ACM: Removing runtime APIs related to playout mode

The playout mode in NetEq can still be set through the constructor
configuration.

BUG=webrtc:3520

Review URL: https://codereview.webrtc.org/1362943004

Cr-Commit-Position: refs/heads/master@{#10089}

webrtc/modules/audio_coding/main/acm2/acm_receiver.cc

@@ -213,51 +213,6 @@ int AcmReceiver::current_sample_rate_hz() const {
   return current_sample_rate_hz_;
 }
 
-// TODO(turajs): use one set of enumerators, e.g. the one defined in
-// common_types.h
-// TODO(henrik.lundin): This method is not used any longer. The call hierarchy
-// stops in voe::Channel::SetNetEQPlayoutMode(). Remove it.
-void AcmReceiver::SetPlayoutMode(AudioPlayoutMode mode) {
-  enum NetEqPlayoutMode playout_mode = kPlayoutOn;
-  switch (mode) {
-    case voice:
-      playout_mode = kPlayoutOn;
-      break;
-    case fax:  // No change to background noise mode.
-      playout_mode = kPlayoutFax;
-      break;
-    case streaming:
-      playout_mode = kPlayoutStreaming;
-      break;
-    case off:
-      playout_mode = kPlayoutOff;
-      break;
-  }
-  neteq_->SetPlayoutMode(playout_mode);
-}
-
-AudioPlayoutMode AcmReceiver::PlayoutMode() const {
-  AudioPlayoutMode acm_mode = voice;
-  NetEqPlayoutMode mode = neteq_->PlayoutMode();
-  switch (mode) {
-    case kPlayoutOn:
-      acm_mode = voice;
-      break;
-    case kPlayoutOff:
-      acm_mode = off;
-      break;
-    case kPlayoutFax:
-      acm_mode = fax;
-      break;
-    case kPlayoutStreaming:
-      acm_mode = streaming;
-      break;
-    default:
-      assert(false);
-  }
-  return acm_mode;
-}
-
 int AcmReceiver::InsertPacket(const WebRtcRTPHeader& rtp_header,
                               const uint8_t* incoming_payload,
                               size_t length_payload) {

webrtc/modules/audio_coding/main/acm2/acm_receiver.h

@@ -177,21 +177,6 @@ class AcmReceiver {
   //
   int current_sample_rate_hz() const;
 
-  //
-  // Sets the playout mode.
-  //
-  // Input:
-  //   - mode                 : an enumerator specifying the playout mode.
-  //
-  void SetPlayoutMode(AudioPlayoutMode mode);
-
-  //
-  // Get the current playout mode.
-  //
-  // Return value             : The current playout mode.
-  //
-  AudioPlayoutMode PlayoutMode() const;
-
   //
   // Get the current network statistics from NetEq.
   //

webrtc/modules/audio_coding/main/acm2/acm_receiver_unittest.cc

@@ -265,21 +265,6 @@ TEST_F(AcmReceiverTest, DISABLED_ON_ANDROID(SampleRate)) {
   }
 }
 
-// Verify that the playout mode is set correctly.
-TEST_F(AcmReceiverTest, DISABLED_ON_ANDROID(PlayoutMode)) {
-  receiver_->SetPlayoutMode(voice);
-  EXPECT_EQ(voice, receiver_->PlayoutMode());
-
-  receiver_->SetPlayoutMode(streaming);
-  EXPECT_EQ(streaming, receiver_->PlayoutMode());
-
-  receiver_->SetPlayoutMode(fax);
-  EXPECT_EQ(fax, receiver_->PlayoutMode());
-
-  receiver_->SetPlayoutMode(off);
-  EXPECT_EQ(off, receiver_->PlayoutMode());
-}
-
 TEST_F(AcmReceiverTest, DISABLED_ON_ANDROID(PostdecodingVad)) {
   receiver_->EnableVad();
   EXPECT_TRUE(receiver_->vad_enabled());

webrtc/modules/audio_coding/main/acm2/acm_receiver_unittest_oldapi.cc

@@ -264,21 +264,6 @@ TEST_F(AcmReceiverTestOldApi, DISABLED_ON_ANDROID(SampleRate)) {
   }
 }
 
-// Verify that the playout mode is set correctly.
-TEST_F(AcmReceiverTestOldApi, DISABLED_ON_ANDROID(PlayoutMode)) {
-  receiver_->SetPlayoutMode(voice);
-  EXPECT_EQ(voice, receiver_->PlayoutMode());
-
-  receiver_->SetPlayoutMode(streaming);
-  EXPECT_EQ(streaming, receiver_->PlayoutMode());
-
-  receiver_->SetPlayoutMode(fax);
-  EXPECT_EQ(fax, receiver_->PlayoutMode());
-
-  receiver_->SetPlayoutMode(off);
-  EXPECT_EQ(off, receiver_->PlayoutMode());
-}
-
 TEST_F(AcmReceiverTestOldApi, DISABLED_ON_ANDROID(PostdecodingVad)) {
   receiver_->EnableVad();
   EXPECT_TRUE(receiver_->vad_enabled());

webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc

@@ -659,17 +659,6 @@ int AudioCodingModuleImpl::SetMaximumPlayoutDelay(int time_ms) {
   return receiver_.SetMaximumDelay(time_ms);
 }
 
-// Set playout mode for: voice, fax, streaming or off.
-int AudioCodingModuleImpl::SetPlayoutMode(AudioPlayoutMode mode) {
-  receiver_.SetPlayoutMode(mode);
-  return 0;  // TODO(turajs): return value is for backward compatibility.
-}
-
-// Get playout mode voice, fax, streaming or off.
-AudioPlayoutMode AudioCodingModuleImpl::PlayoutMode() const {
-  return receiver_.PlayoutMode();
-}
-
 // Get 10 milliseconds of raw audio data to play out.
 // Automatic resample to the requested frequency.
 int AudioCodingModuleImpl::PlayoutData10Ms(int desired_freq_hz,

webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h

@@ -154,12 +154,6 @@ class AudioCodingModuleImpl final : public AudioCodingModule {
   // audio is accumulated in NetEq buffer, then starts decoding payloads.
   int SetInitialPlayoutDelay(int delay_ms) override;
 
-  // Set playout mode voice, fax.
-  int SetPlayoutMode(AudioPlayoutMode mode) override;
-
-  // Get playout mode voice, fax.
-  AudioPlayoutMode PlayoutMode() const override;
-
   // Get playout timestamp.
   int PlayoutTimestamp(uint32_t* timestamp) override;
 

webrtc/modules/audio_coding/main/interface/audio_coding_module.h

@@ -605,43 +605,6 @@ class AudioCodingModule {
   // TODO(tlegrand): Change function to return the timestamp.
   virtual int32_t PlayoutTimestamp(uint32_t* timestamp) = 0;
 
-  ///////////////////////////////////////////////////////////////////////////
-  // int32_t SetPlayoutMode()
-  // Call this API to set the playout mode. Playout mode could be optimized
-  // for i) voice, ii) FAX or iii) streaming. In Voice mode, NetEQ is
-  // optimized to deliver highest audio quality while maintaining a minimum
-  // delay. In FAX mode, NetEQ is optimized to have few delay changes as
-  // possible and maintain a constant delay, perhaps large relative to voice
-  // mode, to avoid PLC. In streaming mode, we tolerate a little more delay
-  // to achieve better jitter robustness.
-  //
-  // Input:
-  //   -mode               : playout mode. Possible inputs are:
-  //                         "voice",
-  //                         "fax" and
-  //                         "streaming".
-  //
-  // Return value:
-  //   -1 if failed to set the mode,
-  //    0 if succeeding.
-  //
-  virtual int32_t SetPlayoutMode(const AudioPlayoutMode mode) = 0;
-
-  ///////////////////////////////////////////////////////////////////////////
-  // AudioPlayoutMode PlayoutMode()
-  // Get playout mode, i.e. whether it is speech, FAX or streaming. See
-  // audio_coding_module_typedefs.h for definition of AudioPlayoutMode.
-  //
-  // Return value:
-  //   voice:       is for voice output,
-  //   fax:         a mode that is optimized for receiving FAX signals.
-  //                In this mode NetEq tries to maintain a constant high
-  //                delay to avoid PLC if possible.
-  //   streaming:   a mode that is suitable for streaming. In this mode we
-  //                accept longer delay to improve jitter robustness.
-  //
-  virtual AudioPlayoutMode PlayoutMode() const = 0;
-
   ///////////////////////////////////////////////////////////////////////////
   // int32_t PlayoutData10Ms(
   // Get 10 milliseconds of raw audio data for playout, at the given sampling

webrtc/modules/audio_coding/main/interface/audio_coding_module_typedefs.h

@@ -18,35 +18,6 @@
 
 namespace webrtc {
 
-///////////////////////////////////////////////////////////////////////////
-// enum AudioPlayoutMode
-// An enumerator for different playout modes.
-//
-// -voice       : This is the standard mode for VoIP calls. The trade-off
-//                between low delay and jitter robustness is optimized
-//                for high-quality two-way communication.
-//                NetEQs packet loss concealment and signal processing
-//                capabilities are fully employed.
-// -fax         : The fax mode is optimized for decodability of fax signals
-//                rather than for perceived audio quality. When this mode
-//                is selected, NetEQ will do as few delay changes as possible,
-//                trying to maintain a high and constant delay. Meanwhile,
-//                the packet loss concealment efforts are reduced.
-//
-// -streaming   : In the case of one-way communication such as passive
-//                conference participant, a webinar, or a streaming application,
-//                this mode can be used to improve the jitter robustness at
-//                the cost of increased delay.
-// -off         : Turns off most of NetEQ's features. Stuffs zeros for lost
-//                packets and during buffer increases.
-//
-enum AudioPlayoutMode {
-  voice = 0,
-  fax = 1,
-  streaming = 2,
-  off = 3,
-};
-
 ///////////////////////////////////////////////////////////////////////////
 // enum ACMSpeechType
 // An enumerator for possible labels of a decoded frame.

webrtc/modules/audio_coding/main/test/APITest.cc

@@ -424,7 +424,7 @@ void APITest::RunTest(char thread) {
   {
     WriteLockScoped cs(_apiTestRWLock);
     if (thread == 'A') {
-      _testNumA = (_testNumB + 1 + (rand() % 4)) % 5;
+      _testNumA = (_testNumB + 1 + (rand() % 3)) % 4;
       testNum = _testNumA;
 
       _movingDot[_dotPositionA] = ' ';
@@ -437,7 +437,7 @@ void APITest::RunTest(char thread) {
       _dotPositionA += _dotMoveDirectionA;
       _movingDot[_dotPositionA] = (_dotMoveDirectionA > 0) ? '>' : '<';
     } else {
-      _testNumB = (_testNumA + 1 + (rand() % 4)) % 5;
+      _testNumB = (_testNumA + 1 + (rand() % 3)) % 4;
       testNum = _testNumB;
 
       _movingDot[_dotPositionB] = ' ';
@@ -459,18 +459,15 @@ void APITest::RunTest(char thread) {
       ChangeCodec('A');
       break;
     case 1:
-      TestPlayout('B');
-      break;
-    case 2:
       if (!_randomTest) {
         fprintf(stdout, "\nTesting Delay ...\n");
       }
       TestDelay('A');
       break;
-    case 3:
+    case 2:
       TestSendVAD('A');
       break;
-    case 4:
+    case 3:
       TestRegisteration('A');
       break;
     default:
@@ -493,7 +490,6 @@ bool APITest::APIRunA() {
   } else {
     CurrentCodec('A');
     ChangeCodec('A');
-    TestPlayout('B');
     if (_codecCntrA == 0) {
       fprintf(stdout, "\nTesting Delay ...\n");
       TestDelay('A');
@@ -922,67 +918,6 @@ void APITest::TestRegisteration(char sendSide) {
   }
 }
 
-// Playout Mode, background noise mode.
-// Receiver Frequency, playout frequency.
-void APITest::TestPlayout(char receiveSide) {
-  AudioCodingModule* receiveACM;
-  AudioPlayoutMode* playoutMode = NULL;
-  switch (receiveSide) {
-    case 'A': {
-      receiveACM = _acmA.get();
-      playoutMode = &_playoutModeA;
-      break;
-    }
-    case 'B': {
-      receiveACM = _acmB.get();
-      playoutMode = &_playoutModeB;
-      break;
-    }
-    default:
-      receiveACM = _acmA.get();
-  }
-
-  int32_t receiveFreqHz = receiveACM->ReceiveFrequency();
-  int32_t playoutFreqHz = receiveACM->PlayoutFrequency();
-
-  CHECK_ERROR_MT(receiveFreqHz);
-  CHECK_ERROR_MT(playoutFreqHz);
-
-
-  char playoutString[25];
-  switch (*playoutMode) {
-    case voice: {
-      *playoutMode = fax;
-      strncpy(playoutString, "FAX", 25);
-      break;
-    }
-    case fax: {
-      *playoutMode = streaming;
-      strncpy(playoutString, "Streaming", 25);
-      break;
-    }
-    case streaming: {
-      *playoutMode = voice;
-      strncpy(playoutString, "Voice", 25);
-      break;
-    }
-    default:
-      *playoutMode = voice;
-      strncpy(playoutString, "Voice", 25);
-  }
-  CHECK_ERROR_MT(receiveACM->SetPlayoutMode(*playoutMode));
-  playoutString[24] = '\0';
-
-  if (!_randomTest) {
-    fprintf(stdout, "\n");
-    fprintf(stdout, "In Side %c\n", receiveSide);
-    fprintf(stdout, "---------------------------------\n");
-    fprintf(stdout, "Receive Frequency....... %d Hz\n", receiveFreqHz);
-    fprintf(stdout, "Playout Frequency....... %d Hz\n", playoutFreqHz);
-    fprintf(stdout, "Audio Playout Mode...... %s\n", playoutString);
-  }
-}
-
 void APITest::TestSendVAD(char side) {
   if (_randomTest) {
     return;

webrtc/modules/audio_coding/main/test/APITest.h

@@ -141,9 +141,6 @@ class APITest : public ACMTest {
   int32_t _minDelayB;
   bool _payloadUsed[32];
 
-  AudioPlayoutMode _playoutModeA;
-  AudioPlayoutMode _playoutModeB;
-
   bool _verbose;
 
   int _dotPositionA;

webrtc/modules/audio_coding/main/test/TwoWayCommunication.cc

@@ -32,10 +32,16 @@ namespace webrtc {
 
 TwoWayCommunication::TwoWayCommunication(int testMode)
     : _acmA(AudioCodingModule::Create(1)),
-      _acmB(AudioCodingModule::Create(2)),
       _acmRefA(AudioCodingModule::Create(3)),
-      _acmRefB(AudioCodingModule::Create(4)),
+      _testMode(testMode) {
-      _testMode(testMode) {}
+  AudioCodingModule::Config config;
+  // The clicks will be more obvious in FAX mode. TODO(henrik.lundin) Really?
+  config.neteq_config.playout_mode = kPlayoutFax;
+  config.id = 2;
+  _acmB.reset(AudioCodingModule::Create(config));
+  config.id = 4;
+  _acmRefB.reset(AudioCodingModule::Create(config));
+}
 
 TwoWayCommunication::~TwoWayCommunication() {
   delete _channel_A2B;
@@ -159,11 +165,6 @@ void TwoWayCommunication::SetUp() {
   _channelRef_B2A = new Channel;
   _acmRefB->RegisterTransportCallback(_channelRef_B2A);
   _channelRef_B2A->RegisterReceiverACM(_acmRefA.get());
-
-  // The clicks will be more obvious when we
-  // are in FAX mode.
-  EXPECT_EQ(_acmB->SetPlayoutMode(fax), 0);
-  EXPECT_EQ(_acmRefB->SetPlayoutMode(fax), 0);
 }
 
 void TwoWayCommunication::SetUpAutotest() {
@@ -233,11 +234,6 @@ void TwoWayCommunication::SetUpAutotest() {
   _channelRef_B2A = new Channel;
   _acmRefB->RegisterTransportCallback(_channelRef_B2A);
   _channelRef_B2A->RegisterReceiverACM(_acmRefA.get());
-
-  // The clicks will be more obvious when we
-  // are in FAX mode.
-  EXPECT_EQ(0, _acmB->SetPlayoutMode(fax));
-  EXPECT_EQ(0, _acmRefB->SetPlayoutMode(fax));
 }
 
 void TwoWayCommunication::Perform() {