blob: 9c4d2049252877439343dd886ab97632ae44b267 [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 <queue>
#include "webrtc/base/format_macros.h"
#include "webrtc/base/timeutils.h"
#include "webrtc/system_wrappers/include/sleep.h"
#include "webrtc/test/gtest.h"
#include "webrtc/test/testsupport/fileutils.h"
#include "webrtc/voice_engine/test/auto_test/fakes/conference_transport.h"
namespace {
const int kRttMs = 25;
bool IsNear(int ref, int comp, int error) {
return (ref - comp <= error) && (comp - ref >= -error);
}
void CreateSilenceFile(const std::string& silence_file, int sample_rate_hz) {
FILE* fid = fopen(silence_file.c_str(), "wb");
int16_t zero = 0;
for (int i = 0; i < sample_rate_hz; ++i) {
// Write 1 second, but it does not matter since the file will be looped.
fwrite(&zero, sizeof(int16_t), 1, fid);
}
fclose(fid);
}
} // namespace
namespace voetest {
TEST(VoeConferenceTest, RttAndStartNtpTime) {
struct Stats {
Stats(int64_t rtt_receiver_1, int64_t rtt_receiver_2, int64_t ntp_delay)
: rtt_receiver_1_(rtt_receiver_1),
rtt_receiver_2_(rtt_receiver_2),
ntp_delay_(ntp_delay) {
}
int64_t rtt_receiver_1_;
int64_t rtt_receiver_2_;
int64_t ntp_delay_;
};
const std::string input_file =
webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm");
const webrtc::FileFormats kInputFormat = webrtc::kFileFormatPcm32kHzFile;
const int kDelayMs = 987;
ConferenceTransport trans;
trans.SetRtt(kRttMs);
unsigned int id_1 = trans.AddStream(input_file, kInputFormat);
unsigned int id_2 = trans.AddStream(input_file, kInputFormat);
EXPECT_TRUE(trans.StartPlayout(id_1));
// Start NTP time is the time when a stream is played out, rather than
// when it is added.
webrtc::SleepMs(kDelayMs);
EXPECT_TRUE(trans.StartPlayout(id_2));
const int kMaxRunTimeMs = 25000;
const int kNeedSuccessivePass = 3;
const int kStatsRequestIntervalMs = 1000;
const int kStatsBufferSize = 3;
int64_t deadline = rtc::TimeAfter(kMaxRunTimeMs);
// Run the following up to |kMaxRunTimeMs| milliseconds.
int successive_pass = 0;
webrtc::CallStatistics stats_1;
webrtc::CallStatistics stats_2;
std::queue<Stats> stats_buffer;
while (rtc::TimeMillis() < deadline &&
successive_pass < kNeedSuccessivePass) {
webrtc::SleepMs(kStatsRequestIntervalMs);
EXPECT_TRUE(trans.GetReceiverStatistics(id_1, &stats_1));
EXPECT_TRUE(trans.GetReceiverStatistics(id_2, &stats_2));
// It is not easy to verify the NTP time directly. We verify it by testing
// the difference of two start NTP times.
int64_t captured_start_ntp_delay = stats_2.capture_start_ntp_time_ms_ -
stats_1.capture_start_ntp_time_ms_;
// For the checks of RTT and start NTP time, We allow 10% accuracy.
if (IsNear(kRttMs, stats_1.rttMs, kRttMs / 10 + 1) &&
IsNear(kRttMs, stats_2.rttMs, kRttMs / 10 + 1) &&
IsNear(kDelayMs, captured_start_ntp_delay, kDelayMs / 10 + 1)) {
successive_pass++;
} else {
successive_pass = 0;
}
if (stats_buffer.size() >= kStatsBufferSize) {
stats_buffer.pop();
}
stats_buffer.push(Stats(stats_1.rttMs, stats_2.rttMs,
captured_start_ntp_delay));
}
EXPECT_GE(successive_pass, kNeedSuccessivePass) << "Expected to get RTT and"
" start NTP time estimate within 10% of the correct value over "
<< kStatsRequestIntervalMs * kNeedSuccessivePass / 1000
<< " seconds.";
if (successive_pass < kNeedSuccessivePass) {
printf("The most recent values (RTT for receiver 1, RTT for receiver 2, "
"NTP delay between receiver 1 and 2) are (from oldest):\n");
while (!stats_buffer.empty()) {
Stats stats = stats_buffer.front();
printf("(%" PRId64 ", %" PRId64 ", %" PRId64 ")\n", stats.rtt_receiver_1_,
stats.rtt_receiver_2_, stats.ntp_delay_);
stats_buffer.pop();
}
}
}
TEST(VoeConferenceTest, ReceivedPackets) {
const int kPackets = 50;
const int kPacketDurationMs = 20; // Correspond to Opus.
const std::string input_file =
webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm");
const webrtc::FileFormats kInputFormat = webrtc::kFileFormatPcm32kHzFile;
const std::string silence_file =
webrtc::test::TempFilename(webrtc::test::OutputPath(), "silence");
CreateSilenceFile(silence_file, 32000);
{
ConferenceTransport trans;
// Add silence to stream 0, so that it will be filtered out.
unsigned int id_0 = trans.AddStream(silence_file, kInputFormat);
unsigned int id_1 = trans.AddStream(input_file, kInputFormat);
unsigned int id_2 = trans.AddStream(input_file, kInputFormat);
unsigned int id_3 = trans.AddStream(input_file, kInputFormat);
EXPECT_TRUE(trans.StartPlayout(id_0));
EXPECT_TRUE(trans.StartPlayout(id_1));
EXPECT_TRUE(trans.StartPlayout(id_2));
EXPECT_TRUE(trans.StartPlayout(id_3));
webrtc::SleepMs(kPacketDurationMs * kPackets);
webrtc::CallStatistics stats_0;
webrtc::CallStatistics stats_1;
webrtc::CallStatistics stats_2;
webrtc::CallStatistics stats_3;
EXPECT_TRUE(trans.GetReceiverStatistics(id_0, &stats_0));
EXPECT_TRUE(trans.GetReceiverStatistics(id_1, &stats_1));
EXPECT_TRUE(trans.GetReceiverStatistics(id_2, &stats_2));
EXPECT_TRUE(trans.GetReceiverStatistics(id_3, &stats_3));
// We expect stream 0 to be filtered out totally, but since it may join the
// call earlier than other streams and the beginning packets might have got
// through. So we only expect |packetsReceived| to be close to zero.
EXPECT_NEAR(stats_0.packetsReceived, 0, 2);
// We expect |packetsReceived| to match |kPackets|, but the actual value
// depends on the sleep timer. So we allow a small off from |kPackets|.
EXPECT_NEAR(stats_1.packetsReceived, kPackets, 2);
EXPECT_NEAR(stats_2.packetsReceived, kPackets, 2);
EXPECT_NEAR(stats_3.packetsReceived, kPackets, 2);
}
remove(silence_file.c_str());
}
} // namespace voetest