VoiceEngine(iOS & Android): removed NOT_SUPPORTED

Also:
- removed underflow of a uint32 creating crazy-large delay values
- removed always-fail AudioDeviceIPhone::MicrophoneIsAvailable() impl (see
  bug 3132)
- removed unnecessary exclusion of features from iOS & Android builds

BUG=2050,3132
R=andrew@webrtc.org, niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10909005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5820 4adac7df-926f-26a2-2b94-8c16560cd09d

webrtc/modules/audio_device/ios/audio_device_ios.cc

@@ -218,23 +218,13 @@ int32_t AudioDeviceIPhone::MicrophoneIsAvailable(bool& available) {
     WEBRTC_TRACE(kTraceModuleCall, kTraceAudioDevice, _id,
                  "%s", __FUNCTION__);
 
+    // This function has never been implemented correctly (boo!) but
+    // the returned |available| value is ignored (yay?) so for now
+    // just hard-code the previous (more buggily-implemented behavior)
+    // until the entire MicrophoneIsAvailable interface is ripped out
+    // (since it's apparently useless).
+    // https://code.google.com/p/webrtc/issues/detail?id=3132
     available = false;
-
-    OSStatus result = -1;
-    UInt32 channel = 0;
-    UInt32 size = sizeof(channel);
-    result = AudioSessionGetProperty(kAudioSessionProperty_AudioInputAvailable,
-                                     &size, &channel);
-    if (channel != 0) {
-        // API is not supported on this platform, we return available = true
-        WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice,
-                     _id, "  API call not supported on this version");
-        available = true;
-        return 0;
-    }
-
-    available = (channel == 0) ? false : true;
-
     return 0;
 }
 
@@ -1716,7 +1706,10 @@ void AudioDeviceIPhone::UpdatePlayoutDelay() {
     if (_playoutDelayMeasurementCounter >= 100) {
         // Update HW and OS delay every second, unlikely to change
 
-        _playoutDelay = 0;
+        // Since this is eventually rounded to integral ms, add 0.5ms
+        // here to get round-to-nearest-int behavior instead of
+        // truncation.
+        float totalDelaySeconds = 0.0005;
 
         // HW output latency
         Float32 f32(0);
@@ -1727,7 +1720,8 @@ void AudioDeviceIPhone::UpdatePlayoutDelay() {
             WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
                          "error HW latency (result=%d)", result);
         }
-        _playoutDelay += static_cast<int>(f32 * 1000000);
+        assert(f32 >= 0);
+        totalDelaySeconds += f32;
 
         // HW buffer duration
         f32 = 0;
@@ -1737,7 +1731,8 @@ void AudioDeviceIPhone::UpdatePlayoutDelay() {
             WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
                          "error HW buffer duration (result=%d)", result);
         }
-        _playoutDelay += static_cast<int>(f32 * 1000000);
+        assert(f32 >= 0);
+        totalDelaySeconds += f32;
 
         // AU latency
         Float64 f64(0);
@@ -1748,10 +1743,11 @@ void AudioDeviceIPhone::UpdatePlayoutDelay() {
             WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
                          "error AU latency (result=%d)", result);
         }
-        _playoutDelay += static_cast<int>(f64 * 1000000);
+        assert(f64 >= 0);
+        totalDelaySeconds += f64;
 
         // To ms
-        _playoutDelay = (_playoutDelay - 500) / 1000;
+        _playoutDelay = static_cast<uint32_t>(totalDelaySeconds / 1000);
 
         // Reset counter
         _playoutDelayMeasurementCounter = 0;
@@ -1766,7 +1762,10 @@ void AudioDeviceIPhone::UpdateRecordingDelay() {
     if (_recordingDelayMeasurementCounter >= 100) {
         // Update HW and OS delay every second, unlikely to change
 
-        _recordingDelayHWAndOS = 0;
+        // Since this is eventually rounded to integral ms, add 0.5ms
+        // here to get round-to-nearest-int behavior instead of
+        // truncation.
+        float totalDelaySeconds = 0.0005;
 
         // HW input latency
         Float32 f32(0);
@@ -1777,7 +1776,8 @@ void AudioDeviceIPhone::UpdateRecordingDelay() {
             WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
                          "error HW latency (result=%d)", result);
         }
-        _recordingDelayHWAndOS += static_cast<int>(f32 * 1000000);
+        assert(f32 >= 0);
+        totalDelaySeconds += f32;
 
         // HW buffer duration
         f32 = 0;
@@ -1787,7 +1787,8 @@ void AudioDeviceIPhone::UpdateRecordingDelay() {
             WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
                          "error HW buffer duration (result=%d)", result);
         }
-        _recordingDelayHWAndOS += static_cast<int>(f32 * 1000000);
+        assert(f32 >= 0);
+        totalDelaySeconds += f32;
 
         // AU latency
         Float64 f64(0);
@@ -1799,10 +1800,12 @@ void AudioDeviceIPhone::UpdateRecordingDelay() {
             WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
                          "error AU latency (result=%d)", result);
         }
-        _recordingDelayHWAndOS += static_cast<int>(f64 * 1000000);
+        assert(f64 >= 0);
+        totalDelaySeconds += f64;
 
         // To ms
-        _recordingDelayHWAndOS = (_recordingDelayHWAndOS - 500) / 1000;
+        _recordingDelayHWAndOS =
+            static_cast<uint32_t>(totalDelaySeconds / 1000);
 
         // Reset counter
         _recordingDelayMeasurementCounter = 0;

webrtc/voice_engine/test/auto_test/standard/dtmf_test.cc

@@ -52,7 +52,6 @@ TEST_F(DtmfTest, TestTwoNonDtmfEvents) {
   EXPECT_EQ(0, voe_dtmf_->SendTelephoneEvent(channel_, 110, true));
 }
 
-#ifndef WEBRTC_IOS
 TEST_F(DtmfTest, ManualCanDisableDtmfPlayoutExceptOnIphone) {
   TEST_LOG("Disabling DTMF playout (no tone should be heard) \n");
   EXPECT_EQ(0, voe_dtmf_->SetDtmfPlayoutStatus(channel_, false));
@@ -64,7 +63,6 @@ TEST_F(DtmfTest, ManualCanDisableDtmfPlayoutExceptOnIphone) {
   EXPECT_EQ(0, voe_dtmf_->SendTelephoneEvent(channel_, 0, true));
   Sleep(500);
 }
-#endif
 
 // This test modifies the DTMF payload type from the default 106 to 88
 // and then runs through 16 DTMF out.of-band events.

webrtc/voice_engine/test/auto_test/standard/video_sync_test.cc

@@ -123,9 +123,7 @@ TEST_F(VideoSyncTest,
   CheckEstimatesConvergeReasonablyWell(kMinimumReasonableDelayEstimateMs);
 }
 
-#if !defined(WEBRTC_ANDROID)
 TEST_F(VideoSyncTest, CanGetPlayoutBufferSize) {
   int ignored;
   EXPECT_EQ(0, voe_vsync_->GetPlayoutBufferSize(ignored));
 }
-#endif  // !ANDROID

webrtc/voice_engine/test/auto_test/standard/volume_test.cc

@@ -95,8 +95,6 @@ TEST_F(VolumeTest, ManualSetVolumeWorks) {
   Sleep(1000);
 }
 
-#if !defined(WEBRTC_IOS)
-
 TEST_F(VolumeTest, DISABLED_ON_LINUX(DefaultMicrophoneVolumeIsAtMost255)) {
   unsigned int volume = 1000;
   EXPECT_EQ(0, voe_volume_control_->GetMicVolume(volume));
@@ -144,10 +142,6 @@ TEST_F(VolumeTest, ManualCanSetChannelScaling) {
   Sleep(2000);
 }
 
-#endif  // !WEBRTC_IOS
-
-#if !defined(WEBRTC_ANDROID) && !defined(WEBRTC_IOS)
-
 TEST_F(VolumeTest, InputMutingIsNotEnabledByDefault) {
   bool is_muted = true;
   EXPECT_EQ(0, voe_volume_control_->GetInputMute(channel_, is_muted));
@@ -284,5 +278,3 @@ TEST_F(VolumeTest, ManualTestChannelPanning) {
   EXPECT_FLOAT_EQ(0.1f, left);
   EXPECT_FLOAT_EQ(0.8f, right);
 }
-
-#endif  // !WEBRTC_ANDROID && !WEBRTC_IOS

webrtc/voice_engine/test/cmd_test/voe_cmd_test.cc

@@ -341,8 +341,6 @@ void RunTest(std::string out_path) {
   // Call loop
   bool newcall = true;
   while (newcall) {
-
-#ifndef WEBRTC_ANDROID
     int rd(-1), pd(-1);
     res = hardware->GetNumOfRecordingDevices(rd);
     VALIDATE;
@@ -374,7 +372,6 @@ void RunTest(std::string out_path) {
     printf("Setting sound devices \n");
     res = hardware->SetRecordingDevice(rd);
     VALIDATE;
-#endif  // WEBRTC_ANDROID
 
     res = codec->SetVADStatus(0, enable_cng);
     VALIDATE;
@@ -415,7 +412,6 @@ void RunTest(std::string out_path) {
       VALIDATE;
     }
 
-#ifndef WEBRTC_ANDROID
     printf("Getting mic volume \n");
     unsigned int vol = 999;
     res = volume->GetMicVolume(vol);
@@ -423,7 +419,6 @@ void RunTest(std::string out_path) {
     if ((vol > 255) || (vol < 1)) {
       printf("\n****ERROR in GetMicVolume");
     }
-#endif
 
     int forever = 1;
     while (forever) {

webrtc/voice_engine/voe_call_report_impl.cc

@@ -56,8 +56,6 @@ int VoECallReportImpl::ResetCallReportStatistics(int channel)
 {
     WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
                  "ResetCallReportStatistics(channel=%d)", channel);
-    ANDROID_NOT_SUPPORTED(_shared->statistics());
-    IPHONE_NOT_SUPPORTED(_shared->statistics());
 
     if (!_shared->statistics().Initialized())
     {
@@ -114,8 +112,6 @@ int VoECallReportImpl::GetEchoMetricSummary(EchoStatistics& stats)
 {
     WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
                  "GetEchoMetricSummary()");
-    ANDROID_NOT_SUPPORTED(_shared->statistics());
-    IPHONE_NOT_SUPPORTED(_shared->statistics());
 
     if (!_shared->statistics().Initialized())
     {
@@ -215,8 +211,6 @@ int VoECallReportImpl::GetRoundTripTimeSummary(int channel, StatVal& delaysMs)
 {
     WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
                  "GetRoundTripTimeSummary()");
-    ANDROID_NOT_SUPPORTED(_shared->statistics());
-    IPHONE_NOT_SUPPORTED(_shared->statistics());
 
     if (!_shared->statistics().Initialized())
     {
@@ -241,8 +235,6 @@ int VoECallReportImpl::GetDeadOrAliveSummary(int channel,
 {
     WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
                  "GetDeadOrAliveSummary(channel=%d)", channel);
-    ANDROID_NOT_SUPPORTED(_shared->statistics());
-    IPHONE_NOT_SUPPORTED(_shared->statistics());
 
     if (!_shared->statistics().Initialized())
     {
@@ -283,8 +275,6 @@ int VoECallReportImpl::WriteReportToFile(const char* fileNameUTF8)
 {
     WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
                  "WriteReportToFile(fileNameUTF8=%s)", fileNameUTF8);
-    ANDROID_NOT_SUPPORTED(_shared->statistics());
-    IPHONE_NOT_SUPPORTED(_shared->statistics());
 
     if (!_shared->statistics().Initialized())
     {

webrtc/voice_engine/voe_codec_impl.cc

@@ -269,8 +269,6 @@ int VoECodecImpl::SetAMRWbEncFormat(int channel, AmrMode mode)
 {
     WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
                  "SetAMRWbEncFormat(channel=%d, mode=%d)", channel, mode);
-    ANDROID_NOT_SUPPORTED(_shared->statistics());
-    IPHONE_NOT_SUPPORTED(_shared->statistics());
 #ifdef WEBRTC_CODEC_AMRWB
     if (!_shared->statistics().Initialized())
     {
@@ -297,8 +295,6 @@ int VoECodecImpl::SetAMRWbDecFormat(int channel, AmrMode mode)
 {
     WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
                  "SetAMRWbDecFormat(channel=%i, mode=%i)", channel, mode);
-    ANDROID_NOT_SUPPORTED(_shared->statistics());
-    IPHONE_NOT_SUPPORTED(_shared->statistics());
 #ifdef WEBRTC_CODEC_AMRWB
     if (!_shared->statistics().Initialized())
     {
@@ -409,8 +405,6 @@ int VoECodecImpl::SetISACInitTargetRate(int channel, int rateBps,
     WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
                  "SetISACInitTargetRate(channel=%d, rateBps=%d, "
                  "useFixedFrameSize=%d)", channel, rateBps, useFixedFrameSize);
-    ANDROID_NOT_SUPPORTED(_shared->statistics());
-    IPHONE_NOT_SUPPORTED(_shared->statistics());
 #ifdef WEBRTC_CODEC_ISAC
     if (!_shared->statistics().Initialized())
     {
@@ -437,8 +431,6 @@ int VoECodecImpl::SetISACMaxRate(int channel, int rateBps)
 {
     WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
                  "SetISACMaxRate(channel=%d, rateBps=%d)", channel, rateBps);
-    ANDROID_NOT_SUPPORTED(_shared->statistics());
-    IPHONE_NOT_SUPPORTED(_shared->statistics());
 #ifdef WEBRTC_CODEC_ISAC
     if (!_shared->statistics().Initialized())
     {
@@ -466,8 +458,6 @@ int VoECodecImpl::SetISACMaxPayloadSize(int channel, int sizeBytes)
     WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
                  "SetISACMaxPayloadSize(channel=%d, sizeBytes=%d)", channel,
                  sizeBytes);
-    ANDROID_NOT_SUPPORTED(_shared->statistics());
-    IPHONE_NOT_SUPPORTED(_shared->statistics());
 #ifdef WEBRTC_CODEC_ISAC
     if (!_shared->statistics().Initialized())
     {

webrtc/voice_engine/voe_dtmf_impl.cc

@@ -307,7 +307,6 @@ int VoEDtmfImpl::SetDtmfPlayoutStatus(int channel, bool enable)
     WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
                  "SetDtmfPlayoutStatus(channel=%d, enable=%d)",
                  channel, enable);
-    IPHONE_NOT_SUPPORTED(_shared->statistics());
 
     if (!_shared->statistics().Initialized())
     {
@@ -329,7 +328,6 @@ int VoEDtmfImpl::GetDtmfPlayoutStatus(int channel, bool& enabled)
 {
     WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
                  "GetDtmfPlayoutStatus(channel=%d, enabled=?)", channel);
-    IPHONE_NOT_SUPPORTED(_shared->statistics());
     if (!_shared->statistics().Initialized())
     {
         _shared->SetLastError(VE_NOT_INITED, kTraceError);

webrtc/voice_engine/voe_hardware_impl.cc

@@ -148,8 +148,6 @@ int VoEHardwareImpl::GetNumOfRecordingDevices(int& devices)
 {
     WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
                  "GetNumOfRecordingDevices(devices=?)");
-    ANDROID_NOT_SUPPORTED(_shared->statistics());
-    IPHONE_NOT_SUPPORTED(_shared->statistics());
 
     if (!_shared->statistics().Initialized())
     {
@@ -169,8 +167,6 @@ int VoEHardwareImpl::GetNumOfPlayoutDevices(int& devices)
 {
     WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
                  "GetNumOfPlayoutDevices(devices=?)");
-    ANDROID_NOT_SUPPORTED(_shared->statistics());
-    IPHONE_NOT_SUPPORTED(_shared->statistics());
 
     if (!_shared->statistics().Initialized())
     {
@@ -193,8 +189,6 @@ int VoEHardwareImpl::GetRecordingDeviceName(int index,
 {
     WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
                  "GetRecordingDeviceName(index=%d)", index);
-    ANDROID_NOT_SUPPORTED(_shared->statistics());
-    IPHONE_NOT_SUPPORTED(_shared->statistics());
 
     if (!_shared->statistics().Initialized())
     {
@@ -251,8 +245,6 @@ int VoEHardwareImpl::GetPlayoutDeviceName(int index,
 {
     WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
                  "GetPlayoutDeviceName(index=%d)", index);
-    ANDROID_NOT_SUPPORTED(_shared->statistics());
-    IPHONE_NOT_SUPPORTED(_shared->statistics());
 
     if (!_shared->statistics().Initialized())
     {
@@ -310,8 +302,6 @@ int VoEHardwareImpl::SetRecordingDevice(int index,
                  "SetRecordingDevice(index=%d, recordingChannel=%d)",
                  index, (int) recordingChannel);
     CriticalSectionScoped cs(_shared->crit_sec());
-    IPHONE_NOT_SUPPORTED(_shared->statistics());
-    // TODO(leozwang): Add this api to Android OpenSL ES implementation.
 
     if (!_shared->statistics().Initialized())
     {
@@ -440,8 +430,6 @@ int VoEHardwareImpl::SetPlayoutDevice(int index)
     WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
                  "SetPlayoutDevice(index=%d)", index);
     CriticalSectionScoped cs(_shared->crit_sec());
-    ANDROID_NOT_SUPPORTED(_shared->statistics());
-    IPHONE_NOT_SUPPORTED(_shared->statistics());
 
     if (!_shared->statistics().Initialized())
     {
@@ -544,8 +532,6 @@ int VoEHardwareImpl::GetRecordingDeviceStatus(bool& isAvailable)
 {
     WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
                  "GetRecordingDeviceStatus()");
-    ANDROID_NOT_SUPPORTED(_shared->statistics());
-    IPHONE_NOT_SUPPORTED(_shared->statistics());
 
     if (!_shared->statistics().Initialized())
     {
@@ -578,8 +564,6 @@ int VoEHardwareImpl::GetPlayoutDeviceStatus(bool& isAvailable)
 {
     WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
                  "GetPlayoutDeviceStatus()");
-    ANDROID_NOT_SUPPORTED(_shared->statistics());
-    IPHONE_NOT_SUPPORTED(_shared->statistics());
 
     if (!_shared->statistics().Initialized())
     {
@@ -612,7 +596,6 @@ int VoEHardwareImpl::ResetAudioDevice()
 {
     WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
                  "ResetAudioDevice()");
-    ANDROID_NOT_SUPPORTED(_shared->statistics());
 
     if (!_shared->statistics().Initialized())
     {
@@ -640,8 +623,6 @@ int VoEHardwareImpl::AudioDeviceControl(unsigned int par1, unsigned int par2,
 {
     WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
                  "AudioDeviceControl(%i, %i, %i)", par1, par2, par3);
-    ANDROID_NOT_SUPPORTED(_shared->statistics());
-    IPHONE_NOT_SUPPORTED(_shared->statistics());
     if (!_shared->statistics().Initialized())
     {
         _shared->SetLastError(VE_NOT_INITED, kTraceError);
@@ -656,7 +637,6 @@ int VoEHardwareImpl::SetLoudspeakerStatus(bool enable)
 {
     WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
                  "SetLoudspeakerStatus(enable=%i)", (int) enable);
-    IPHONE_NOT_SUPPORTED(_shared->statistics());
 
     if (!_shared->statistics().Initialized())
     {
@@ -683,7 +663,6 @@ int VoEHardwareImpl::GetLoudspeakerStatus(bool& enabled)
 {
     WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
                  "GetLoudspeakerStatus()");
-    IPHONE_NOT_SUPPORTED(_shared->statistics());
 
 #if defined(WEBRTC_ANDROID)
     if (!_shared->statistics().Initialized())
@@ -711,8 +690,6 @@ int VoEHardwareImpl::GetCPULoad(int& loadPercent)
 {
     WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
                  "GetCPULoad()");
-    ANDROID_NOT_SUPPORTED(_shared->statistics());
-    IPHONE_NOT_SUPPORTED(_shared->statistics());
 
     if (!_shared->statistics().Initialized())
     {

webrtc/voice_engine/voe_neteq_stats_impl.cc

@@ -53,7 +53,6 @@ int VoENetEqStatsImpl::GetNetworkStatistics(int channel,
 {
     WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
                  "GetNetworkStatistics(channel=%d, stats=?)", channel);
-    ANDROID_NOT_SUPPORTED(_shared->statistics());
 
     if (!_shared->statistics().Initialized())
     {
@@ -74,7 +73,6 @@ int VoENetEqStatsImpl::GetNetworkStatistics(int channel,
 
 int VoENetEqStatsImpl::GetDecodingCallStatistics(
     int channel, AudioDecodingCallStats* stats) const {
-  ANDROID_NOT_SUPPORTED(_shared->statistics());
 
   if (!_shared->statistics().Initialized()) {
     _shared->SetLastError(VE_NOT_INITED, kTraceError);

webrtc/voice_engine/voe_video_sync_impl.cc

@@ -51,7 +51,6 @@ int VoEVideoSyncImpl::GetPlayoutTimestamp(int channel, unsigned int& timestamp)
 {
     WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
                  "GetPlayoutTimestamp(channel=%d, timestamp=?)", channel);
-    IPHONE_NOT_SUPPORTED(_shared->statistics());
 
     if (!_shared->statistics().Initialized())
     {
@@ -75,7 +74,6 @@ int VoEVideoSyncImpl::SetInitTimestamp(int channel,
     WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
                  "SetInitTimestamp(channel=%d, timestamp=%lu)",
                  channel, timestamp);
-    IPHONE_NOT_SUPPORTED(_shared->statistics());
 
     if (!_shared->statistics().Initialized())
     {
@@ -99,7 +97,6 @@ int VoEVideoSyncImpl::SetInitSequenceNumber(int channel,
     WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
                  "SetInitSequenceNumber(channel=%d, sequenceNumber=%hd)",
                  channel, sequenceNumber);
-    IPHONE_NOT_SUPPORTED(_shared->statistics());
 
     if (!_shared->statistics().Initialized())
     {
@@ -122,7 +119,6 @@ int VoEVideoSyncImpl::SetMinimumPlayoutDelay(int channel,int delayMs)
     WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
                  "SetMinimumPlayoutDelay(channel=%d, delayMs=%d)",
                  channel, delayMs);
-    IPHONE_NOT_SUPPORTED(_shared->statistics());
 
     if (!_shared->statistics().Initialized())
     {
@@ -145,7 +141,6 @@ int VoEVideoSyncImpl::SetInitialPlayoutDelay(int channel, int delay_ms)
     WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
                  "SetInitialPlayoutDelay(channel=%d, delay_ms=%d)",
                  channel, delay_ms);
-    IPHONE_NOT_SUPPORTED(_shared->statistics());
 
     if (!_shared->statistics().Initialized())
     {
@@ -168,7 +163,6 @@ int VoEVideoSyncImpl::GetDelayEstimate(int channel,
                                        int* playout_buffer_delay_ms) {
   WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
                "GetDelayEstimate(channel=%d, delayMs=?)", channel);
-  IPHONE_NOT_SUPPORTED(_shared->statistics());
 
   if (!_shared->statistics().Initialized()) {
     _shared->SetLastError(VE_NOT_INITED, kTraceError);
@@ -192,7 +186,6 @@ int VoEVideoSyncImpl::GetPlayoutBufferSize(int& bufferMs)
 {
     WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
                "GetPlayoutBufferSize(bufferMs=?)");
-    IPHONE_NOT_SUPPORTED(_shared->statistics());
 
     if (!_shared->statistics().Initialized())
     {
@@ -240,7 +233,6 @@ int VoEVideoSyncImpl::GetRtpRtcp(int channel, RtpRtcp** rtpRtcpModule,
 int VoEVideoSyncImpl::GetLeastRequiredDelayMs(int channel) const {
   WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
                "GetLeastRequiredDelayMS(channel=%d)", channel);
-  IPHONE_NOT_SUPPORTED(_shared->statistics());
 
   if (!_shared->statistics().Initialized()) {
     _shared->SetLastError(VE_NOT_INITED, kTraceError);

webrtc/voice_engine/voe_volume_control_impl.cc

@@ -54,7 +54,6 @@ int VoEVolumeControlImpl::SetSpeakerVolume(unsigned int volume)
 {
     WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
                "SetSpeakerVolume(volume=%u)", volume);
-    IPHONE_NOT_SUPPORTED(_shared->statistics());
 
     if (!_shared->statistics().Initialized())
     {
@@ -96,7 +95,6 @@ int VoEVolumeControlImpl::GetSpeakerVolume(unsigned int& volume)
 {
     WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
                "GetSpeakerVolume()");
-    IPHONE_NOT_SUPPORTED(_shared->statistics());
 
     if (!_shared->statistics().Initialized())
     {
@@ -179,8 +177,6 @@ int VoEVolumeControlImpl::SetMicVolume(unsigned int volume)
 {
     WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
                "SetMicVolume(volume=%u)", volume);
-    ANDROID_NOT_SUPPORTED(_shared->statistics());
-    IPHONE_NOT_SUPPORTED(_shared->statistics());
 
     if (!_shared->statistics().Initialized())
     {
@@ -238,8 +234,6 @@ int VoEVolumeControlImpl::GetMicVolume(unsigned int& volume)
 {
     WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
                "GetMicVolume()");
-    ANDROID_NOT_SUPPORTED(_shared->statistics());
-    IPHONE_NOT_SUPPORTED(_shared->statistics());
 
     if (!_shared->statistics().Initialized())
     {
@@ -538,8 +532,6 @@ int VoEVolumeControlImpl::SetOutputVolumePan(int channel,
     WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
                "SetOutputVolumePan(channel=%d, left=%2.1f, right=%2.1f)",
                channel, left, right);
-    ANDROID_NOT_SUPPORTED(_shared->statistics());
-    IPHONE_NOT_SUPPORTED(_shared->statistics());
 
     if (!_shared->statistics().Initialized())
     {
@@ -588,8 +580,6 @@ int VoEVolumeControlImpl::GetOutputVolumePan(int channel,
 {
     WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
                "GetOutputVolumePan(channel=%d, left=?, right=?)", channel);
-    ANDROID_NOT_SUPPORTED(_shared->statistics());
-    IPHONE_NOT_SUPPORTED(_shared->statistics());
 
     if (!_shared->statistics().Initialized())
     {

webrtc/voice_engine/voice_engine_defines.h

@@ -279,30 +279,6 @@ inline int VoEChannelId(int moduleId)
 // Default device for Linux and Android
 #define WEBRTC_VOICE_ENGINE_DEFAULT_DEVICE 0
 
-#ifdef ANDROID
-
-// ----------------------------------------------------------------------------
-//  Defines
-// ----------------------------------------------------------------------------
-
-  // Always excluded for Android builds
-  #undef WEBRTC_CODEC_ISAC
-  #undef WEBRTC_VOE_EXTERNAL_REC_AND_PLAYOUT
-
-  #define ANDROID_NOT_SUPPORTED(stat) NOT_SUPPORTED(stat)
-
-#else // LINUX PC
-
-// ----------------------------------------------------------------------------
-//  Defines
-// ----------------------------------------------------------------------------
-
-  #define ANDROID_NOT_SUPPORTED(stat)
-
-#endif // ANDROID - LINUX PC
-
-#else
-#define ANDROID_NOT_SUPPORTED(stat)
 #endif  // #ifdef WEBRTC_LINUX
 
 // *** WEBRTC_MAC ***
@@ -359,35 +335,6 @@ inline int VoEChannelId(int moduleId)
 
 // Default device for Mac and iPhone
 #define WEBRTC_VOICE_ENGINE_DEFAULT_DEVICE 0
-
-// iPhone specific
-#if defined(WEBRTC_IOS)
-
-// ----------------------------------------------------------------------------
-//  Defines
-// ----------------------------------------------------------------------------
-
-  // Always excluded for iPhone builds
-  #undef WEBRTC_CODEC_ISAC
-  #undef WEBRTC_VOE_EXTERNAL_REC_AND_PLAYOUT
-
-  #define IPHONE_NOT_SUPPORTED(stat) NOT_SUPPORTED(stat)
-
-#else // Non-iPhone
-
-// ----------------------------------------------------------------------------
-//  Enumerators
-// ----------------------------------------------------------------------------
-
-// ----------------------------------------------------------------------------
-//  Defines
-// ----------------------------------------------------------------------------
-
-  #define IPHONE_NOT_SUPPORTED(stat)
-#endif
-
-#else
-#define IPHONE_NOT_SUPPORTED(stat)
 #endif  // #ifdef WEBRTC_MAC
 
 #endif // WEBRTC_VOICE_ENGINE_VOICE_ENGINE_DEFINES_H