Implement AudioSendStream::GetStats().

webrtc/audio/audio_send_stream_unittest.cc

 /*
  *  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_GT(config.ToString().size(), 0u);
 }
 
 TEST(AudioSendStreamTest, ConstructDestruct) {
+  FakeVoiceEngine voice_engine;
   AudioSendStream::Config config(nullptr);
   config.voe_channel_id = 1;
-  internal::AudioSendStream send_stream(config);
+  internal::AudioSendStream send_stream(config, &voice_engine);
 }
+
+TEST(AudioSendStreamTest, GetStats) {
+  FakeVoiceEngine voice_engine;
+  AudioSendStream::Config config(nullptr);
+  config.rtp.ssrc = voice_engine.kSendSsrc;
+  config.voe_channel_id = voice_engine.kSendChannelId;
+  internal::AudioSendStream send_stream(config, &voice_engine);
+
+  AudioSendStream::Stats stats = send_stream.GetStats();
+  const CallStatistics& call_stats = voice_engine.GetSendCallStats();
+  const CodecInst& codec_inst = voice_engine.GetSendCodecInst();
+  const ReportBlock& report_block = voice_engine.GetSendReportBlock();
+  EXPECT_EQ(voice_engine.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>(voice_engine.kSendSpeechInputLevel),
+            stats.audio_level);
+  EXPECT_EQ(-1, stats.aec_quality_min);
+  EXPECT_EQ(voice_engine.kSendEchoDelayMedian, stats.echo_delay_median_ms);
+  EXPECT_EQ(voice_engine.kSendEchoDelayStdDev, stats.echo_delay_std_ms);
+  EXPECT_EQ(voice_engine.kSendEchoReturnLoss, stats.echo_return_loss);
+  EXPECT_EQ(voice_engine.kSendEchoReturnLossEnhancement,
+            stats.echo_return_loss_enhancement);
+  EXPECT_EQ(false, stats.typing_noise_detected);
+}
+}  // namespace test
 }  // namespace webrtc