Changed FakeVoiceEngine into a MockVoiceEngine.
BUG=webrtc:4690
R=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1402403008 .
Cr-Commit-Position: refs/heads/master@{#10491}
diff --git a/webrtc/audio/audio_receive_stream_unittest.cc b/webrtc/audio/audio_receive_stream_unittest.cc
index 4e267f1..0bb9b51 100644
--- a/webrtc/audio/audio_receive_stream_unittest.cc
+++ b/webrtc/audio/audio_receive_stream_unittest.cc
@@ -14,12 +14,15 @@
#include "webrtc/audio/conversion.h"
#include "webrtc/modules/remote_bitrate_estimator/include/mock/mock_remote_bitrate_estimator.h"
#include "webrtc/modules/rtp_rtcp/source/byte_io.h"
-#include "webrtc/test/fake_voice_engine.h"
+#include "webrtc/test/mock_voice_engine.h"
+namespace webrtc {
+namespace test {
namespace {
-using webrtc::ByteWriter;
-
+const int kChannelId = 2;
+const uint32_t kRemoteSsrc = 1234;
+const uint32_t kLocalSsrc = 5678;
const size_t kAbsoluteSendTimeLength = 4;
void BuildAbsoluteSendTimeExtension(uint8_t* buffer,
@@ -58,39 +61,38 @@
}
} // namespace
-namespace webrtc {
-namespace test {
-
TEST(AudioReceiveStreamTest, ConfigToString) {
const int kAbsSendTimeId = 3;
AudioReceiveStream::Config config;
- config.rtp.remote_ssrc = 1234;
- config.rtp.local_ssrc = 5678;
+ config.rtp.remote_ssrc = kRemoteSsrc;
+ config.rtp.local_ssrc = kLocalSsrc;
config.rtp.extensions.push_back(
RtpExtension(RtpExtension::kAbsSendTime, kAbsSendTimeId));
- config.voe_channel_id = 1;
+ config.voe_channel_id = kChannelId;
config.combined_audio_video_bwe = true;
- EXPECT_EQ("{rtp: {remote_ssrc: 1234, local_ssrc: 5678, extensions: [{name: "
+ EXPECT_EQ(
+ "{rtp: {remote_ssrc: 1234, local_ssrc: 5678, extensions: [{name: "
"http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time, id: 3}]}, "
"receive_transport: nullptr, rtcp_send_transport: nullptr, "
- "voe_channel_id: 1, combined_audio_video_bwe: true}", config.ToString());
+ "voe_channel_id: 2, combined_audio_video_bwe: true}",
+ config.ToString());
}
TEST(AudioReceiveStreamTest, ConstructDestruct) {
MockRemoteBitrateEstimator remote_bitrate_estimator;
- FakeVoiceEngine voice_engine;
+ MockVoiceEngine voice_engine;
AudioReceiveStream::Config config;
- config.voe_channel_id = 1;
+ config.voe_channel_id = kChannelId;
internal::AudioReceiveStream recv_stream(&remote_bitrate_estimator, config,
&voice_engine);
}
TEST(AudioReceiveStreamTest, AudioPacketUpdatesBweWithTimestamp) {
MockRemoteBitrateEstimator remote_bitrate_estimator;
- FakeVoiceEngine voice_engine;
+ MockVoiceEngine voice_engine;
AudioReceiveStream::Config config;
config.combined_audio_video_bwe = true;
- config.voe_channel_id = FakeVoiceEngine::kRecvChannelId;
+ config.voe_channel_id = kChannelId;
const int kAbsSendTimeId = 3;
config.rtp.extensions.push_back(
RtpExtension(RtpExtension::kAbsSendTime, kAbsSendTimeId));
@@ -102,60 +104,100 @@
PacketTime packet_time(5678000, 0);
const size_t kExpectedHeaderLength = 20;
EXPECT_CALL(remote_bitrate_estimator,
- IncomingPacket(packet_time.timestamp / 1000,
- sizeof(rtp_packet) - kExpectedHeaderLength, testing::_, false))
+ IncomingPacket(packet_time.timestamp / 1000,
+ sizeof(rtp_packet) - kExpectedHeaderLength,
+ testing::_, false))
.Times(1);
EXPECT_TRUE(
recv_stream.DeliverRtp(rtp_packet, sizeof(rtp_packet), packet_time));
}
TEST(AudioReceiveStreamTest, GetStats) {
+ const int kJitterBufferDelay = -7;
+ const int kPlayoutBufferDelay = 302;
+ const unsigned int kSpeechOutputLevel = 99;
+ const CallStatistics kCallStats = {345, 678, 901, 234, -12,
+ 3456, 7890, 567, 890, 123};
+
+ const CodecInst kCodecInst = {123, "codec_name_recv", 96000, -187, -198,
+ -103};
+
+ const NetworkStatistics kNetworkStats = {
+ 123, 456, false, 0, 0, 789, 12, 345, 678, 901, -1, -1, -1, -1, -1, 0};
+
+ webrtc::AudioDecodingCallStats audio_decode_stats;
+ {
+ audio_decode_stats.calls_to_silence_generator = 234;
+ audio_decode_stats.calls_to_neteq = 567;
+ audio_decode_stats.decoded_normal = 890;
+ audio_decode_stats.decoded_plc = 123;
+ audio_decode_stats.decoded_cng = 456;
+ audio_decode_stats.decoded_plc_cng = 789;
+ }
+
MockRemoteBitrateEstimator remote_bitrate_estimator;
- FakeVoiceEngine voice_engine;
+ MockVoiceEngine voice_engine;
AudioReceiveStream::Config config;
- config.rtp.remote_ssrc = FakeVoiceEngine::kRecvSsrc;
- config.voe_channel_id = FakeVoiceEngine::kRecvChannelId;
+ config.rtp.remote_ssrc = kRemoteSsrc;
+ config.voe_channel_id = kChannelId;
internal::AudioReceiveStream recv_stream(&remote_bitrate_estimator, config,
&voice_engine);
+ using testing::_;
+ using testing::DoAll;
+ using testing::Return;
+ using testing::SetArgPointee;
+ using testing::SetArgReferee;
+ EXPECT_CALL(voice_engine, GetRemoteSSRC(kChannelId, _))
+ .WillOnce(DoAll(SetArgReferee<1>(0), Return(0)));
+ EXPECT_CALL(voice_engine, GetRTCPStatistics(kChannelId, _))
+ .WillOnce(DoAll(SetArgReferee<1>(kCallStats), Return(0)));
+ EXPECT_CALL(voice_engine, GetRecCodec(kChannelId, _))
+ .WillOnce(DoAll(SetArgReferee<1>(kCodecInst), Return(0)));
+ EXPECT_CALL(voice_engine, GetDelayEstimate(kChannelId, _, _))
+ .WillOnce(DoAll(SetArgPointee<1>(kJitterBufferDelay),
+ SetArgPointee<2>(kPlayoutBufferDelay), Return(0)));
+ EXPECT_CALL(voice_engine, GetSpeechOutputLevelFullRange(kChannelId, _))
+ .WillOnce(DoAll(SetArgReferee<1>(kSpeechOutputLevel), Return(0)));
+ EXPECT_CALL(voice_engine, GetNetworkStatistics(kChannelId, _))
+ .WillOnce(DoAll(SetArgReferee<1>(kNetworkStats), Return(0)));
+ EXPECT_CALL(voice_engine, GetDecodingCallStatistics(kChannelId, _))
+ .WillOnce(DoAll(SetArgPointee<1>(audio_decode_stats), Return(0)));
+
AudioReceiveStream::Stats stats = recv_stream.GetStats();
- const CallStatistics& call_stats = FakeVoiceEngine::kRecvCallStats;
- const CodecInst& codec_inst = FakeVoiceEngine::kRecvCodecInst;
- const NetworkStatistics& net_stats = FakeVoiceEngine::kRecvNetworkStats;
- const AudioDecodingCallStats& decode_stats =
- FakeVoiceEngine::kRecvAudioDecodingCallStats;
- EXPECT_EQ(FakeVoiceEngine::kRecvSsrc, stats.remote_ssrc);
- EXPECT_EQ(static_cast<int64_t>(call_stats.bytesReceived), stats.bytes_rcvd);
- EXPECT_EQ(static_cast<uint32_t>(call_stats.packetsReceived),
+ EXPECT_EQ(kRemoteSsrc, stats.remote_ssrc);
+ EXPECT_EQ(static_cast<int64_t>(kCallStats.bytesReceived), stats.bytes_rcvd);
+ EXPECT_EQ(static_cast<uint32_t>(kCallStats.packetsReceived),
stats.packets_rcvd);
- EXPECT_EQ(call_stats.cumulativeLost, stats.packets_lost);
- EXPECT_EQ(Q8ToFloat(call_stats.fractionLost), stats.fraction_lost);
- EXPECT_EQ(std::string(codec_inst.plname), stats.codec_name);
- EXPECT_EQ(call_stats.extendedMax, stats.ext_seqnum);
- EXPECT_EQ(call_stats.jitterSamples / (codec_inst.plfreq / 1000),
+ EXPECT_EQ(kCallStats.cumulativeLost, stats.packets_lost);
+ EXPECT_EQ(Q8ToFloat(kCallStats.fractionLost), stats.fraction_lost);
+ EXPECT_EQ(std::string(kCodecInst.plname), stats.codec_name);
+ EXPECT_EQ(kCallStats.extendedMax, stats.ext_seqnum);
+ EXPECT_EQ(kCallStats.jitterSamples / (kCodecInst.plfreq / 1000),
stats.jitter_ms);
- EXPECT_EQ(net_stats.currentBufferSize, stats.jitter_buffer_ms);
- EXPECT_EQ(net_stats.preferredBufferSize, stats.jitter_buffer_preferred_ms);
- EXPECT_EQ(static_cast<uint32_t>(FakeVoiceEngine::kRecvJitterBufferDelay +
- FakeVoiceEngine::kRecvPlayoutBufferDelay), stats.delay_estimate_ms);
- EXPECT_EQ(static_cast<int32_t>(FakeVoiceEngine::kRecvSpeechOutputLevel),
- stats.audio_level);
- EXPECT_EQ(Q14ToFloat(net_stats.currentExpandRate), stats.expand_rate);
- EXPECT_EQ(Q14ToFloat(net_stats.currentSpeechExpandRate),
+ EXPECT_EQ(kNetworkStats.currentBufferSize, stats.jitter_buffer_ms);
+ EXPECT_EQ(kNetworkStats.preferredBufferSize,
+ stats.jitter_buffer_preferred_ms);
+ EXPECT_EQ(static_cast<uint32_t>(kJitterBufferDelay + kPlayoutBufferDelay),
+ stats.delay_estimate_ms);
+ EXPECT_EQ(static_cast<int32_t>(kSpeechOutputLevel), stats.audio_level);
+ EXPECT_EQ(Q14ToFloat(kNetworkStats.currentExpandRate), stats.expand_rate);
+ EXPECT_EQ(Q14ToFloat(kNetworkStats.currentSpeechExpandRate),
stats.speech_expand_rate);
- EXPECT_EQ(Q14ToFloat(net_stats.currentSecondaryDecodedRate),
+ EXPECT_EQ(Q14ToFloat(kNetworkStats.currentSecondaryDecodedRate),
stats.secondary_decoded_rate);
- EXPECT_EQ(Q14ToFloat(net_stats.currentAccelerateRate), stats.accelerate_rate);
- EXPECT_EQ(Q14ToFloat(net_stats.currentPreemptiveRate),
+ EXPECT_EQ(Q14ToFloat(kNetworkStats.currentAccelerateRate),
+ stats.accelerate_rate);
+ EXPECT_EQ(Q14ToFloat(kNetworkStats.currentPreemptiveRate),
stats.preemptive_expand_rate);
- EXPECT_EQ(decode_stats.calls_to_silence_generator,
+ EXPECT_EQ(audio_decode_stats.calls_to_silence_generator,
stats.decoding_calls_to_silence_generator);
- EXPECT_EQ(decode_stats.calls_to_neteq, stats.decoding_calls_to_neteq);
- EXPECT_EQ(decode_stats.decoded_normal, stats.decoding_normal);
- EXPECT_EQ(decode_stats.decoded_plc, stats.decoding_plc);
- EXPECT_EQ(decode_stats.decoded_cng, stats.decoding_cng);
- EXPECT_EQ(decode_stats.decoded_plc_cng, stats.decoding_plc_cng);
- EXPECT_EQ(call_stats.capture_start_ntp_time_ms_,
+ EXPECT_EQ(audio_decode_stats.calls_to_neteq, stats.decoding_calls_to_neteq);
+ EXPECT_EQ(audio_decode_stats.decoded_normal, stats.decoding_normal);
+ EXPECT_EQ(audio_decode_stats.decoded_plc, stats.decoding_plc);
+ EXPECT_EQ(audio_decode_stats.decoded_cng, stats.decoding_cng);
+ EXPECT_EQ(audio_decode_stats.decoded_plc_cng, stats.decoding_plc_cng);
+ EXPECT_EQ(kCallStats.capture_start_ntp_time_ms_,
stats.capture_start_ntp_time_ms);
}
} // namespace test
