Revert of Implement AudioReceiveStream::GetStats().

webrtc/audio/audio_receive_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_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 {
-
-using webrtc::ByteWriter;
+namespace webrtc {
 
 const size_t kAbsoluteSendTimeLength = 4;
 
 void BuildAbsoluteSendTimeExtension(uint8_t* buffer,
                                     int id,
                                     uint32_t abs_send_time) {
   const size_t kRtpOneByteHeaderLength = 4;
   const uint16_t kRtpOneByteHeaderExtensionId = 0xBEDE;
   ByteWriter<uint16_t>::WriteBigEndian(buffer, kRtpOneByteHeaderExtensionId);
 
@@ skipping 10 matching lines @@
 size_t CreateRtpHeaderWithAbsSendTime(uint8_t* header,
                                       int extension_id,
                                       uint32_t abs_send_time) {
   header[0] = 0x80;   // Version 2.
   header[0] |= 0x10;  // Set extension bit.
   header[1] = 100;    // Payload type.
   header[1] |= 0x80;  // Marker bit is set.
   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 = webrtc::kRtpHeaderSize;
+  int32_t rtp_header_length = 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;
-  FakeVoiceEngine fve;
   AudioReceiveStream::Config config;
   config.combined_audio_video_bwe = true;
-  config.voe_channel_id = fve.kReceiveChannelId;
+  config.voe_channel_id = 1;
   const int kAbsSendTimeId = 3;
   config.rtp.extensions.push_back(
       RtpExtension(RtpExtension::kAbsSendTime, kAbsSendTimeId));
-  internal::AudioReceiveStream recv_stream(&rbe, config, &fve);
+  internal::AudioReceiveStream recv_stream(&rbe, config);
   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))
       .Times(1);
   EXPECT_TRUE(
       recv_stream.DeliverRtp(rtp_packet, sizeof(rtp_packet), packet_time));
 }
-
-TEST(AudioReceiveStreamTest, GetStats) {
-  const uint32_t kSsrc1 = 667;
-
-  MockRemoteBitrateEstimator rbe;
-  FakeVoiceEngine fve;
-  AudioReceiveStream::Config config;
-  config.rtp.remote_ssrc = kSsrc1;
-  config.voe_channel_id = fve.kReceiveChannelId;
-  internal::AudioReceiveStream recv_stream(&rbe, config, &fve);
-
-  AudioReceiveStream::Stats stats = recv_stream.GetStats();
-  const CallStatistics& call_stats = fve.GetRecvCallStats();
-  const CodecInst& codec_inst = fve.GetRecvRecCodecInst();
-  const NetworkStatistics& net_stats = fve.GetRecvNetworkStats();
-  const AudioDecodingCallStats& decode_stats =
-      fve.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>(fve.kRecvJitterBufferDelay +
-      fve.kRecvPlayoutBufferDelay), stats.delay_estimate_ms);
-  EXPECT_EQ(static_cast<int32_t>(fve.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