Re-Land: Implement AudioReceiveStream::GetStats().
R=tommi@webrtc.org
BUG=webrtc:4690
Committed: https://chromium.googlesource.com/external/webrtc/+/a457752f4afc496ed7f4d6b584b08d8635f18cc0
Review URL: https://codereview.webrtc.org/1390753002 .
Cr-Commit-Position: refs/heads/master@{#10369}
diff --git a/webrtc/audio/audio_receive_stream_unittest.cc b/webrtc/audio/audio_receive_stream_unittest.cc
index d6cce69..8809b35 100644
--- a/webrtc/audio/audio_receive_stream_unittest.cc
+++ b/webrtc/audio/audio_receive_stream_unittest.cc
@@ -11,10 +11,14 @@
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/audio/audio_receive_stream.h"
+#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"
-namespace webrtc {
+namespace {
+
+using webrtc::ByteWriter;
const size_t kAbsoluteSendTimeLength = 4;
@@ -45,33 +49,93 @@
ByteWriter<uint16_t>::WriteBigEndian(header + 2, 0x1234); // Sequence number.
ByteWriter<uint32_t>::WriteBigEndian(header + 4, 0x5678); // Timestamp.
ByteWriter<uint32_t>::WriteBigEndian(header + 8, 0x4321); // SSRC.
- int32_t rtp_header_length = kRtpHeaderSize;
+ int32_t rtp_header_length = webrtc::kRtpHeaderSize;
BuildAbsoluteSendTimeExtension(header + rtp_header_length, extension_id,
abs_send_time);
rtp_header_length += kAbsoluteSendTimeLength;
return rtp_header_length;
}
+} // namespace
+
+namespace webrtc {
+namespace test {
TEST(AudioReceiveStreamTest, AudioPacketUpdatesBweWithTimestamp) {
- MockRemoteBitrateEstimator rbe;
+ MockRemoteBitrateEstimator remote_bitrate_estimator;
+ FakeVoiceEngine voice_engine;
AudioReceiveStream::Config config;
config.combined_audio_video_bwe = true;
- config.voe_channel_id = 1;
+ config.voe_channel_id = voice_engine.kReceiveChannelId;
const int kAbsSendTimeId = 3;
config.rtp.extensions.push_back(
RtpExtension(RtpExtension::kAbsSendTime, kAbsSendTimeId));
- internal::AudioReceiveStream recv_stream(&rbe, config);
+ internal::AudioReceiveStream recv_stream(&remote_bitrate_estimator, config,
+ &voice_engine);
uint8_t rtp_packet[30];
const int kAbsSendTimeValue = 1234;
CreateRtpHeaderWithAbsSendTime(rtp_packet, kAbsSendTimeId, kAbsSendTimeValue);
PacketTime packet_time(5678000, 0);
const size_t kExpectedHeaderLength = 20;
- EXPECT_CALL(rbe, IncomingPacket(packet_time.timestamp / 1000,
- sizeof(rtp_packet) - kExpectedHeaderLength,
- testing::_, false))
+ EXPECT_CALL(remote_bitrate_estimator,
+ 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 uint32_t kSsrc1 = 667;
+
+ MockRemoteBitrateEstimator remote_bitrate_estimator;
+ FakeVoiceEngine voice_engine;
+ AudioReceiveStream::Config config;
+ config.rtp.remote_ssrc = kSsrc1;
+ config.voe_channel_id = voice_engine.kReceiveChannelId;
+ internal::AudioReceiveStream recv_stream(&remote_bitrate_estimator, config,
+ &voice_engine);
+
+ AudioReceiveStream::Stats stats = recv_stream.GetStats();
+ const CallStatistics& call_stats = voice_engine.GetRecvCallStats();
+ const CodecInst& codec_inst = voice_engine.GetRecvRecCodecInst();
+ const NetworkStatistics& net_stats = voice_engine.GetRecvNetworkStats();
+ const AudioDecodingCallStats& decode_stats =
+ voice_engine.GetRecvAudioDecodingCallStats();
+ EXPECT_EQ(kSsrc1, stats.remote_ssrc);
+ EXPECT_EQ(static_cast<int64_t>(call_stats.bytesReceived), stats.bytes_rcvd);
+ EXPECT_EQ(static_cast<uint32_t>(call_stats.packetsReceived),
+ stats.packets_rcvd);
+ EXPECT_EQ(call_stats.cumulativeLost, stats.packets_lost);
+ EXPECT_EQ(static_cast<float>(call_stats.fractionLost) / 256,
+ 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),
+ 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>(voice_engine.kRecvJitterBufferDelay +
+ voice_engine.kRecvPlayoutBufferDelay), stats.delay_estimate_ms);
+ EXPECT_EQ(static_cast<int32_t>(voice_engine.kRecvSpeechOutputLevel),
+ stats.audio_level);
+ EXPECT_EQ(Q14ToFloat(net_stats.currentExpandRate), stats.expand_rate);
+ EXPECT_EQ(Q14ToFloat(net_stats.currentSpeechExpandRate),
+ stats.speech_expand_rate);
+ EXPECT_EQ(Q14ToFloat(net_stats.currentSecondaryDecodedRate),
+ stats.secondary_decoded_rate);
+ EXPECT_EQ(Q14ToFloat(net_stats.currentAccelerateRate), stats.accelerate_rate);
+ EXPECT_EQ(Q14ToFloat(net_stats.currentPreemptiveRate),
+ stats.preemptive_expand_rate);
+ EXPECT_EQ(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_,
+ stats.capture_start_ntp_time_ms);
+}
+} // namespace test
} // namespace webrtc