frame_size,
double psnr,
double ssim)
: dropped(dropped),
input_time_ms(input_time_ms),
send_time_ms(send_time_ms),
recv_time_ms(recv_time_ms),
render_time_ms(render_time_ms),
encoded_frame_size(encoded_frame_size),
psnr(psnr),
ssim(ssim) {}
int dropped;
int64_t input_time_ms;
int64_t send_time_ms;
int64_t recv_time_ms;
int64_t render_time_ms;
size_t encoded_frame_size;
double psnr;
double ssim;
};
// This class receives the send-side OnEncodeTiming and is provided to not
// conflict with the receiver-side pre_decode_callback.
class OnEncodeTimingProxy : public EncodedFrameObserver {
public:
explicit OnEncodeTimingProxy(VideoAnalyzer* parent) : parent_(parent) {}
void OnEncodeTiming(int64_t ntp_time_ms, int encode_time_ms) override {
parent_->MeasuredEncodeTiming(ntp_time_ms, encode_time_ms);
}
void EncodedFrameCallback(const EncodedFrame& frame) override {
parent_->PostEncodeFrameCallback(frame);
}
private:
VideoAnalyzer* const parent_;
};
// This class receives the send-side OnFrame callback and is provided to not
// conflict with the receiver-side renderer callback.
class PreEncodeProxy : public rtc::VideoSinkInterface<VideoFrame> {
public:
explicit PreEncodeProxy(VideoAnalyzer* parent) : parent_(parent) {}
void OnFrame(const VideoFrame& video_frame) override {
parent_->PreEncodeOnFrame(video_frame);
}
private:
VideoAnalyzer* const parent_;
};
bool IsInSelectedSpatialAndTemporalLayer(const uint8_t* packet,
size_t length,
const RTPHeader& header) {
if (header.payloadType != kPayloadTypeVP9 &&
header.payloadType != kPayloadTypeVP8) {
return true;
} else {
// Get VP8 and VP9 specific header to check layers indexes.
const uint8_t* payload = packet + header.headerLength;
const size_t payload_length = length - header.headerLength;
const size_t payload_data_length = payload_length - header.paddingLength;
const bool is_vp8 = header.payloadType == kPayloadTypeVP8;
std::unique_ptr<RtpDepacketizer> depacketizer(
RtpDepacketizer::Create(is_vp8 ? kRtpVideoVp8 : kRtpVideoVp9));
RtpDepacketizer::ParsedPayload parsed_payload;
bool result =
depacketizer->Parse(&parsed_payload, payload, payload_data_length);
RTC_DCHECK(result);
const int temporal_idx = static_cast<int>(
is_vp8 ? parsed_payload.type.Video.codecHeader.VP8.temporalIdx
: parsed_payload.type.Video.codecHeader.VP9.temporal_idx);
const int spatial_idx = static_cast<int>(
is_vp8 ? kNoSpatialIdx
: parsed_payload.type.Video.codecHeader.VP9.spatial_idx);
return (selected_tl_ < 0 || temporal_idx == kNoTemporalIdx ||
temporal_idx <= selected_tl_) &&
(selected_sl_ < 0 || spatial_idx == kNoSpatialIdx ||
spatial_idx <= selected_sl_);
}
}
void AddFrameComparison(const VideoFrame& reference,
const VideoFrame& render,
bool dropped,
int64_t render_time_ms)
EXCLUSIVE_LOCKS_REQUIRED(crit_) {
int64_t reference_timestamp = wrap_handler_.Unwrap(reference.timestamp());
int64_t send_time_ms = send_times_[reference_timestamp];
send_times_.erase(reference_timestamp);
int64_t recv_time_ms = recv_times_[reference_timestamp];
recv_times_.erase(reference_timestamp);
// TODO(ivica): Make this work for > 2 streams.
auto it = encoded_frame_sizes_.find(reference_timestamp);
if (it == encoded_frame_sizes_.end())
it = encoded_frame_sizes_.find(reference_timestamp - 1);
size_t encoded_size = it == encoded_frame_sizes_.end() ? 0 : it->second;
if (it != encoded_frame_sizes_.end())
encoded_frame_sizes_.erase(it);
rtc::CritScope crit(&comparison_lock_);
if (comparisons_.size() < kMaxComparisons) {
comparisons_.push_back(FrameComparison(reference, render, dropped,
reference.ntp_time_ms(),
send_time_ms, recv_time_ms,
render_time_ms, encoded_size));
} else {
comparisons_.push_back(FrameComparison(dropped,
reference.ntp_time_ms(),
send_time_ms, recv_time_ms,
render_time_ms, encoded_size));
}
comparison_available_event_.Set();
}
static void PollStatsThread(void* obj) {
static_cast<VideoAnalyzer*>(obj)->PollStats();
}
void PollStats() {
while (!done_.Wait(kSendStatsPollingIntervalMs)) {
rtc::CritScope crit(&comparison_lock_);
VideoSendStream::Stats send_stats = send_stream_->GetStats();
// It's not certain that we yet have estimates for any of these stats.
// Check that they are positive before mixing them in.
if (send_stats.encode_frame_rate > 0)
encode_frame_rate_.AddSample(send_stats.encode_frame_rate);
if (send_stats.avg_encode_time_ms > 0)
encode_time_ms_.AddSample(send_stats.avg_encode_time_ms);
if (send_stats.encode_usage_percent > 0)
encode_usage_percent_.AddSample(send_stats.encode_usage_percent);
if (send_stats.media_bitrate_bps > 0)
media_bitrate_bps_.AddSample(send_stats.media_bitrate_bps);
if (receive_stream_ != nullptr) {
VideoReceiveStream::Stats receive_stats = receive_stream_->GetStats();
if (receive_stats.decode_ms > 0)
decode_time_ms_.AddSample(receive_stats.decode_ms);
if (receive_stats.max_decode_ms > 0)
decode_time_max_ms_.AddSample(receive_stats.max_decode_ms);
}
memory_usage_.AddSample(rtc::GetProcessResidentSizeBytes());
}
}
static bool FrameComparisonThread(void* obj) {
return static_cast<VideoAnalyzer*>(obj)->CompareFrames();
}
bool CompareFrames() {
if (AllFramesRecorded())
return false;
FrameComparison comparison;
if (!PopComparison(&comparison)) {
// Wait until new comparison task is available, or test is done.
// If done, wake up remaining threads waiting.
comparison_available_event_.Wait(1000);
if (AllFramesRecorded()) {
comparison_available_event_.Set();
return false;
}
return true; // Try again.
}
StartExcludingCpuThreadTime();
PerformFrameComparison(comparison);
StopExcludingCpuThreadTime();
if (FrameProcessed()) {
PrintResults();
if (graph_data_output_file_)
PrintSamplesToFile();
done_.Set();
comparison_available_event_.Set();
return false;
}
return true;
}
bool PopComparison(FrameComparison* comparison) {
rtc::CritScope crit(&comparison_lock_);
// If AllFramesRecorded() is true, it means we have already popped
// frames_to_process_ frames from comparisons_, so there is no more work
// for this thread to be done. frames_processed_ might still be lower if
// all comparisons are not done, but those frames are currently being
// worked on by other threads.
if (comparisons_.empty() || AllFramesRecorded())
return false;
*comparison = comparisons_.front();
comparisons_.pop_front();
FrameRecorded();
return true;
}
// Increment counter for number of frames received for comparison.
void FrameRecorded() {
rtc::CritScope crit(&comparison_lock_);
++frames_recorded_;
}
// Returns true if all frames to be compared have been taken from the queue.
bool AllFramesRecorded() {
rtc::CritScope crit(&comparison_lock_);
assert(frames_recorded_ <= frames_to_process_);
return frames_recorded_ == frames_to_process_;
}
// Increase count of number of frames processed. Returns true if this was the
// last frame to be processed.
bool FrameProcessed() {
rtc::CritScope crit(&comparison_lock_);
++frames_processed_;
assert(frames_processed_ <= frames_to_process_);
return frames_processed_ == frames_to_process_;
}
void PrintResults() {
StopMeasuringCpuProcessTime();
rtc::CritScope crit(&comparison_lock_);
PrintResult("psnr", psnr_, " dB");
PrintResult("ssim", ssim_, " score");
PrintResult("sender_time", sender_time_, " ms");
PrintResult("receiver_time", receiver_time_, " ms");
PrintResult("total_delay_incl_network", end_to_end_, " ms");
PrintResult("time_between_rendered_frames", rendered_delta_, " ms");
PrintResult("encode_frame_rate", encode_frame_rate_, " fps");
PrintResult("encode_time", encode_time_ms_, " ms");
PrintResult("media_bitrate", media_bitrate_bps_, " bps");
if (receive_stream_ != nullptr) {
PrintResult("decode_time", decode_time_ms_, " ms");
}
printf("RESULT dropped_frames: %s = %d frames\n", test_label_.c_str(),
dropped_frames_);
printf("RESULT cpu_usage: %s = %lf %%\n", test_label_.c_str(),
GetCpuUsagePercent());
#if defined(WEBRTC_WIN)
// On Linux and Mac in Resident Set some unused pages may be counted.
// Therefore this metric will depend on order in which tests are run and
// will be flaky.
PrintResult("memory_usage", memory_usage_, " bytes");
#endif
// Disable quality check for quick test, as quality checks may fail
// because too few samples were collected.
if (!is_quick_test_enabled_) {
EXPECT_GT(psnr_.Mean(), avg_psnr_threshold_);
EXPECT_GT(ssim_.Mean(), avg_ssim_threshold_);
}
}
void PerformFrameComparison(const FrameComparison& comparison) {
// Perform expensive psnr and ssim calculations while not holding lock.
double psnr = -1.0;
double ssim = -1.0;
if (comparison.reference) {
psnr = I420PSNR(&*comparison.reference, &*comparison.render);
ssim = I420SSIM(&*comparison.reference, &*comparison.render);
}
rtc::CritScope crit(&comparison_lock_);
if (graph_data_output_file_) {
samples_.push_back(Sample(
comparison.dropped, comparison.input_time_ms, comparison.send_time_ms,
comparison.recv_time_ms, comparison.render_time_ms,
comparison.encoded_frame_size, psnr, ssim));
}
if (psnr >= 0.0)
psnr_.AddSample(psnr);
if (ssim >= 0.0)
ssim_.AddSample(ssim);
if (comparison.dropped) {
++dropped_frames_;
return;
}
if (last_render_time_ != 0)
rendered_delta_.AddSample(comparison.render_time_ms - last_render_time_);
last_render_time_ = comparison.render_time_ms;
sender_time_.AddSample(comparison.send_time_ms - comparison.input_time_ms);
if (comparison.recv_time_ms > 0) {
// If recv_time_ms == 0, this frame consisted of a packets which were all
// lost in the transport. Since we were able to render the frame, however,
// the dropped packets were recovered by FlexFEC. The FlexFEC recovery
// happens internally in Call, and we can therefore here not know which
// FEC packets that protected the lost media packets. Consequently, we
// were not able to record a meaningful recv_time_ms. We therefore skip
// this sample.
//
// The reasoning above does not hold for ULPFEC and RTX, as for those
// strategies the timestamp of the received packets is set to the
// timestamp of the protected/retransmitted media packet. I.e., then
// recv_time_ms != 0, even though the media packets were lost.
receiver_time_.AddSample(comparison.render_time_ms -
comparison.recv_time_ms);
}
end_to_end_.AddSample(comparison.render_time_ms - comparison.input_time_ms);
encoded_frame_size_.AddSample(comparison.encoded_frame_size);
}
void PrintResult(const char* result_type,
test::Statistics stats,
const char* unit) {
printf("RESULT %s: %s = {%f, %f}%s\n",
result_type,
test_label_.c_str(),
stats.Mean(),
stats.StandardDeviation(),
unit);
}
void PrintSamplesToFile(void) {
FILE* out = graph_data_output_file_;
rtc::CritScope crit(&comparison_lock_);
std::sort(samples_.begin(), samples_.end(),
[](const Sample& A, const Sample& B) -> bool {
return A.input_time_ms < B.input_time_ms;
});
fprintf(out, "%s\n", graph_title_.c_str());
fprintf(out, "%" PRIuS "\n", samples_.size());
fprintf(out,
"dropped "
"input_time_ms "
"send_time_ms "
"recv_time_ms "
"render_time_ms "
"encoded_frame_size "
"psnr "
"ssim "
"encode_time_ms\n");
int missing_encode_time_samples = 0;
for (const Sample& sample : samples_) {
auto it = samples_encode_time_ms_.find(sample.input_time_ms);
int encode_time_ms;
if (it != samples_encode_time_ms_.end()) {
encode_time_ms = it->second;
} else {
++missing_encode_time_samples;
encode_time_ms = -1;
}
fprintf(out, "%d %" PRId64 " %" PRId64 " %" PRId64 " %" PRId64 " %" PRIuS
" %lf %lf %d\n",
sample.dropped, sample.input_time_ms, sample.send_time_ms,
sample.recv_time_ms, sample.render_time_ms,
sample.encoded_frame_size, sample.psnr, sample.ssim,
encode_time_ms);
}
if (missing_encode_time_samples) {
fprintf(stderr,
"Warning: Missing encode_time_ms samples for %d frame(s).\n",
missing_encode_time_samples);
}
}
double GetAverageMediaBitrateBps() {
if (last_sending_time_ == first_sending_time_) {
return 0;
} else {
return static_cast<double>(total_media_bytes_) * 8 /
(last_sending_time_ - first_sending_time_) *
rtc::kNumMillisecsPerSec;
}
}
// Implements VideoSinkInterface to receive captured frames from a
// FrameGeneratorCapturer. Implements VideoSourceInterface to be able to act
// as a source to VideoSendStream.
// It forwards all input frames to the VideoAnalyzer for later comparison and
// forwards the captured frames to the VideoSendStream.
class CapturedFrameForwarder : public rtc::VideoSinkInterface<VideoFrame>,
public rtc::VideoSourceInterface<VideoFrame> {
public:
explicit CapturedFrameForwarder(VideoAnalyzer* analyzer)
: analyzer_(analyzer), send_stream_input_(nullptr) {}
private:
void OnFrame(const VideoFrame& video_frame) override {
VideoFrame copy = video_frame;
// Frames from the capturer does not have a rtp timestamp.
// Create one so it can be used for comparison.
RTC_DCHECK_EQ(0, video_frame.timestamp());
if (copy.ntp_time_ms() == 0)
copy.set_ntp_time_ms(rtc::TimeMillis());
copy.set_timestamp(copy.ntp_time_ms() * 90);
analyzer_->AddCapturedFrameForComparison(copy);
rtc::CritScope lock(&crit_);
if (send_stream_input_)
send_stream_input_->OnFrame(copy);
}
// Called when |send_stream_.SetSource()| is called.
void AddOrUpdateSink(rtc::VideoSinkInterface<VideoFrame>* sink,
const rtc::VideoSinkWants& wants) override {
rtc::CritScope lock(&crit_);
RTC_DCHECK(!send_stream_input_ || send_stream_input_ == sink);
send_stream_input_ = sink;
}
// Called by |send_stream_| when |send_stream_.SetSource()| is called.
void RemoveSink(rtc::VideoSinkInterface<VideoFrame>* sink) override {
rtc::CritScope lock(&crit_);
RTC_DCHECK(sink == send_stream_input_);
send_stream_input_ = nullptr;
}
VideoAnalyzer* const analyzer_;
rtc::CriticalSection crit_;
rtc::VideoSinkInterface<VideoFrame>* send_stream_input_ GUARDED_BY(crit_);
};
void AddCapturedFrameForComparison(const VideoFrame& video_frame) {
rtc::CritScope lock(&crit_);
frames_.push_back(video_frame);
}
VideoSendStream* send_stream_;
VideoReceiveStream* receive_stream_;
CapturedFrameForwarder captured_frame_forwarder_;
const std::string test_label_;
FILE* const graph_data_output_file_;
const std::string graph_title_;
const uint32_t ssrc_to_analyze_;
const uint32_t rtx_ssrc_to_analyze_;
const size_t selected_stream_;
const int selected_sl_;
const int selected_tl_;
PreEncodeProxy pre_encode_proxy_;
OnEncodeTimingProxy encode_timing_proxy_;
std::vector<Sample> samples_ GUARDED_BY(comparison_lock_);
std::map<int64_t, int> samples_encode_time_ms_ GUARDED_BY(comparison_lock_);
test::Statistics sender_time_ GUARDED_BY(comparison_lock_);
test::Statistics receiver_time_ GUARDED_BY(comparison_lock_);
test::Statistics psnr_ GUARDED_BY(comparison_lock_);
test::Statistics ssim_ GUARDED_BY(comparison_lock_);
test::Statistics end_to_end_ GUARDED_BY(comparison_lock_);
test::Statistics rendered_delta_ GUARDED_BY(comparison_lock_);
test::Statistics encoded_frame_size_ GUARDED_BY(comparison_lock_);
test::Statistics encode_frame_rate_ GUARDED_BY(comparison_lock_);
test::Statistics encode_time_ms_ GUARDED_BY(comparison_lock_);
test::Statistics encode_usage_percent_ GUARDED_BY(comparison_lock_);
test::Statistics decode_time_ms_ GUARDED_BY(comparison_lock_);
test::Statistics decode_time_max_ms_ GUARDED_BY(comparison_lock_);
test::Statistics media_bitrate_bps_ GUARDED_BY(comparison_lock_);
test::Statistics memory_usage_ GUARDED_BY(comparison_lock_);
const int frames_to_process_;
int frames_recorded_;
int frames_processed_;
int dropped_frames_;
int dropped_frames_before_first_encode_;
int dropped_frames_before_rendering_;
int64_t last_render_time_;
uint32_t rtp_timestamp_delta_;
int64_t total_media_bytes_;
int64_t first_sending_time_;
int64_t last_sending_time_;
int64_t cpu_time_ GUARDED_BY(cpu_measurement_lock_);
int64_t wallclock_time_ GUARDED_BY(cpu_measurement_lock_);
rtc::CriticalSection cpu_measurement_lock_;
rtc::CriticalSection crit_;
std::deque<VideoFrame> frames_ GUARDED_BY(crit_);
rtc::Optional<VideoFrame> last_rendered_frame_ GUARDED_BY(crit_);
rtc::TimestampWrapAroundHandler wrap_handler_ GUARDED_BY(crit_);
std::map<int64_t, int64_t> send_times_ GUARDED_BY(crit_);
std::map<int64_t, int64_t> recv_times_ GUARDED_BY(crit_);
std::map<int64_t, size_t> encoded_frame_sizes_ GUARDED_BY(crit_);
rtc::Optional<uint32_t> first_encoded_timestamp_ GUARDED_BY(crit_);
rtc::Optional<uint32_t> first_sent_timestamp_ GUARDED_BY(crit_);
const double avg_psnr_threshold_;
const double avg_ssim_threshold_;
bool is_quick_test_enabled_;
rtc::CriticalSection comparison_lock_;
std::vector<rtc::PlatformThread*> comparison_thread_pool_;
rtc::PlatformThread stats_polling_thread_;
rtc::Event comparison_available_event_;
std::deque<FrameComparison> comparisons_ GUARDED_BY(comparison_lock_);
rtc::Event done_;
};
class Vp8EncoderFactory : public VideoEncoderFactory {
public:
Vp8EncoderFactory() = default;
~Vp8EncoderFactory() override { RTC_CHECK(live_encoders_.empty()); }
VideoEncoder* Create() override {
VideoEncoder* encoder = VP8Encoder::Create();
live_encoders_.insert(encoder);
return encoder;
}
void Destroy(VideoEncoder* encoder) override {
auto it = live_encoders_.find(encoder);
RTC_CHECK(it != live_encoders_.end());
live_encoders_.erase(it);
delete encoder;
}
std::set<VideoEncoder*> live_encoders_;
};
VideoQualityTest::VideoQualityTest()
: clock_(Clock::GetRealTimeClock()), receive_logs_(0), send_logs_(0) {
payload_type_map_ = test::CallTest::payload_type_map_;
RTC_DCHECK(payload_type_map_.find(kPayloadTypeH264) ==
payload_type_map_.end());
RTC_DCHECK(payload_type_map_.find(kPayloadTypeVP8) ==
payload_type_map_.end());
RTC_DCHECK(payload_type_map_.find(kPayloadTypeVP9) ==
payload_type_map_.end());
payload_type_map_[kPayloadTypeH264] = webrtc::MediaType::VIDEO;
payload_type_map_[kPayloadTypeVP8] = webrtc::MediaType::VIDEO;
payload_type_map_[kPayloadTypeVP9] = webrtc::MediaType::VIDEO;
}
VideoQualityTest::Params::Params()
: call({false, Call::Config::BitrateConfig()}),
video({false, 640, 480, 30, 50, 800, 800, false, "VP8", 1, -1, 0, false,
false, "", ""}),
audio({false, false, false}),
screenshare({false, 10, 0}),
analyzer({"", 0.0, 0.0, 0, "", ""}),
pipe(),
logs(false),
ss({std::vector<VideoStream>(), 0, 0, -1, std::vector<SpatialLayer>()}),
num_thumbnails(0) {}
VideoQualityTest::Params::~Params() = default;
void VideoQualityTest::TestBody() {}
std::string VideoQualityTest::GenerateGraphTitle() const {
std::stringstream ss;
ss << params_.video.codec;
ss << " (" << params_.video.target_bitrate_bps / 1000 << "kbps";
ss << ", " << params_.video.fps << " FPS";
if (params_.screenshare.scroll_duration)
ss << ", " << params_.screenshare.scroll_duration << "s scroll";
if (params_.ss.streams.size() > 1)
ss << ", Stream #" << params_.ss.selected_stream;
if (params_.ss.num_spatial_layers > 1)
ss << ", Layer #" << params_.ss.selected_sl;
ss << ")";
return ss.str();
}
void VideoQualityTest::CheckParams() {
if (!params_.video.enabled)
return;
// Add a default stream in none specified.
if (params_.ss.streams.empty())
params_.ss.streams.push_back(VideoQualityTest::DefaultVideoStream(params_));
if (params_.ss.num_spatial_layers == 0)
params_.ss.num_spatial_layers = 1;
if (params_.pipe.loss_percent != 0 ||
params_.pipe.queue_length_packets != 0) {
// Since LayerFilteringTransport changes the sequence numbers, we can't
// use that feature with pack loss, since the NACK request would end up
// retransmitting the wrong packets.
RTC_CHECK(params_.ss.selected_sl == -1 ||
params_.ss.selected_sl == params_.ss.num_spatial_layers - 1);
RTC_CHECK(params_.video.selected_tl == -1 ||
params_.video.selected_tl ==
params_.video.num_temporal_layers - 1);
}
// TODO(ivica): Should max_bitrate_bps == -1 represent inf max bitrate, as it
// does in some parts of the code?
RTC_CHECK_GE(params_.video.max_bitrate_bps, params_.video.target_bitrate_bps);
RTC_CHECK_GE(params_.video.target_bitrate_bps, params_.video.min_bitrate_bps);
RTC_CHECK_LT(params_.video.selected_tl, params_.video.num_temporal_layers);
RTC_CHECK_LT(params_.ss.selected_stream, params_.ss.streams.size());
for (const VideoStream& stream : params_.ss.streams) {
RTC_CHECK_GE(stream.min_bitrate_bps, 0);
RTC_CHECK_GE(stream.target_bitrate_bps, stream.min_bitrate_bps);
RTC_CHECK_GE(stream.max_bitrate_bps, stream.target_bitrate_bps);
RTC_CHECK_LE(stream.temporal_layer_thresholds_bps.size(),
params_.video.num_temporal_layers - 1);
}
// TODO(ivica): Should we check if the sum of all streams/layers is equal to
// the total bitrate? We anyway have to update them in the case bitrate
// estimator changes the total bitrates.
RTC_CHECK_GE(params_.ss.num_spatial_layers, 1);
RTC_CHECK_LE(params_.ss.selected_sl, params_.ss.num_spatial_layers);
RTC_CHECK(params_.ss.spatial_layers.empty() ||
params_.ss.spatial_layers.size() ==
static_cast<size_t>(params_.ss.num_spatial_layers));
if (params_.video.codec == "VP8") {
RTC_CHECK_EQ(params_.ss.num_spatial_layers, 1);
} else if (params_.video.codec == "VP9") {
RTC_CHECK_EQ(params_.ss.streams.size(), 1);
}
RTC_CHECK_GE(params_.num_thumbnails, 0);
if (params_.num_thumbnails > 0) {
RTC_CHECK_EQ(params_.ss.num_spatial_layers, 1);
RTC_CHECK_EQ(params_.ss.streams.size(), 3);
RTC_CHECK_EQ(params_.video.num_temporal_layers, 3);
RTC_CHECK_EQ(params_.video.codec, "VP8");
}
}
// Static.
std::vector<int> VideoQualityTest::ParseCSV(const std::string& str) {
// Parse comma separated nonnegative integers, where some elements may be
// empty. The empty values are replaced with -1.
// E.g. "10,-20,,30,40" --> {10, 20, -1, 30,40}
// E.g. ",,10,,20," --> {-1, -1, 10, -1, 20, -1}
std::vector<int> result;
if (str.empty())
return result;
const char* p = str.c_str();
int value = -1;
int pos;
while (*p) {
if (*p == ',') {
result.push_back(value);
value = -1;
++p;
continue;
}
RTC_CHECK_EQ(sscanf(p, "%d%n", &value, &pos), 1)
<< "Unexpected non-number value.";
p += pos;
}
result.push_back(value);
return result;
}
// Static.
VideoStream VideoQualityTest::DefaultVideoStream(const Params& params) {
VideoStream stream;
stream.width = params.video.width;
stream.height = params.video.height;
stream.max_framerate = params.video.fps;
stream.min_bitrate_bps = params.video.min_bitrate_bps;
stream.target_bitrate_bps = params.video.target_bitrate_bps;
stream.max_bitrate_bps = params.video.max_bitrate_bps;
stream.max_qp = 52;
// TODO(sprang): Can we make this less of a hack?
if (params.video.num_temporal_layers == 2) {
stream.temporal_layer_thresholds_bps.push_back(stream.target_bitrate_bps);
} else if (params.video.num_temporal_layers == 3) {
stream.temporal_layer_thresholds_bps.push_back(stream.max_bitrate_bps / 4);
stream.temporal_layer_thresholds_bps.push_back(stream.target_bitrate_bps);
}
return stream;
}
// Static.
VideoStream VideoQualityTest::DefaultThumbnailStream() {
VideoStream stream;
stream.width = 320;
stream.height = 180;
stream.max_framerate = 7;
stream.min_bitrate_bps = 7500;
stream.target_bitrate_bps = 37500;
stream.max_bitrate_bps = 50000;
stream.max_qp = 52;
return stream;
}
// Static.
void VideoQualityTest::FillScalabilitySettings(
Params* params,
const std::vector<std::string>& stream_descriptors,
size_t selected_stream,
int num_spatial_layers,
int selected_sl,
const std::vector<std::string>& sl_descriptors) {
// Read VideoStream and SpatialLayer elements from a list of comma separated
// lists. To use a default value for an element, use -1 or leave empty.
// Validity checks performed in CheckParams.
RTC_CHECK(params->ss.streams.empty());
for (auto descriptor : stream_descriptors) {
if (descriptor.empty())
continue;
VideoStream stream = VideoQualityTest::DefaultVideoStream(*params);
std::vector<int> v = VideoQualityTest::ParseCSV(descriptor);
if (v[0] != -1)
stream.width = static_cast<size_t>(v[0]);
if (v[1] != -1)
stream.height = static_cast<size_t>(v[1]);
if (v[2] != -1)
stream.max_framerate = v[2];
if (v[3] != -1)
stream.min_bitrate_bps = v[3];
if (v[4] != -1)
stream.target_bitrate_bps = v[4];
if (v[5] != -1)
stream.max_bitrate_bps = v[5];
if (v.size() > 6 && v[6] != -1)
stream.max_qp = v[6];
if (v.size() > 7) {
stream.temporal_layer_thresholds_bps.clear();
stream.temporal_layer_thresholds_bps.insert(
stream.temporal_layer_thresholds_bps.end(), v.begin() + 7, v.end());
} else {
// Automatic TL thresholds for more than two layers not supported.
RTC_CHECK_LE(params->video.num_temporal_layers, 2);
}
params->ss.streams.push_back(stream);
}
params->ss.selected_stream = selected_stream;
params->ss.num_spatial_layers = num_spatial_layers ? num_spatial_layers : 1;
params->ss.selected_sl = selected_sl;
RTC_CHECK(params->ss.spatial_layers.empty());
for (auto descriptor : sl_descriptors) {
if (descriptor.empty())
continue;
std::vector<int> v = VideoQualityTest::ParseCSV(descriptor);
RTC_CHECK_GT(v[2], 0);
SpatialLayer layer;
layer.scaling_factor_num = v[0] == -1 ? 1 : v[0];
layer.scaling_factor_den = v[1] == -1 ? 1 : v[1];
layer.target_bitrate_bps = v[2];
params->ss.spatial_layers.push_back(layer);
}
}
void VideoQualityTest::SetupVideo(Transport* send_transport,
Transport* recv_transport) {
if (params_.logs)
trace_to_stderr_.reset(new test::TraceToStderr);
size_t num_video_streams = params_.ss.streams.size();
size_t num_flexfec_streams = params_.video.flexfec ? 1 : 0;
CreateSendConfig(num_video_streams, 0, num_flexfec_streams, send_transport);
int payload_type;
if (params_.video.codec == "H264") {
video_encoder_.reset(H264Encoder::Create(cricket::VideoCodec("H264")));
payload_type = kPayloadTypeH264;
} else if (params_.video.codec == "VP8") {
if (params_.screenshare.enabled && params_.ss.streams.size() > 1) {
// Simulcast screenshare needs a simulcast encoder adapter to work, since
// encoders usually can't natively do simulcast with different frame rates
// for the different layers.
video_encoder_.reset(
new SimulcastEncoderAdapter(new Vp8EncoderFactory()));
} else {
video_encoder_.reset(VP8Encoder::Create());
}
payload_type = kPayloadTypeVP8;
} else if (params_.video.codec == "VP9") {
video_encoder_.reset(VP9Encoder::Create());
payload_type = kPayloadTypeVP9;
} else {
RTC_NOTREACHED() << "Codec not supported!";
return;
}
video_send_config_.encoder_settings.encoder = video_encoder_.get();
video_send_config_.encoder_settings.payload_name = params_.video.codec;
video_send_config_.encoder_settings.payload_type = payload_type;
video_send_config_.rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
video_send_config_.rtp.rtx.payload_type = kSendRtxPayloadType;
for (size_t i = 0; i < num_video_streams; ++i)
video_send_config_.rtp.rtx.ssrcs.push_back(kSendRtxSsrcs[i]);
video_send_config_.rtp.extensions.clear();
if (params_.call.send_side_bwe) {
video_send_config_.rtp.extensions.push_back(
RtpExtension(RtpExtension::kTransportSequenceNumberUri,
test::kTransportSequenceNumberExtensionId));
} else {
video_send_config_.rtp.extensions.push_back(RtpExtension(
RtpExtension::kAbsSendTimeUri, test::kAbsSendTimeExtensionId));
}
video_send_config_.rtp.extensions.push_back(RtpExtension(
RtpExtension::kVideoContentTypeUri, test::kVideoContentTypeExtensionId));
video_encoder_config_.min_transmit_bitrate_bps =
params_.video.min_transmit_bps;
video_send_config_.suspend_below_min_bitrate =
params_.video.suspend_below_min_bitrate;
video_encoder_config_.number_of_streams = params_.ss.streams.size();
video_encoder_config_.max_bitrate_bps = 0;
for (size_t i = 0; i < params_.ss.streams.size(); ++i) {
video_encoder_config_.max_bitrate_bps +=
params_.ss.streams[i].max_bitrate_bps;
}
video_encoder_config_.video_stream_factory =
new rtc::RefCountedObject<VideoStreamFactory>(params_.ss.streams);
video_encoder_config_.spatial_layers = params_.ss.spatial_layers;
CreateMatchingReceiveConfigs(recv_transport);
for (size_t i = 0; i < num_video_streams; ++i) {
video_receive_configs_[i].rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
video_receive_configs_[i].rtp.rtx_ssrc = kSendRtxSsrcs[i];
video_receive_configs_[i].rtp.rtx_payload_types[payload_type] =
kSendRtxPayloadType;
video_receive_configs_[i].rtp.transport_cc = params_.call.send_side_bwe;
video_receive_configs_[i].rtp.remb = !params_.call.send_side_bwe;
// Enable RTT calculation so NTP time estimator will work.
video_receive_configs_[i].rtp.rtcp_xr.receiver_reference_time_report = true;
// Force fake decoders on non-selected simulcast streams.
if (i != params_.ss.selected_stream) {
VideoReceiveStream::Decoder decoder;
decoder.decoder = new test::FakeDecoder();
decoder.payload_type = video_send_config_.encoder_settings.payload_type;
decoder.payload_name = video_send_config_.encoder_settings.payload_name;
video_receive_configs_[i].decoders.clear();
allocated_decoders_.emplace_back(decoder.decoder);
video_receive_configs_[i].decoders.push_back(decoder);
}
}
if (params_.video.flexfec) {
// Override send config constructed by CreateSendConfig.
video_send_config_.rtp.flexfec.protected_media_ssrcs = {
kVideoSendSsrcs[params_.ss.selected_stream]};
// The matching receive config is _not_ created by
// CreateMatchingReceiveConfigs, since VideoQualityTest is not a BaseTest.
// Set up the receive config manually instead.
FlexfecReceiveStream::Config flexfec_receive_config(recv_transport);
flexfec_receive_config.payload_type =
video_send_config_.rtp.flexfec.payload_type;
flexfec_receive_config.remote_ssrc = video_send_config_.rtp.flexfec.ssrc;
flexfec_receive_config.protected_media_ssrcs =
video_send_config_.rtp.flexfec.protected_media_ssrcs;
flexfec_receive_config.local_ssrc = kReceiverLocalVideoSsrc;
flexfec_receive_config.transport_cc = params_.call.send_side_bwe;
if (params_.call.send_side_bwe) {
flexfec_receive_config.rtp_header_extensions.push_back(
RtpExtension(RtpExtension::kTransportSequenceNumberUri,
test::kTransportSequenceNumberExtensionId));
} else {
flexfec_receive_config.rtp_header_extensions.push_back(RtpExtension(
RtpExtension::kAbsSendTimeUri, test::kAbsSendTimeExtensionId));
}
flexfec_receive_configs_.push_back(flexfec_receive_config);
}
if (params_.video.ulpfec) {
video_send_config_.rtp.ulpfec.red_payload_type = kRedPayloadType;
video_send_config_.rtp.ulpfec.ulpfec_payload_type = kUlpfecPayloadType;
video_send_config_.rtp.ulpfec.red_rtx_payload_type = kRtxRedPayloadType;
video_receive_configs_[params_.ss.selected_stream]
.rtp.ulpfec.red_payload_type =
video_send_config_.rtp.ulpfec.red_payload_type;
video_receive_configs_[params_.ss.selected_stream]
.rtp.ulpfec.ulpfec_payload_type =
video_send_config_.rtp.ulpfec.ulpfec_payload_type;
video_receive_configs_[params_.ss.selected_stream]
.rtp.ulpfec.red_rtx_payload_type =
video_send_config_.rtp.ulpfec.red_rtx_payload_type;
}
}
void VideoQualityTest::SetupThumbnails(Transport* send_transport,
Transport* recv_transport) {
for (int i = 0; i < params_.num_thumbnails; ++i) {
thumbnail_encoders_.emplace_back(VP8Encoder::Create());
// Thumbnails will be send in the other way: from receiver_call to
// sender_call.
VideoSendStream::Config thumbnail_send_config(recv_transport);
thumbnail_send_config.rtp.ssrcs.push_back(kThumbnailSendSsrcStart + i);
thumbnail_send_config.encoder_settings.encoder =
thumbnail_encoders_.back().get();
thumbnail_send_config.encoder_settings.payload_name = params_.video.codec;
thumbnail_send_config.encoder_settings.payload_type = kPayloadTypeVP8;
thumbnail_send_config.rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
thumbnail_send_config.rtp.rtx.payload_type = kSendRtxPayloadType;
thumbnail_send_config.rtp.rtx.ssrcs.push_back(kThumbnailRtxSsrcStart + i);
thumbnail_send_config.rtp.extensions.clear();
if (params_.call.send_side_bwe) {
thumbnail_send_config.rtp.extensions.push_back(
RtpExtension(RtpExtension::kTransportSequenceNumberUri,
test::kTransportSequenceNumberExtensionId));
} else {
thumbnail_send_config.rtp.extensions.push_back(RtpExtension(
RtpExtension::kAbsSendTimeUri, test::kAbsSendTimeExtensionId));
}
VideoEncoderConfig thumbnail_encoder_config;
thumbnail_encoder_config.min_transmit_bitrate_bps = 7500;
thumbnail_send_config.suspend_below_min_bitrate =
params_.video.suspend_below_min_bitrate;
thumbnail_encoder_config.number_of_streams = 1;
thumbnail_encoder_config.max_bitrate_bps = 50000;
thumbnail_encoder_config.video_stream_factory =
new rtc::RefCountedObject<VideoStreamFactory>(
std::vector<webrtc::VideoStream>{DefaultThumbnailStream()});
thumbnail_encoder_config.spatial_layers = params_.ss.spatial_layers;
VideoReceiveStream::Config thumbnail_receive_config(send_transport);
thumbnail_receive_config.rtp.remb = false;
thumbnail_receive_config.rtp.transport_cc = true;
thumbnail_receive_config.rtp.local_ssrc = kReceiverLocalVideoSsrc;
for (const RtpExtension& extension : thumbnail_send_config.rtp.extensions)
thumbnail_receive_config.rtp.extensions.push_back(extension);
thumbnail_receive_config.renderer = &fake_renderer_;
VideoReceiveStream::Decoder decoder =
test::CreateMatchingDecoder(thumbnail_send_config.encoder_settings);
allocated_decoders_.push_back(
std::unique_ptr<VideoDecoder>(decoder.decoder));
thumbnail_receive_config.decoders.clear();
thumbnail_receive_config.decoders.push_back(decoder);
thumbnail_receive_config.rtp.remote_ssrc =
thumbnail_send_config.rtp.ssrcs[0];
thumbnail_receive_config.rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
thumbnail_receive_config.rtp.rtx_ssrc = kThumbnailRtxSsrcStart + i;
thumbnail_receive_config.rtp.rtx_payload_types[kPayloadTypeVP8] =
kSendRtxPayloadType;
thumbnail_receive_config.rtp.transport_cc = params_.call.send_side_bwe;
thumbnail_receive_config.rtp.remb = !params_.call.send_side_bwe;
thumbnail_encoder_configs_.push_back(thumbnail_encoder_config.Copy());
thumbnail_send_configs_.push_back(thumbnail_send_config.Copy());
thumbnail_receive_configs_.push_back(thumbnail_receive_config.Copy());
}
for (int i = 0; i < params_.num_thumbnails; ++i) {
thumbnail_send_streams_.push_back(receiver_call_->CreateVideoSendStream(
thumbnail_send_configs_[i].Copy(),
thumbnail_encoder_configs_[i].Copy()));
thumbnail_receive_streams_.push_back(sender_call_->CreateVideoReceiveStream(
thumbnail_receive_configs_[i].Copy()));
}
}
void VideoQualityTest::DestroyThumbnailStreams() {
for (VideoSendStream* thumbnail_send_stream : thumbnail_send_streams_)
receiver_call_->DestroyVideoSendStream(thumbnail_send_stream);
thumbnail_send_streams_.clear();
for (VideoReceiveStream* thumbnail_receive_stream :
thumbnail_receive_streams_)
sender_call_->DestroyVideoReceiveStream(thumbnail_receive_stream);
thumbnail_send_streams_.clear();
thumbnail_receive_streams_.clear();
}
void VideoQualityTest::SetupScreenshareOrSVC() {
if (params_.screenshare.enabled) {
// Fill out codec settings.
video_encoder_config_.content_type =
VideoEncoderConfig::ContentType::kScreen;
degradation_preference_ =
VideoSendStream::DegradationPreference::kMaintainResolution;
if (params_.video.codec == "VP8") {
VideoCodecVP8 vp8_settings = VideoEncoder::GetDefaultVp8Settings();
vp8_settings.denoisingOn = false;
vp8_settings.frameDroppingOn = false;
vp8_settings.numberOfTemporalLayers =
static_cast<unsigned char>(params_.video.num_temporal_layers);
video_encoder_config_.encoder_specific_settings =
new rtc::RefCountedObject<
VideoEncoderConfig::Vp8EncoderSpecificSettings>(vp8_settings);
} else if (params_.video.codec == "VP9") {
VideoCodecVP9 vp9_settings = VideoEncoder::GetDefaultVp9Settings();
vp9_settings.denoisingOn = false;
vp9_settings.frameDroppingOn = false;
vp9_settings.numberOfTemporalLayers =
static_cast<unsigned char>(params_.video.num_temporal_layers);
vp9_settings.numberOfSpatialLayers =
static_cast<unsigned char>(params_.ss.num_spatial_layers);
video_encoder_config_.encoder_specific_settings =
new rtc::RefCountedObject<
VideoEncoderConfig::Vp9EncoderSpecificSettings>(vp9_settings);
}
// Setup frame generator.
const size_t kWidth = 1850;
const size_t kHeight = 1110;
std::vector<std::string> slides = params_.screenshare.slides;
if (slides.size() == 0) {
slides.push_back(test::ResourcePath("web_screenshot_1850_1110", "yuv"));
slides.push_back(test::ResourcePath("presentation_1850_1110", "yuv"));
slides.push_back(test::ResourcePath("photo_1850_1110", "yuv"));
slides.push_back(test::ResourcePath("difficult_photo_1850_1110", "yuv"));
}
if (params_.screenshare.scroll_duration == 0) {
// Cycle image every slide_change_interval seconds.
frame_generator_ = test::FrameGenerator::CreateFromYuvFile(
slides, kWidth, kHeight,
params_.screenshare.slide_change_interval * params_.video.fps);
} else {
RTC_CHECK_LE(params_.video.width, kWidth);
RTC_CHECK_LE(params_.video.height, kHeight);
RTC_CHECK_GT(params_.screenshare.slide_change_interval, 0);
const int kPauseDurationMs = (params_.screenshare.slide_change_interval -
params_.screenshare.scroll_duration) *
1000;
RTC_CHECK_LE(params_.screenshare.scroll_duration,
params_.screenshare.slide_change_interval);
frame_generator_ = test::FrameGenerator::CreateScrollingInputFromYuvFiles(
clock_, slides, kWidth, kHeight, params_.video.width,
params_.video.height, params_.screenshare.scroll_duration * 1000,
kPauseDurationMs);
}
} else if (params_.ss.num_spatial_layers > 1) { // For non-screenshare case.
RTC_CHECK(params_.video.codec == "VP9");
VideoCodecVP9 vp9_settings = VideoEncoder::GetDefaultVp9Settings();
vp9_settings.numberOfTemporalLayers =
static_cast<unsigned char>(params_.video.num_temporal_layers);
vp9_settings.numberOfSpatialLayers =
static_cast<unsigned char>(params_.ss.num_spatial_layers);
video_encoder_config_.encoder_specific_settings = new rtc::RefCountedObject<
VideoEncoderConfig::Vp9EncoderSpecificSettings>(vp9_settings);
}
}
void VideoQualityTest::SetupThumbnailCapturers(size_t num_thumbnail_streams) {
VideoStream thumbnail = DefaultThumbnailStream();
for (size_t i = 0; i < num_thumbnail_streams; ++i) {
thumbnail_capturers_.emplace_back(test::FrameGeneratorCapturer::Create(
static_cast<int>(thumbnail.width), static_cast<int>(thumbnail.height),
thumbnail.max_framerate, clock_));
RTC_DCHECK(thumbnail_capturers_.back());
}
}
void VideoQualityTest::CreateCapturer() {
if (params_.screenshare.enabled) {
test::FrameGeneratorCapturer* frame_generator_capturer =
new test::FrameGeneratorCapturer(clock_, std::move(frame_generator_),
params_.video.fps);
EXPECT_TRUE(frame_generator_capturer->Init());
video_capturer_.reset(frame_generator_capturer);
} else {
if (params_.video.clip_name.empty()) {
video_capturer_.reset(test::VcmCapturer::Create(
params_.video.width, params_.video.height, params_.video.fps));
if (!video_capturer_) {
// Failed to get actual camera, use chroma generator as backup.
video_capturer_.reset(test::FrameGeneratorCapturer::Create(
static_cast<int>(params_.video.width),
static_cast<int>(params_.video.height), params_.video.fps, clock_));
}
} else {
video_capturer_.reset(test::FrameGeneratorCapturer::CreateFromYuvFile(
test::ResourcePath(params_.video.clip_name, "yuv"),
params_.video.width, params_.video.height, params_.video.fps,
clock_));
ASSERT_TRUE(video_capturer_) << "Could not create capturer for "
<< params_.video.clip_name
<< ".yuv. Is this resource file present?";
}
}
RTC_DCHECK(video_capturer_.get());
}
void VideoQualityTest::RunWithAnalyzer(const Params& params) {
params_ = params;
RTC_CHECK(!params_.audio.enabled);
// TODO(ivica): Merge with RunWithRenderer and use a flag / argument to
// differentiate between the analyzer and the renderer case.
CheckParams();
FILE* graph_data_output_file = nullptr;
if (!params_.analyzer.graph_data_output_filename.empty()) {
graph_data_output_file =
fopen(params_.analyzer.graph_data_output_filename.c_str(), "w");
RTC_CHECK(graph_data_output_file)
<< "Can't open the file " << params_.analyzer.graph_data_output_filename
<< "!";
}
Call::Config call_config(event_log_.get());
call_config.bitrate_config = params.call.call_bitrate_config;
CreateCalls(call_config, call_config);
test::LayerFilteringTransport send_transport(
params_.pipe, sender_call_.get(), kPayloadTypeVP8, kPayloadTypeVP9,
params_.video.selected_tl, params_.ss.selected_sl, payload_type_map_);
test::DirectTransport recv_transport(params_.pipe, receiver_call_.get(),
payload_type_map_);
std::string graph_title = params_.analyzer.graph_title;
if (graph_title.empty())
graph_title = VideoQualityTest::GenerateGraphTitle();
bool is_quick_test_enabled = field_trial::IsEnabled("WebRTC-QuickPerfTest");
VideoAnalyzer analyzer(
&send_transport, params_.analyzer.test_label,
params_.analyzer.avg_psnr_threshold, params_.analyzer.avg_ssim_threshold,
is_quick_test_enabled
? kFramesSentInQuickTest
: params_.analyzer.test_durations_secs * params_.video.fps,
graph_data_output_file, graph_title,
kVideoSendSsrcs[params_.ss.selected_stream],
kSendRtxSsrcs[params_.ss.selected_stream],
static_cast<size_t>(params_.ss.selected_stream), params.ss.selected_sl,
params_.video.selected_tl, is_quick_test_enabled);
analyzer.SetReceiver(receiver_call_->Receiver());
send_transport.SetReceiver(&analyzer);
recv_transport.SetReceiver(sender_call_->Receiver());
SetupVideo(&analyzer, &recv_transport);
SetupThumbnails(&analyzer, &recv_transport);
video_receive_configs_[params_.ss.selected_stream].renderer = &analyzer;
video_send_config_.pre_encode_callback = analyzer.pre_encode_proxy();
RTC_DCHECK(!video_send_config_.post_encode_callback);
video_send_config_.post_encode_callback = analyzer.encode_timing_proxy();
SetupScreenshareOrSVC();
CreateFlexfecStreams();
CreateVideoStreams();
analyzer.SetSendStream(video_send_stream_);
if (video_receive_streams_.size() == 1)
analyzer.SetReceiveStream(video_receive_streams_[0]);
video_send_stream_->SetSource(analyzer.OutputInterface(),
degradation_preference_);
SetupThumbnailCapturers(params_.num_thumbnails);
for (size_t i = 0; i < thumbnail_send_streams_.size(); ++i) {
thumbnail_send_streams_[i]->SetSource(thumbnail_capturers_[i].get(),
degradation_preference_);
}
CreateCapturer();
rtc::VideoSinkWants wants;
video_capturer_->AddOrUpdateSink(analyzer.InputInterface(), wants);
StartEncodedFrameLogs(video_send_stream_);
StartEncodedFrameLogs(video_receive_streams_[0]);
video_send_stream_->Start();
for (VideoSendStream* thumbnail_send_stream : thumbnail_send_streams_)
thumbnail_send_stream->Start();
for (VideoReceiveStream* receive_stream : video_receive_streams_)
receive_stream->Start();
for (FlexfecReceiveStream* receive_stream : flexfec_receive_streams_)
receive_stream->Start();
for (VideoReceiveStream* thumbnail_receive_stream :
thumbnail_receive_streams_)
thumbnail_receive_stream->Start();
analyzer.StartMeasuringCpuProcessTime();
video_capturer_->Start();
for (std::unique_ptr<test::VideoCapturer>& video_caputurer :
thumbnail_capturers_) {
video_caputurer->Start();
}
analyzer.Wait();
send_transport.StopSending();
recv_transport.StopSending();
for (std::unique_ptr<test::VideoCapturer>& video_caputurer :
thumbnail_capturers_)
video_caputurer->Stop();
video_capturer_->Stop();
for (VideoReceiveStream* thumbnail_receive_stream :
thumbnail_receive_streams_)
thumbnail_receive_stream->Stop();
for (FlexfecReceiveStream* receive_stream : flexfec_receive_streams_)
receive_stream->Stop();
for (VideoReceiveStream* receive_stream : video_receive_streams_)
receive_stream->Stop();
for (VideoSendStream* thumbnail_send_stream : thumbnail_send_streams_)
thumbnail_send_stream->Stop();
video_send_stream_->Stop();
DestroyStreams();
DestroyThumbnailStreams();
if (graph_data_output_file)
fclose(graph_data_output_file);
}
void VideoQualityTest::SetupAudio(int send_channel_id,
int receive_channel_id,
Call* call,
Transport* transport,
AudioReceiveStream** audio_receive_stream) {
audio_send_config_ = AudioSendStream::Config(transport);
audio_send_config_.voe_channel_id = send_channel_id;
audio_send_config_.rtp.ssrc = kAudioSendSsrc;
// Add extension to enable audio send side BWE, and allow audio bit rate
// adaptation.
audio_send_config_.rtp.extensions.clear();
if (params_.call.send_side_bwe) {
audio_send_config_.rtp.extensions.push_back(
webrtc::RtpExtension(webrtc::RtpExtension::kTransportSequenceNumberUri,
test::kTransportSequenceNumberExtensionId));
audio_send_config_.min_bitrate_bps = kOpusMinBitrateBps;
audio_send_config_.max_bitrate_bps = kOpusBitrateFbBps;
}
audio_send_config_.send_codec_spec =
rtc::Optional<AudioSendStream::Config::SendCodecSpec>(
{kAudioSendPayloadType,
{"OPUS", 48000, 2,
{{"usedtx", (params_.audio.dtx ? "1" : "0")},
{"stereo", "1"}}}});
audio_send_config_.encoder_factory = encoder_factory_;
audio_send_stream_ = call->CreateAudioSendStream(audio_send_config_);
AudioReceiveStream::Config audio_config;
audio_config.rtp.local_ssrc = kReceiverLocalAudioSsrc;
audio_config.rtcp_send_transport = transport;
audio_config.voe_channel_id = receive_channel_id;
audio_config.rtp.remote_ssrc = audio_send_config_.rtp.ssrc;
audio_config.rtp.transport_cc = params_.call.send_side_bwe;
audio_config.rtp.extensions = audio_send_config_.rtp.extensions;
audio_config.decoder_factory = decoder_factory_;
audio_config.decoder_map = {{kAudioSendPayloadType, {"OPUS", 48000, 2}}};
if (params_.video.enabled && params_.audio.sync_video)
audio_config.sync_group = kSyncGroup;
*audio_receive_stream = call->CreateAudioReceiveStream(audio_config);
}
void VideoQualityTest::RunWithRenderers(const Params& params) {
params_ = params;
CheckParams();
// TODO(ivica): Remove bitrate_config and use the default Call::Config(), to
// match the full stack tests.
Call::Config call_config(event_log_.get());
call_config.bitrate_config = params_.call.call_bitrate_config;
::VoiceEngineState voe;
if (params_.audio.enabled) {
CreateVoiceEngine(&voe, decoder_factory_);
AudioState::Config audio_state_config;
audio_state_config.voice_engine = voe.voice_engine;
audio_state_config.audio_mixer = AudioMixerImpl::Create();
call_config.audio_state = AudioState::Create(audio_state_config);
}
std::unique_ptr<Call> call(Call::Create(call_config));
// TODO(minyue): consider if this is a good transport even for audio only
// calls.
test::LayerFilteringTransport transport(
params.pipe, call.get(), kPayloadTypeVP8, kPayloadTypeVP9,
params.video.selected_tl, params_.ss.selected_sl, payload_type_map_);
// TODO(ivica): Use two calls to be able to merge with RunWithAnalyzer or at
// least share as much code as possible. That way this test would also match
// the full stack tests better.
transport.SetReceiver(call->Receiver());
VideoReceiveStream* video_receive_stream = nullptr;
FlexfecReceiveStream* flexfec_receive_stream = nullptr;
std::unique_ptr<test::VideoRenderer> local_preview;
std::unique_ptr<test::VideoRenderer> loopback_video;
if (params_.video.enabled) {
// Create video renderers.
local_preview.reset(test::VideoRenderer::Create(
"Local Preview", params_.video.width, params_.video.height));
size_t stream_id = params_.ss.selected_stream;
std::string title = "Loopback Video";
if (params_.ss.streams.size() > 1) {
std::ostringstream s;
s << stream_id;
title += " - Stream #" + s.str();
}
loopback_video.reset(test::VideoRenderer::Create(
title.c_str(), params_.ss.streams[stream_id].width,
params_.ss.streams[stream_id].height));
SetupVideo(&transport, &transport);
video_send_config_.pre_encode_callback = local_preview.get();
video_receive_configs_[stream_id].renderer = loopback_video.get();
if (params_.audio.enabled && params_.audio.sync_video)
video_receive_configs_[stream_id].sync_group = kSyncGroup;
if (params_.screenshare.enabled)
SetupScreenshareOrSVC();
video_send_stream_ = call->CreateVideoSendStream(
video_send_config_.Copy(), video_encoder_config_.Copy());
if (params_.video.flexfec) {
RTC_DCHECK(!flexfec_receive_configs_.empty());
flexfec_receive_stream =
call->CreateFlexfecReceiveStream(flexfec_receive_configs_[0]);
}
video_receive_stream = call->CreateVideoReceiveStream(
video_receive_configs_[stream_id].Copy());
CreateCapturer();
video_send_stream_->SetSource(video_capturer_.get(),
degradation_preference_);
}
AudioReceiveStream* audio_receive_stream = nullptr;
if (params_.audio.enabled) {
SetupAudio(voe.send_channel_id, voe.receive_channel_id, call.get(),
&transport, &audio_receive_stream);
}
StartEncodedFrameLogs(video_receive_stream);
StartEncodedFrameLogs(video_send_stream_);
// Start sending and receiving video.
if (params_.video.enabled) {
if (flexfec_receive_stream)
flexfec_receive_stream->Start();
video_receive_stream->Start();
video_send_stream_->Start();
video_capturer_->Start();
}
if (params_.audio.enabled) {
// Start receiving audio.
audio_receive_stream->Start();
EXPECT_EQ(0, voe.base->StartPlayout(voe.receive_channel_id));
// Start sending audio.
audio_send_stream_->Start();
EXPECT_EQ(0, voe.base->StartSend(voe.send_channel_id));
}
test::PressEnterToContinue();
if (params_.audio.enabled) {
// Stop sending audio.
EXPECT_EQ(0, voe.base->StopSend(voe.send_channel_id));
audio_send_stream_->Stop();
// Stop receiving audio.
EXPECT_EQ(0, voe.base->StopPlayout(voe.receive_channel_id));
audio_receive_stream->Stop();
call->DestroyAudioSendStream(audio_send_stream_);
call->DestroyAudioReceiveStream(audio_receive_stream);
}
// Stop receiving and sending video.
if (params_.video.enabled) {
video_capturer_->Stop();
video_send_stream_->Stop();
video_receive_stream->Stop();
if (flexfec_receive_stream) {
flexfec_receive_stream->Stop();
call->DestroyFlexfecReceiveStream(flexfec_receive_stream);
}
call->DestroyVideoReceiveStream(video_receive_stream);
call->DestroyVideoSendStream(video_send_stream_);
}
transport.StopSending();
if (params_.audio.enabled)
DestroyVoiceEngine(&voe);
}
void VideoQualityTest::StartEncodedFrameLogs(VideoSendStream* stream) {
if (!params_.video.encoded_frame_base_path.empty()) {
std::ostringstream str;
str << send_logs_++;
std::string prefix =
params_.video.encoded_frame_base_path + "." + str.str() + ".send.";
stream->EnableEncodedFrameRecording(
std::vector<rtc::PlatformFile>(
{rtc::CreatePlatformFile(prefix + "1.ivf"),
rtc::CreatePlatformFile(prefix + "2.ivf"),
rtc::CreatePlatformFile(prefix + "3.ivf")}),
10000000);
}
}
void VideoQualityTest::StartEncodedFrameLogs(VideoReceiveStream* stream) {
if (!params_.video.encoded_frame_base_path.empty()) {
std::ostringstream str;
str << receive_logs_++;
std::string path =
params_.video.encoded_frame_base_path + "." + str.str() + ".recv.ivf";
stream->EnableEncodedFrameRecording(rtc::CreatePlatformFile(path),
10000000);
}
}
} // namespace webrtc