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webrtc / src / 66ac50e58c790624d51ede10ae438cbadbca9d2e / . / webrtc / video / video_quality_test.cc
blob: e6feb2a0e88209a13534acacecde26f00f5db6ad [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 <sstream>
#include <string>
#include <vector>
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/base/checks.h"
#include "webrtc/base/event.h"
#include "webrtc/base/format_macros.h"
#include "webrtc/base/optional.h"
#include "webrtc/base/timeutils.h"
#include "webrtc/call.h"
#include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
#include "webrtc/system_wrappers/include/cpu_info.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"
#include "webrtc/voice_engine/include/voe_codec.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 kOpusMinBitrate = 6000;
constexpr int kOpusBitrateFb = 32000;
struct VoiceEngineState {
VoiceEngineState()
: voice_engine(nullptr),
base(nullptr),
codec(nullptr),
send_channel_id(-1),
receive_channel_id(-1) {}
webrtc::VoiceEngine* voice_engine;
webrtc::VoEBase* base;
webrtc::VoECodec* codec;
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);
voe->codec = webrtc::VoECodec::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;
voe->codec->Release();
voe->codec = nullptr;
webrtc::VoiceEngine::Delete(voe->voice_engine);
voe->voice_engine = nullptr;
}
} // namespace
namespace webrtc {
class VideoAnalyzer : public PacketReceiver,
public Transport,
public rtc::VideoSinkInterface<VideoFrame>,
public EncodedFrameObserver {
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)
: transport_(transport),
receiver_(nullptr),
send_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),
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),
avg_psnr_threshold_(avg_psnr_threshold),
avg_ssim_threshold_(avg_ssim_threshold),
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, 1u);
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;
}
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);
{
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_send_timestamp_ && rtp_timestamp_delta_ == 0) {
while (frames_.front().timestamp() != video_frame.timestamp()) {
++dropped_frames_before_first_encode_;
frames_.pop_front();
RTC_CHECK(!frames_.empty());
}
first_send_timestamp_ = rtc::Optional<uint32_t>(video_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) {
rtp_timestamp_delta_ = header.timestamp - *first_send_timestamp_;
first_send_timestamp_ = rtc::Optional<uint32_t>();
}
int64_t timestamp =
wrap_handler_.Unwrap(header.timestamp - rtp_timestamp_delta_);
send_times_[timestamp] = current_time;
if (!transport_->DiscardedLastPacket() &&
header.ssrc == ssrc_to_analyze_) {
encoded_frame_sizes_[timestamp] +=
length - (header.headerLength + header.paddingLength);
}
}
return result;
}
bool SendRtcp(const uint8_t* packet, size_t length) override {
return transport_->SendRtcp(packet, length);
}
void EncodedFrameCallback(const EncodedFrame& frame) override {
rtc::CritScope lock(&comparison_lock_);
if (frames_recorded_ < frames_to_process_)
encoded_frame_size_.AddSample(frame.length_);
}
void OnFrame(const VideoFrame& video_frame) override {
int64_t render_time_ms =
Clock::GetRealTimeClock()->CurrentNtpInMilliseconds();
rtc::CritScope lock(&crit_);
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_;
frames_.pop_front();
RTC_CHECK(!frames_.empty());
continue;
}
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);
}
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_; }
test::LayerFilteringTransport* const transport_;
PacketReceiver* receiver_;
private:
struct FrameComparison {
FrameComparison()
: dropped(false),
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 send_time_ms,
int64_t recv_time_ms,
int64_t render_time_ms,
size_t encoded_frame_size)
: reference(reference),
render(render),
dropped(dropped),
send_time_ms(send_time_ms),
recv_time_ms(recv_time_ms),
render_time_ms(render_time_ms),
encoded_frame_size(encoded_frame_size) {}
VideoFrame reference;
VideoFrame render;
bool dropped;
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 {}
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_;
};
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);
VideoFrame reference_copy;
VideoFrame render_copy;
rtc::CritScope crit(&comparison_lock_);
if (comparisons_.size() < kMaxComparisons) {
reference_copy.CopyFrame(reference);
render_copy.CopyFrame(render);
} else {
// Copy the time to ensure that delay calculations can still be made.
reference_copy.set_ntp_time_ms(reference.ntp_time_ms());
render_copy.set_ntp_time_ms(render.ntp_time_ms());
}
comparisons_.push_back(FrameComparison(reference_copy, render_copy, dropped,
send_time_ms, recv_time_ms,
render_time_ms, encoded_size));
comparison_available_event_.Set();
}
static bool PollStatsThread(void* obj) {
return static_cast<VideoAnalyzer*>(obj)->PollStats();
}
bool PollStats() {
if (done_.Wait(kSendStatsPollingIntervalMs))
return false;
VideoSendStream::Stats stats = send_stream_->GetStats();
rtc::CritScope crit(&comparison_lock_);
// It's not certain that we yet have estimates for any of these stats. Check
// that they are positive before mixing them in.
if (stats.encode_frame_rate > 0)
encode_frame_rate_.AddSample(stats.encode_frame_rate);
if (stats.avg_encode_time_ms > 0)
encode_time_ms.AddSample(stats.avg_encode_time_ms);
if (stats.encode_usage_percent > 0)
encode_usage_percent.AddSample(stats.encode_usage_percent);
if (stats.media_bitrate_bps > 0)
media_bitrate_bps.AddSample(stats.media_bitrate_bps);
return true;
}
static bool FrameComparisonThread(void* obj) {
return static_cast<VideoAnalyzer*>(obj)->CompareFrames();
}
bool CompareFrames() {
if (AllFramesRecorded())
return false;
VideoFrame reference;
VideoFrame render;
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.
}
PerformFrameComparison(comparison);
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() {
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("encoded_frame_size", encoded_frame_size_, " bytes");
PrintResult("encode_frame_rate", encode_frame_rate_, " fps");
PrintResult("encode_time", encode_time_ms, " ms");
PrintResult("encode_usage_percent", encode_usage_percent, " percent");
PrintResult("media_bitrate", media_bitrate_bps, " bps");
printf("RESULT dropped_frames: %s = %d frames\n", test_label_.c_str(),
dropped_frames_);
printf("RESULT dropped_frames_before_first_encode: %s = %d frames\n",
test_label_.c_str(), dropped_frames_before_first_encode_);
printf("RESULT dropped_frames_before_rendering: %s = %d frames\n",
test_label_.c_str(), dropped_frames_before_rendering_);
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.IsZeroSize()) {
psnr = I420PSNR(&comparison.reference, &comparison.render);
ssim = I420SSIM(&comparison.reference, &comparison.render);
}
int64_t input_time_ms = comparison.reference.ntp_time_ms();
rtc::CritScope crit(&comparison_lock_);
if (graph_data_output_file_) {
samples_.push_back(
Sample(comparison.dropped, 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 - input_time_ms);
receiver_time_.AddSample(comparison.render_time_ms -
comparison.recv_time_ms);
end_to_end_.AddSample(comparison.render_time_ms - 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);
}
}
// 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;
copy.set_timestamp(copy.ntp_time_ms() * 90);
analyzer_->AddCapturedFrameForComparison(video_frame);
rtc::CritScope lock(&crit_);
if (send_stream_input_)
send_stream_input_->OnFrame(video_frame);
}
// 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_);
RTC_DCHECK_EQ(0u, video_frame.timestamp());
// Frames from the capturer does not have a rtp timestamp. Create one so it
// can be used for comparison.
VideoFrame copy = video_frame;
copy.set_timestamp(copy.ntp_time_ms() * 90);
frames_.push_back(copy);
}
VideoSendStream* send_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_;
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 media_bitrate_bps 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_;
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_send_timestamp_ GUARDED_BY(crit_);
const double avg_psnr_threshold_;
const double avg_ssim_threshold_;
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_;
};
VideoQualityTest::VideoQualityTest() : clock_(Clock::GetRealTimeClock()) {}
void VideoQualityTest::TestBody() {}
std::string VideoQualityTest::GenerateGraphTitle() const {
std::stringstream ss;
ss << params_.common.codec;
ss << " (" << params_.common.target_bitrate_bps / 1000 << "kbps";
ss << ", " << params_.common.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() {
// 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_.common.selected_tl == -1 ||
params_.common.selected_tl ==
params_.common.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_.common.max_bitrate_bps,
params_.common.target_bitrate_bps);
RTC_CHECK_GE(params_.common.target_bitrate_bps,
params_.common.min_bitrate_bps);
RTC_CHECK_LT(params_.common.selected_tl, params_.common.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_EQ(static_cast<int>(stream.temporal_layer_thresholds_bps.size()),
params_.common.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_.common.codec == "VP8") {
RTC_CHECK_EQ(params_.ss.num_spatial_layers, 1);
} else if (params_.common.codec == "VP9") {
RTC_CHECK_EQ(params_.ss.streams.size(), 1u);
}
}
// 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.common.width;
stream.height = params.common.height;
stream.max_framerate = params.common.fps;
stream.min_bitrate_bps = params.common.min_bitrate_bps;
stream.target_bitrate_bps = params.common.target_bitrate_bps;
stream.max_bitrate_bps = params.common.max_bitrate_bps;
stream.max_qp = 52;
if (params.common.num_temporal_layers == 2)
stream.temporal_layer_thresholds_bps.push_back(stream.target_bitrate_bps);
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->common.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::SetupCommon(Transport* send_transport,
Transport* recv_transport) {
if (params_.logs)
trace_to_stderr_.reset(new test::TraceToStderr);
size_t num_streams = params_.ss.streams.size();
CreateSendConfig(num_streams, 0, send_transport);
int payload_type;
if (params_.common.codec == "H264") {
encoder_.reset(VideoEncoder::Create(VideoEncoder::kH264));
payload_type = kPayloadTypeH264;
} else if (params_.common.codec == "VP8") {
encoder_.reset(VideoEncoder::Create(VideoEncoder::kVp8));
payload_type = kPayloadTypeVP8;
} else if (params_.common.codec == "VP9") {
encoder_.reset(VideoEncoder::Create(VideoEncoder::kVp9));
payload_type = kPayloadTypeVP9;
} else {
RTC_NOTREACHED() << "Codec not supported!";
return;
}
video_send_config_.encoder_settings.encoder = encoder_.get();
video_send_config_.encoder_settings.payload_name = params_.common.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_streams; ++i)
video_send_config_.rtp.rtx.ssrcs.push_back(kSendRtxSsrcs[i]);
video_send_config_.rtp.extensions.clear();
if (params_.common.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_encoder_config_.min_transmit_bitrate_bps =
params_.common.min_transmit_bps;
video_encoder_config_.streams = params_.ss.streams;
video_encoder_config_.spatial_layers = params_.ss.spatial_layers;
CreateMatchingReceiveConfigs(recv_transport);
for (size_t i = 0; i < num_streams; ++i) {
video_receive_configs_[i].rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
video_receive_configs_[i].rtp.rtx[payload_type].ssrc = kSendRtxSsrcs[i];
video_receive_configs_[i].rtp.rtx[payload_type].payload_type =
kSendRtxPayloadType;
video_receive_configs_[i].rtp.transport_cc = params_.common.send_side_bwe;
}
}
void VideoQualityTest::SetupScreenshare() {
RTC_CHECK(params_.screenshare.enabled);
// Fill out codec settings.
video_encoder_config_.content_type = VideoEncoderConfig::ContentType::kScreen;
if (params_.common.codec == "VP8") {
codec_settings_.VP8 = VideoEncoder::GetDefaultVp8Settings();
codec_settings_.VP8.denoisingOn = false;
codec_settings_.VP8.frameDroppingOn = false;
codec_settings_.VP8.numberOfTemporalLayers =
static_cast<unsigned char>(params_.common.num_temporal_layers);
video_encoder_config_.encoder_specific_settings = &codec_settings_.VP8;
} else if (params_.common.codec == "VP9") {
codec_settings_.VP9 = VideoEncoder::GetDefaultVp9Settings();
codec_settings_.VP9.denoisingOn = false;
codec_settings_.VP9.frameDroppingOn = false;
codec_settings_.VP9.numberOfTemporalLayers =
static_cast<unsigned char>(params_.common.num_temporal_layers);
video_encoder_config_.encoder_specific_settings = &codec_settings_.VP9;
codec_settings_.VP9.numberOfSpatialLayers =
static_cast<unsigned char>(params_.ss.num_spatial_layers);
}
// Setup frame generator.
const size_t kWidth = 1850;
const size_t kHeight = 1110;
std::vector<std::string> slides;
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_.reset(test::FrameGenerator::CreateFromYuvFile(
slides, kWidth, kHeight,
params_.screenshare.slide_change_interval * params_.common.fps));
} else {
RTC_CHECK_LE(params_.common.width, kWidth);
RTC_CHECK_LE(params_.common.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_.reset(
test::FrameGenerator::CreateScrollingInputFromYuvFiles(
clock_, slides, kWidth, kHeight, params_.common.width,
params_.common.height, params_.screenshare.scroll_duration * 1000,
kPauseDurationMs));
}
}
void VideoQualityTest::CreateCapturer() {
if (params_.screenshare.enabled) {
test::FrameGeneratorCapturer* frame_generator_capturer =
new test::FrameGeneratorCapturer(clock_, frame_generator_.release(),
params_.common.fps);
EXPECT_TRUE(frame_generator_capturer->Init());
capturer_.reset(frame_generator_capturer);
} else {
if (params_.video.clip_name.empty()) {
capturer_.reset(test::VcmCapturer::Create(
params_.common.width, params_.common.height, params_.common.fps));
} else {
capturer_.reset(test::FrameGeneratorCapturer::CreateFromYuvFile(
test::ResourcePath(params_.video.clip_name, "yuv"),
params_.common.width, params_.common.height, params_.common.fps,
clock_));
ASSERT_TRUE(capturer_) << "Could not create capturer for "
<< params_.video.clip_name
<< ".yuv. Is this resource file present?";
}
}
}
void VideoQualityTest::RunWithAnalyzer(const Params& params) {
params_ = params;
RTC_CHECK(!params_.audio);
// 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;
call_config.bitrate_config = params.common.call_bitrate_config;
CreateCalls(call_config, call_config);
test::LayerFilteringTransport send_transport(
params.pipe, sender_call_.get(), kPayloadTypeVP8, kPayloadTypeVP9,
params.common.selected_tl, params_.ss.selected_sl);
test::DirectTransport recv_transport(params.pipe, receiver_call_.get());
std::string graph_title = params_.analyzer.graph_title;
if (graph_title.empty())
graph_title = VideoQualityTest::GenerateGraphTitle();
// In the case of different resolutions, the functions calculating PSNR and
// SSIM return -1.0, instead of a positive value as usual. VideoAnalyzer
// aborts if the average psnr/ssim are below the given threshold, which is
// 0.0 by default. Setting the thresholds to -1.1 prevents the unnecessary
// abort.
VideoStream& selected_stream = params_.ss.streams[params_.ss.selected_stream];
int selected_sl = params_.ss.selected_sl != -1
? params_.ss.selected_sl
: params_.ss.num_spatial_layers - 1;
bool disable_quality_check =
selected_stream.width != params_.common.width ||
selected_stream.height != params_.common.height ||
(!params_.ss.spatial_layers.empty() &&
params_.ss.spatial_layers[selected_sl].scaling_factor_num !=
params_.ss.spatial_layers[selected_sl].scaling_factor_den);
if (disable_quality_check) {
fprintf(stderr,
"Warning: Calculating PSNR and SSIM for downsized resolution "
"not implemented yet! Skipping PSNR and SSIM calculations!");
}
VideoAnalyzer analyzer(
&send_transport, params_.analyzer.test_label,
disable_quality_check ? -1.1 : params_.analyzer.avg_psnr_threshold,
disable_quality_check ? -1.1 : params_.analyzer.avg_ssim_threshold,
params_.analyzer.test_durations_secs * params_.common.fps,
graph_data_output_file, graph_title,
kVideoSendSsrcs[params_.ss.selected_stream]);
analyzer.SetReceiver(receiver_call_->Receiver());
send_transport.SetReceiver(&analyzer);
recv_transport.SetReceiver(sender_call_->Receiver());
SetupCommon(&analyzer, &recv_transport);
video_receive_configs_[params_.ss.selected_stream].renderer = &analyzer;
video_send_config_.pre_encode_callback = analyzer.pre_encode_proxy();
for (auto& config : video_receive_configs_)
config.pre_decode_callback = &analyzer;
RTC_DCHECK(!video_send_config_.post_encode_callback);
video_send_config_.post_encode_callback = analyzer.encode_timing_proxy();
if (params_.screenshare.enabled)
SetupScreenshare();
CreateVideoStreams();
analyzer.SetSendStream(video_send_stream_);
video_send_stream_->SetSource(analyzer.OutputInterface());
CreateCapturer();
rtc::VideoSinkWants wants;
capturer_->AddOrUpdateSink(analyzer.InputInterface(), wants);
video_send_stream_->Start();
for (VideoReceiveStream* receive_stream : video_receive_streams_)
receive_stream->Start();
capturer_->Start();
analyzer.Wait();
send_transport.StopSending();
recv_transport.StopSending();
capturer_->Stop();
for (VideoReceiveStream* receive_stream : video_receive_streams_)
receive_stream->Stop();
video_send_stream_->Stop();
DestroyStreams();
if (graph_data_output_file)
fclose(graph_data_output_file);
}
void VideoQualityTest::RunWithRenderers(const Params& params) {
params_ = params;
CheckParams();
std::unique_ptr<test::VideoRenderer> local_preview(
test::VideoRenderer::Create("Local Preview", params_.common.width,
params_.common.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();
}
std::unique_ptr<test::VideoRenderer> loopback_video(
test::VideoRenderer::Create(title.c_str(),
params_.ss.streams[stream_id].width,
params_.ss.streams[stream_id].height));
// TODO(ivica): Remove bitrate_config and use the default Call::Config(), to
// match the full stack tests.
Call::Config call_config;
call_config.bitrate_config = params_.common.call_bitrate_config;
::VoiceEngineState voe;
if (params_.audio) {
CreateVoiceEngine(&voe, decoder_factory_);
AudioState::Config audio_state_config;
audio_state_config.voice_engine = voe.voice_engine;
call_config.audio_state = AudioState::Create(audio_state_config);
}
std::unique_ptr<Call> call(Call::Create(call_config));
test::LayerFilteringTransport transport(
params.pipe, call.get(), kPayloadTypeVP8, kPayloadTypeVP9,
params.common.selected_tl, params_.ss.selected_sl);
// 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());
SetupCommon(&transport, &transport);
video_send_config_.pre_encode_callback = local_preview.get();
video_receive_configs_[stream_id].renderer = loopback_video.get();
if (params_.audio && params_.audio_video_sync)
video_receive_configs_[stream_id].sync_group = kSyncGroup;
video_send_config_.suspend_below_min_bitrate =
params_.common.suspend_below_min_bitrate;
if (params.common.fec) {
video_send_config_.rtp.fec.red_payload_type = kRedPayloadType;
video_send_config_.rtp.fec.ulpfec_payload_type = kUlpfecPayloadType;
video_receive_configs_[stream_id].rtp.fec.red_payload_type =
kRedPayloadType;
video_receive_configs_[stream_id].rtp.fec.ulpfec_payload_type =
kUlpfecPayloadType;
}
if (params_.screenshare.enabled)
SetupScreenshare();
video_send_stream_ = call->CreateVideoSendStream(
video_send_config_.Copy(), video_encoder_config_.Copy());
VideoReceiveStream* video_receive_stream =
call->CreateVideoReceiveStream(video_receive_configs_[stream_id].Copy());
CreateCapturer();
video_send_stream_->SetSource(capturer_.get());
AudioReceiveStream* audio_receive_stream = nullptr;
if (params_.audio) {
audio_send_config_ = AudioSendStream::Config(&transport);
audio_send_config_.voe_channel_id = voe.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_.common.send_side_bwe) {
audio_send_config_.rtp.extensions.push_back(webrtc::RtpExtension(
webrtc::RtpExtension::kTransportSequenceNumberUri,
test::kTransportSequenceNumberExtensionId));
audio_send_config_.min_bitrate_kbps = kOpusMinBitrate / 1000;
audio_send_config_.max_bitrate_kbps = kOpusBitrateFb / 1000;
}
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 = voe.receive_channel_id;
audio_config.rtp.remote_ssrc = audio_send_config_.rtp.ssrc;
audio_config.rtp.transport_cc = params_.common.send_side_bwe;
audio_config.rtp.extensions = audio_send_config_.rtp.extensions;
audio_config.decoder_factory = decoder_factory_;
if (params_.audio_video_sync)
audio_config.sync_group = kSyncGroup;
audio_receive_stream =call->CreateAudioReceiveStream(audio_config);
const CodecInst kOpusInst = {120, "OPUS", 48000, 960, 2, 64000};
EXPECT_EQ(0, voe.codec->SetSendCodec(voe.send_channel_id, kOpusInst));
}
// Start sending and receiving video.
video_receive_stream->Start();
video_send_stream_->Start();
capturer_->Start();
if (params_.audio) {
// Start receiving audio.
audio_receive_stream->Start();
EXPECT_EQ(0, voe.base->StartPlayout(voe.receive_channel_id));
EXPECT_EQ(0, voe.base->StartReceive(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) {
// 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->StopReceive(voe.receive_channel_id));
EXPECT_EQ(0, voe.base->StopPlayout(voe.receive_channel_id));
audio_receive_stream->Stop();
}
// Stop receiving and sending video.
capturer_->Stop();
video_send_stream_->Stop();
video_receive_stream->Stop();
call->DestroyVideoReceiveStream(video_receive_stream);
call->DestroyVideoSendStream(video_send_stream_);
if (params_.audio) {
call->DestroyAudioSendStream(audio_send_stream_);
call->DestroyAudioReceiveStream(audio_receive_stream);
}
transport.StopSending();
if (params_.audio)
DestroyVoiceEngine(&voe);
}
} // namespace webrtc