Chromium Code Reviews

Side by Side Diff: webrtc/audio/audio_send_stream_unittest.cc

Issue 1403363003: Move VoiceEngineObserver into AudioSendStream so that we detect typing noises and return properly i… (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: better comments Created 5 years, 1 month ago
Use n/p to move between diff chunks; N/P to move between comments.
Jump to:
View unified diff |
OLDNEW
1 /* 1 /*
2 * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
3 * 3 *
4 * Use of this source code is governed by a BSD-style license 4 * Use of this source code is governed by a BSD-style license
5 * that can be found in the LICENSE file in the root of the source 5 * that can be found in the LICENSE file in the root of the source
6 * tree. An additional intellectual property rights grant can be found 6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may 7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree. 8 * be found in the AUTHORS file in the root of the source tree.
9 */ 9 */
10 10
11 #include "testing/gtest/include/gtest/gtest.h" 11 #include "testing/gtest/include/gtest/gtest.h"
12 12
13 #include "webrtc/audio/audio_send_stream.h" 13 #include "webrtc/audio/audio_send_stream.h"
14 #include "webrtc/audio/audio_state.h"
14 #include "webrtc/audio/conversion.h" 15 #include "webrtc/audio/conversion.h"
16 #include "webrtc/base/scoped_ptr.h"
15 #include "webrtc/test/fake_voice_engine.h" 17 #include "webrtc/test/fake_voice_engine.h"
16 18
19 namespace {
20
21 struct ConfigHelper {
22 ConfigHelper() : stream_config_(nullptr) {
23 webrtc::AudioState::Config config;
24 config.voice_engine = &voice_engine_;
25 audio_state_.reset(new webrtc::internal::AudioState(config));
26 }
27
28 webrtc::AudioSendStream::Config& config() {
29 return stream_config_;
30 }
31 webrtc::internal::AudioState* audio_state() {
32 return audio_state_.get();
33 }
34
35 private:
36 webrtc::test::FakeVoiceEngine voice_engine_;
37 rtc::scoped_ptr<webrtc::internal::AudioState> audio_state_;
38 webrtc::AudioSendStream::Config stream_config_;
39 };
40 } // namespace
41
17 namespace webrtc { 42 namespace webrtc {
18 namespace test { 43 namespace test {
19 44
20 TEST(AudioSendStreamTest, ConfigToString) { 45 TEST(AudioSendStreamTest, ConfigToString) {
21 const int kAbsSendTimeId = 3; 46 const int kAbsSendTimeId = 3;
22 AudioSendStream::Config config(nullptr); 47 AudioSendStream::Config config(nullptr);
23 config.rtp.ssrc = 1234; 48 config.rtp.ssrc = 1234;
24 config.rtp.extensions.push_back( 49 config.rtp.extensions.push_back(
25 RtpExtension(RtpExtension::kAbsSendTime, kAbsSendTimeId)); 50 RtpExtension(RtpExtension::kAbsSendTime, kAbsSendTimeId));
26 config.voe_channel_id = 1; 51 config.voe_channel_id = 1;
27 config.cng_payload_type = 42; 52 config.cng_payload_type = 42;
28 config.red_payload_type = 17; 53 config.red_payload_type = 17;
29 EXPECT_EQ("{rtp: {ssrc: 1234, extensions: [{name: " 54 EXPECT_EQ("{rtp: {ssrc: 1234, extensions: [{name: "
30 "http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time, id: 3}]}, " 55 "http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time, id: 3}]}, "
31 "voe_channel_id: 1, cng_payload_type: 42, red_payload_type: 17}", 56 "voe_channel_id: 1, cng_payload_type: 42, red_payload_type: 17}",
32 config.ToString()); 57 config.ToString());
33 } 58 }
34 59
35 TEST(AudioSendStreamTest, ConstructDestruct) { 60 TEST(AudioSendStreamTest, ConstructDestruct) {
36 FakeVoiceEngine voice_engine; 61 ConfigHelper helper;
37 AudioSendStream::Config config(nullptr); 62 helper.config().voe_channel_id = 1;
38 config.voe_channel_id = 1; 63 internal::AudioSendStream send_stream(helper.config(), helper.audio_state());
39 internal::AudioSendStream send_stream(config, &voice_engine);
40 } 64 }
41 65
42 TEST(AudioSendStreamTest, GetStats) { 66 TEST(AudioSendStreamTest, GetStats) {
43 FakeVoiceEngine voice_engine; 67 ConfigHelper helper;
44 AudioSendStream::Config config(nullptr); 68 helper.config().rtp.ssrc = FakeVoiceEngine::kSendSsrc;
45 config.rtp.ssrc = FakeVoiceEngine::kSendSsrc; 69 helper.config().voe_channel_id = FakeVoiceEngine::kSendChannelId;
46 config.voe_channel_id = FakeVoiceEngine::kSendChannelId; 70 internal::AudioSendStream send_stream(helper.config(), helper.audio_state());
47 internal::AudioSendStream send_stream(config, &voice_engine);
48 71
49 AudioSendStream::Stats stats = send_stream.GetStats(); 72 AudioSendStream::Stats stats = send_stream.GetStats();
50 const CallStatistics& call_stats = FakeVoiceEngine::kSendCallStats; 73 const CallStatistics& call_stats = FakeVoiceEngine::kSendCallStats;
51 const CodecInst& codec_inst = FakeVoiceEngine::kSendCodecInst; 74 const CodecInst& codec_inst = FakeVoiceEngine::kSendCodecInst;
52 const ReportBlock& report_block = FakeVoiceEngine::kSendReportBlock; 75 const ReportBlock& report_block = FakeVoiceEngine::kSendReportBlock;
53 EXPECT_EQ(FakeVoiceEngine::kSendSsrc, stats.local_ssrc); 76 EXPECT_EQ(FakeVoiceEngine::kSendSsrc, stats.local_ssrc);
54 EXPECT_EQ(static_cast<int64_t>(call_stats.bytesSent), stats.bytes_sent); 77 EXPECT_EQ(static_cast<int64_t>(call_stats.bytesSent), stats.bytes_sent);
55 EXPECT_EQ(call_stats.packetsSent, stats.packets_sent); 78 EXPECT_EQ(call_stats.packetsSent, stats.packets_sent);
56 EXPECT_EQ(static_cast<int32_t>(report_block.cumulative_num_packets_lost), 79 EXPECT_EQ(static_cast<int32_t>(report_block.cumulative_num_packets_lost),
57 stats.packets_lost); 80 stats.packets_lost);
58 EXPECT_EQ(Q8ToFloat(report_block.fraction_lost), stats.fraction_lost); 81 EXPECT_EQ(Q8ToFloat(report_block.fraction_lost), stats.fraction_lost);
59 EXPECT_EQ(std::string(codec_inst.plname), stats.codec_name); 82 EXPECT_EQ(std::string(codec_inst.plname), stats.codec_name);
60 EXPECT_EQ(static_cast<int32_t>(report_block.extended_highest_sequence_number), 83 EXPECT_EQ(static_cast<int32_t>(report_block.extended_highest_sequence_number),
61 stats.ext_seqnum); 84 stats.ext_seqnum);
62 EXPECT_EQ(static_cast<int32_t>(report_block.interarrival_jitter / 85 EXPECT_EQ(static_cast<int32_t>(report_block.interarrival_jitter /
63 (codec_inst.plfreq / 1000)), stats.jitter_ms); 86 (codec_inst.plfreq / 1000)), stats.jitter_ms);
64 EXPECT_EQ(call_stats.rttMs, stats.rtt_ms); 87 EXPECT_EQ(call_stats.rttMs, stats.rtt_ms);
65 EXPECT_EQ(static_cast<int32_t>(FakeVoiceEngine::kSendSpeechInputLevel), 88 EXPECT_EQ(static_cast<int32_t>(FakeVoiceEngine::kSendSpeechInputLevel),
66 stats.audio_level); 89 stats.audio_level);
67 EXPECT_EQ(-1, stats.aec_quality_min); 90 EXPECT_EQ(-1, stats.aec_quality_min);
68 EXPECT_EQ(FakeVoiceEngine::kSendEchoDelayMedian, stats.echo_delay_median_ms); 91 EXPECT_EQ(FakeVoiceEngine::kSendEchoDelayMedian, stats.echo_delay_median_ms);
69 EXPECT_EQ(FakeVoiceEngine::kSendEchoDelayStdDev, stats.echo_delay_std_ms); 92 EXPECT_EQ(FakeVoiceEngine::kSendEchoDelayStdDev, stats.echo_delay_std_ms);
70 EXPECT_EQ(FakeVoiceEngine::kSendEchoReturnLoss, stats.echo_return_loss); 93 EXPECT_EQ(FakeVoiceEngine::kSendEchoReturnLoss, stats.echo_return_loss);
71 EXPECT_EQ(FakeVoiceEngine::kSendEchoReturnLossEnhancement, 94 EXPECT_EQ(FakeVoiceEngine::kSendEchoReturnLossEnhancement,
72 stats.echo_return_loss_enhancement); 95 stats.echo_return_loss_enhancement);
73 EXPECT_FALSE(stats.typing_noise_detected); 96 EXPECT_FALSE(stats.typing_noise_detected);
74 } 97 }
75 } // namespace test 98 } // namespace test
76 } // namespace webrtc 99 } // namespace webrtc
OLDNEW

Powered by Google App Engine