Variable audio bitrate.

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 32 matching lines @@
 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;
 
+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()
       : simulated_clock_(123456),
         stream_config_(nullptr),
         congestion_controller_(&simulated_clock_,
                                &bitrate_observer_,
                                &remote_bitrate_observer_,
-                               &event_log_) {
+                               &event_log_),
+        bitrate_allocator_(&limit_observer_) {
     using testing::Invoke;
     using testing::StrEq;
 
     EXPECT_CALL(voice_engine_,
         RegisterVoiceEngineObserver(_)).WillOnce(Return(0));
     EXPECT_CALL(voice_engine_,
         DeRegisterVoiceEngineObserver()).WillOnce(Return(0));
     AudioState::Config config;
     config.voice_engine = &voice_engine_;
     audio_state_ = AudioState::Create(config);
@@ skipping 38 matching lines @@
     stream_config_.rtp.extensions.push_back(RtpExtension(
         RtpExtension::kTransportSequenceNumberUri, kTransportSequenceNumberId));
   }
 
   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_; }
 
   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));
   }
@@ skipping 34 matching lines @@
  private:
   SimulatedClock simulated_clock_;
   testing::StrictMock<MockVoiceEngine> voice_engine_;
   rtc::scoped_refptr<AudioState> audio_state_;
   AudioSendStream::Config stream_config_;
   testing::StrictMock<MockVoEChannelProxy>* channel_proxy_ = nullptr;
   testing::NiceMock<MockCongestionObserver> bitrate_observer_;
   testing::NiceMock<MockRemoteBitrateObserver> remote_bitrate_observer_;
   CongestionController congestion_controller_;
   MockRtcEventLog event_log_;
+  testing::NiceMock<MockLimitObserver> limit_observer_;
+  BitrateAllocator bitrate_allocator_;
 };
 }  // 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;
   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;
   internal::AudioSendStream send_stream(helper.config(), helper.audio_state(),
-                                        helper.congestion_controller());
+                                        helper.congestion_controller(),
+                                        helper.bitrate_allocator());
 }
 
 TEST(AudioSendStreamTest, SendTelephoneEvent) {
   ConfigHelper helper;
   internal::AudioSendStream send_stream(helper.config(), helper.audio_state(),
-                                        helper.congestion_controller());
+                                        helper.congestion_controller(),
+                                        helper.bitrate_allocator());
   helper.SetupMockForSendTelephoneEvent();
   EXPECT_TRUE(send_stream.SendTelephoneEvent(kTelephoneEventPayloadType,
       kTelephoneEventCode, kTelephoneEventDuration));
 }
 
 TEST(AudioSendStreamTest, SetMuted) {
   ConfigHelper helper;
   internal::AudioSendStream send_stream(helper.config(), helper.audio_state(),
-                                        helper.congestion_controller());
+                                        helper.congestion_controller(),
+                                        helper.bitrate_allocator());
   EXPECT_CALL(*helper.channel_proxy(), SetInputMute(true));
   send_stream.SetMuted(true);
 }
 
 TEST(AudioSendStreamTest, GetStats) {
   ConfigHelper helper;
   internal::AudioSendStream send_stream(helper.config(), helper.audio_state(),
-                                        helper.congestion_controller());
+                                        helper.congestion_controller(),
+                                        helper.bitrate_allocator());
   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(static_cast<int32_t>(kReportBlock.extended_highest_sequence_number),
             stats.ext_seqnum);
   EXPECT_EQ(static_cast<int32_t>(kReportBlock.interarrival_jitter /
                                  (kCodecInst.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.congestion_controller());
+                                        helper.congestion_controller(),
+                                        helper.bitrate_allocator());
   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);
 }
 }  // namespace test
 }  // namespace webrtc