Moving WebRtcVoiceMediaChannel::SendSetCodec to AudioSendStream.

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.
  */
 
@@ skipping 26 matching lines @@
 const int kAudioLevelId = 2;
 const int kAbsSendTimeId = 3;
 const int kTransportSequenceNumberId = 4;
 const int kEchoDelayMedian = 254;
 const int kEchoDelayStdDev = -3;
 const int kEchoReturnLoss = -65;
 const int kEchoReturnLossEnhancement = 101;
 const unsigned int kSpeechInputLevel = 96;
 const CallStatistics kCallStats = {
     1345,  1678,  1901, 1234,  112, 13456, 17890, 1567, -1890, -1123};
-const CodecInst kCodecInst = {-121, "codec_name_send", 48000, -231, 0, -671};
 const ReportBlock kReportBlock = {456, 780, 123, 567, 890, 132, 143, 13354};
 const int kTelephoneEventPayloadType = 123;
 const int kTelephoneEventCode = 45;
 const int kTelephoneEventDuration = 6789;
+constexpr CodecInst kOpusCodec = {111, "opus", 48000, 960, 2, 64000};
 
 class MockLimitObserver : public BitrateAllocator::LimitObserver {
  public:
   MOCK_METHOD2(OnAllocationLimitsChanged,
                void(uint32_t min_send_bitrate_bps,
                     uint32_t max_padding_bitrate_bps));
 };
 
 struct ConfigHelper {
   ConfigHelper()
@@ skipping 42 matching lines @@
           EXPECT_CALL(*channel_proxy_, RegisterExternalTransport(nullptr))
               .Times(1);
           EXPECT_CALL(*channel_proxy_, DeRegisterExternalTransport())
               .Times(1);
           EXPECT_CALL(*channel_proxy_, SetRtcEventLog(testing::NotNull()))
               .Times(1);
           EXPECT_CALL(*channel_proxy_, SetRtcEventLog(testing::IsNull()))
               .Times(1);  // Destructor resets the event log
           return channel_proxy_;
         }));
+    SetupMockForSetupSendCodec();
     stream_config_.voe_channel_id = kChannelId;
     stream_config_.rtp.ssrc = kSsrc;
     stream_config_.rtp.nack.rtp_history_ms = 200;
     stream_config_.rtp.c_name = kCName;
     stream_config_.rtp.extensions.push_back(
         RtpExtension(RtpExtension::kAudioLevelUri, kAudioLevelId));
     stream_config_.rtp.extensions.push_back(
         RtpExtension(RtpExtension::kAbsSendTimeUri, kAbsSendTimeId));
     stream_config_.rtp.extensions.push_back(RtpExtension(
         RtpExtension::kTransportSequenceNumberUri, kTransportSequenceNumberId));
+    stream_config_.send_codec_spec.codec_inst = kOpusCodec;
   }
 
   AudioSendStream::Config& config() { return stream_config_; }
   rtc::scoped_refptr<AudioState> audio_state() { return audio_state_; }
   MockVoEChannelProxy* channel_proxy() { return channel_proxy_; }
   CongestionController* congestion_controller() {
     return &congestion_controller_;
   }
   BitrateAllocator* bitrate_allocator() { return &bitrate_allocator_; }
   rtc::TaskQueue* worker_queue() { return &worker_queue_; }
   RtcEventLog* event_log() { return &event_log_; }
 
+  void SetupMockForSetupSendCodec() {
+    EXPECT_CALL(voice_engine_, SetVADStatus(kChannelId, _, _, _))
+        .WillRepeatedly(Return(0));
+    EXPECT_CALL(voice_engine_, SetFECStatus(kChannelId, _))
+        .WillRepeatedly(Return(0));
+    EXPECT_CALL(voice_engine_, SetSendCodec(kChannelId, _))
+        .WillRepeatedly(Return(0));
+    // Let |GetSendCodec| return -1 for the first time to indicate that no send
+    // codec has been set.
+    EXPECT_CALL(voice_engine_, GetSendCodec(kChannelId, _))
+        .WillOnce(Return(-1));
+  }
+
   void SetupMockForSendTelephoneEvent() {
     EXPECT_TRUE(channel_proxy_);
     EXPECT_CALL(*channel_proxy_,
         SetSendTelephoneEventPayloadType(kTelephoneEventPayloadType))
             .WillOnce(Return(true));
     EXPECT_CALL(*channel_proxy_,
         SendTelephoneEventOutband(kTelephoneEventCode, kTelephoneEventDuration))
             .WillOnce(Return(true));
   }
 
   void SetupMockForGetStats() {
     using testing::DoAll;
     using testing::SetArgReferee;
 
     std::vector<ReportBlock> report_blocks;
     webrtc::ReportBlock block = kReportBlock;
     report_blocks.push_back(block);  // Has wrong SSRC.
     block.source_SSRC = kSsrc;
     report_blocks.push_back(block);  // Correct block.
     block.fraction_lost = 0;
     report_blocks.push_back(block);  // Duplicate SSRC, bad fraction_lost.
 
     EXPECT_TRUE(channel_proxy_);
     EXPECT_CALL(*channel_proxy_, GetRTCPStatistics())
         .WillRepeatedly(Return(kCallStats));
     EXPECT_CALL(*channel_proxy_, GetRemoteRTCPReportBlocks())
         .WillRepeatedly(Return(report_blocks));
 
     EXPECT_CALL(voice_engine_, GetSendCodec(kChannelId, _))
-        .WillRepeatedly(DoAll(SetArgReferee<1>(kCodecInst), Return(0)));
+        .WillRepeatedly(DoAll(SetArgReferee<1>(kOpusCodec), Return(0)));
     EXPECT_CALL(voice_engine_, GetSpeechInputLevelFullRange(_))
         .WillRepeatedly(DoAll(SetArgReferee<0>(kSpeechInputLevel), Return(0)));
     EXPECT_CALL(voice_engine_, GetEcMetricsStatus(_))
         .WillRepeatedly(DoAll(SetArgReferee<0>(true), Return(0)));
     EXPECT_CALL(voice_engine_, GetEchoMetrics(_, _, _, _))
         .WillRepeatedly(DoAll(SetArgReferee<0>(kEchoReturnLoss),
                         SetArgReferee<1>(kEchoReturnLossEnhancement),
                         Return(0)));
     EXPECT_CALL(voice_engine_, GetEcDelayMetrics(_, _, _))
         .WillRepeatedly(DoAll(SetArgReferee<0>(kEchoDelayMedian),
@@ skipping 18 matching lines @@
 };
 }  // namespace
 
 TEST(AudioSendStreamTest, ConfigToString) {
   AudioSendStream::Config config(nullptr);
   config.rtp.ssrc = kSsrc;
   config.rtp.extensions.push_back(
       RtpExtension(RtpExtension::kAbsSendTimeUri, kAbsSendTimeId));
   config.rtp.c_name = kCName;
   config.voe_channel_id = kChannelId;
-  config.cng_payload_type = 42;
+  config.send_codec_spec.cng_payload_type = 42;
   EXPECT_EQ(
       "{rtp: {ssrc: 1234, extensions: [{uri: "
       "http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time, id: 3}], "
       "nack: {rtp_history_ms: 0}, c_name: foo_name}, voe_channel_id: 1, "
       "cng_payload_type: 42}",
       config.ToString());
 }
 
 TEST(AudioSendStreamTest, ConstructDestruct) {
   ConfigHelper helper;
@@ skipping 31 matching lines @@
       helper.congestion_controller(), helper.bitrate_allocator(),
       helper.event_log());
   helper.SetupMockForGetStats();
   AudioSendStream::Stats stats = send_stream.GetStats();
   EXPECT_EQ(kSsrc, stats.local_ssrc);
   EXPECT_EQ(static_cast<int64_t>(kCallStats.bytesSent), stats.bytes_sent);
   EXPECT_EQ(kCallStats.packetsSent, stats.packets_sent);
   EXPECT_EQ(static_cast<int32_t>(kReportBlock.cumulative_num_packets_lost),
             stats.packets_lost);
   EXPECT_EQ(Q8ToFloat(kReportBlock.fraction_lost), stats.fraction_lost);
-  EXPECT_EQ(std::string(kCodecInst.plname), stats.codec_name);
+  EXPECT_EQ(std::string(kOpusCodec.plname), stats.codec_name);
   EXPECT_EQ(static_cast<int32_t>(kReportBlock.extended_highest_sequence_number),
             stats.ext_seqnum);
   EXPECT_EQ(static_cast<int32_t>(kReportBlock.interarrival_jitter /
-                                 (kCodecInst.plfreq / 1000)),
+                                 (kOpusCodec.plfreq / 1000)),
             stats.jitter_ms);
   EXPECT_EQ(kCallStats.rttMs, stats.rtt_ms);
   EXPECT_EQ(static_cast<int32_t>(kSpeechInputLevel), stats.audio_level);
   EXPECT_EQ(-1, stats.aec_quality_min);
   EXPECT_EQ(kEchoDelayMedian, stats.echo_delay_median_ms);
   EXPECT_EQ(kEchoDelayStdDev, stats.echo_delay_std_ms);
   EXPECT_EQ(kEchoReturnLoss, stats.echo_return_loss);
   EXPECT_EQ(kEchoReturnLossEnhancement, stats.echo_return_loss_enhancement);
   EXPECT_FALSE(stats.typing_noise_detected);
 }
 
 TEST(AudioSendStreamTest, GetStatsTypingNoiseDetected) {
   ConfigHelper helper;
   internal::AudioSendStream send_stream(
       helper.config(), helper.audio_state(), helper.worker_queue(),
       helper.congestion_controller(), helper.bitrate_allocator(),
       helper.event_log());
   helper.SetupMockForGetStats();
   EXPECT_FALSE(send_stream.GetStats().typing_noise_detected);
 
   internal::AudioState* internal_audio_state =
       static_cast<internal::AudioState*>(helper.audio_state().get());
   VoiceEngineObserver* voe_observer =
       static_cast<VoiceEngineObserver*>(internal_audio_state);
   voe_observer->CallbackOnError(-1, VE_TYPING_NOISE_WARNING);
   EXPECT_TRUE(send_stream.GetStats().typing_noise_detected);
   voe_observer->CallbackOnError(-1, VE_TYPING_NOISE_OFF_WARNING);
   EXPECT_FALSE(send_stream.GetStats().typing_noise_detected);
 }
+
+// TODO(minyue): Add tests on logics involved in SetSendCodecs.
+
 }  // namespace test
 }  // namespace webrtc