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/*
* 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 "testing/gtest/include/gtest/gtest.h"
#include "webrtc/audio/audio_send_stream.h"
#include "webrtc/audio/conversion.h"
#include "webrtc/test/fake_voice_engine.h"
namespace webrtc {
namespace test {
TEST(AudioSendStreamTest, ConfigToString) {
const int kAbsSendTimeId = 3;
AudioSendStream::Config config(nullptr);
config.rtp.ssrc = 1234;
config.rtp.extensions.push_back(
RtpExtension(RtpExtension::kAbsSendTime, kAbsSendTimeId));
config.voe_channel_id = 1;
config.cng_payload_type = 42;
config.red_payload_type = 17;
EXPECT_EQ("{rtp: {ssrc: 1234, extensions: [{name: "
"http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time, id: 3}]}, "
"voe_channel_id: 1, cng_payload_type: 42, red_payload_type: 17}",
config.ToString());
}
TEST(AudioSendStreamTest, ConstructDestruct) {
FakeVoiceEngine voice_engine;
AudioSendStream::Config config(nullptr);
config.voe_channel_id = 1;
internal::AudioSendStream send_stream(config, &voice_engine);
}
TEST(AudioSendStreamTest, GetStats) {
FakeVoiceEngine voice_engine;
AudioSendStream::Config config(nullptr);
config.rtp.ssrc = FakeVoiceEngine::kSendSsrc;
config.voe_channel_id = FakeVoiceEngine::kSendChannelId;
internal::AudioSendStream send_stream(config, &voice_engine);
AudioSendStream::Stats stats = send_stream.GetStats();
const CallStatistics& call_stats = FakeVoiceEngine::kSendCallStats;
const CodecInst& codec_inst = FakeVoiceEngine::kSendCodecInst;
const ReportBlock& report_block = FakeVoiceEngine::kSendReportBlock;
EXPECT_EQ(FakeVoiceEngine::kSendSsrc, stats.local_ssrc);
EXPECT_EQ(static_cast<int64_t>(call_stats.bytesSent), stats.bytes_sent);
EXPECT_EQ(call_stats.packetsSent, stats.packets_sent);
EXPECT_EQ(static_cast<int32_t>(report_block.cumulative_num_packets_lost),
stats.packets_lost);
EXPECT_EQ(Q8ToFloat(report_block.fraction_lost), stats.fraction_lost);
EXPECT_EQ(std::string(codec_inst.plname), stats.codec_name);
EXPECT_EQ(static_cast<int32_t>(report_block.extended_highest_sequence_number),
stats.ext_seqnum);
EXPECT_EQ(static_cast<int32_t>(report_block.interarrival_jitter /
(codec_inst.plfreq / 1000)), stats.jitter_ms);
EXPECT_EQ(call_stats.rttMs, stats.rtt_ms);
EXPECT_EQ(static_cast<int32_t>(FakeVoiceEngine::kSendSpeechInputLevel),
stats.audio_level);
EXPECT_EQ(-1, stats.aec_quality_min);
EXPECT_EQ(FakeVoiceEngine::kSendEchoDelayMedian, stats.echo_delay_median_ms);
EXPECT_EQ(FakeVoiceEngine::kSendEchoDelayStdDev, stats.echo_delay_std_ms);
EXPECT_EQ(FakeVoiceEngine::kSendEchoReturnLoss, stats.echo_return_loss);
EXPECT_EQ(FakeVoiceEngine::kSendEchoReturnLossEnhancement,
stats.echo_return_loss_enhancement);
EXPECT_FALSE(stats.typing_noise_detected);
}
} // namespace test
} // namespace webrtc
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