| /* |
| * 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 |