Clean out platform specific things from voice_engine_defines.h.

BUG=webrtc:4690

Review-Url: https://codereview.webrtc.org/2689183002
Cr-Commit-Position: refs/heads/master@{#16578}

webrtc/audio/audio_send_stream.cc

@@ -35,7 +35,7 @@ namespace {
 constexpr char kOpusCodecName[] = "opus";
 
 bool IsCodec(const webrtc::CodecInst& codec, const char* ref_name) {
-  return (_stricmp(codec.plname, ref_name) == 0);
+  return (STR_CASE_CMP(codec.plname, ref_name) == 0);
 }
 }  // namespace
 

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

@@ -31,7 +31,7 @@ class CodecTest : public AfterStreamingFixture {
 };
 
 static void SetRateIfILBC(webrtc::CodecInst* codec_instance, int packet_size) {
-  if (!_stricmp(codec_instance->plname, "ilbc")) {
+  if (!STR_CASE_CMP(codec_instance->plname, "ilbc")) {
     if (packet_size == 160 || packet_size == 320) {
       codec_instance->rate = 15200;
     } else {
@@ -41,9 +41,9 @@ static void SetRateIfILBC(webrtc::CodecInst* codec_instance, int packet_size) {
 }
 
 static bool IsNotViableSendCodec(const char* codec_name) {
-  return !_stricmp(codec_name, "CN") ||
-         !_stricmp(codec_name, "telephone-event") ||
-         !_stricmp(codec_name, "red");
+  return !STR_CASE_CMP(codec_name, "CN") ||
+         !STR_CASE_CMP(codec_name, "telephone-event") ||
+         !STR_CASE_CMP(codec_name, "red");
 }
 
 TEST_F(CodecTest, PcmuIsDefaultCodecAndHasTheRightValues) {
@@ -138,7 +138,7 @@ TEST_F(CodecTest, VoiceActivityDetectionCanBeTurnedOff) {
 TEST_F(CodecTest, OpusMaxPlaybackRateCanBeSet) {
   for (int i = 0; i < voe_codec_->NumOfCodecs(); ++i) {
     voe_codec_->GetCodec(i, codec_instance_);
-    if (_stricmp("opus", codec_instance_.plname)) {
+    if (STR_CASE_CMP("opus", codec_instance_.plname)) {
       continue;
     }
     voe_codec_->SetSendCodec(channel_, codec_instance_);
@@ -154,7 +154,7 @@ TEST_F(CodecTest, OpusMaxPlaybackRateCanBeSet) {
 TEST_F(CodecTest, OpusDtxCanBeSetForOpus) {
   for (int i = 0; i < voe_codec_->NumOfCodecs(); ++i) {
     voe_codec_->GetCodec(i, codec_instance_);
-    if (_stricmp("opus", codec_instance_.plname)) {
+    if (STR_CASE_CMP("opus", codec_instance_.plname)) {
       continue;
     }
     voe_codec_->SetSendCodec(channel_, codec_instance_);
@@ -166,7 +166,7 @@ TEST_F(CodecTest, OpusDtxCanBeSetForOpus) {
 TEST_F(CodecTest, OpusDtxCannotBeSetForNonOpus) {
   for (int i = 0; i < voe_codec_->NumOfCodecs(); ++i) {
     voe_codec_->GetCodec(i, codec_instance_);
-    if (!_stricmp("opus", codec_instance_.plname)) {
+    if (!STR_CASE_CMP("opus", codec_instance_.plname)) {
       continue;
     }
     voe_codec_->SetSendCodec(channel_, codec_instance_);

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

@@ -44,7 +44,7 @@ TEST_F(DtmfTest, ManualCanChangeDtmfPayloadType) {
   // Start by modifying the receiving side.
   for (int i = 0; i < voe_codec_->NumOfCodecs(); i++) {
     EXPECT_EQ(0, voe_codec_->GetCodec(i, codec_instance));
-    if (!_stricmp("telephone-event", codec_instance.plname)) {
+    if (!STR_CASE_CMP("telephone-event", codec_instance.plname)) {
       codec_instance.pltype = 88;  // Use 88 instead of default 106.
       EXPECT_EQ(0, voe_base_->StopSend(channel_));
       EXPECT_EQ(0, voe_base_->StopPlayout(channel_));

webrtc/voice_engine/voice_engine_defines.h

@@ -10,7 +10,7 @@
 
 /*
  *  This file contains common constants for VoiceEngine, as well as
- *  platform specific settings and include files.
+ *  platform specific settings.
  */
 
 #ifndef WEBRTC_VOICE_ENGINE_VOICE_ENGINE_DEFINES_H
@@ -20,16 +20,8 @@
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
 #include "webrtc/typedefs.h"
 
-// ----------------------------------------------------------------------------
-//  Enumerators
-// ----------------------------------------------------------------------------
-
 namespace webrtc {
 
-// Internal buffer size required for mono audio, based on the highest sample
-// rate voice engine supports (10 ms of audio at 192 kHz).
-static const size_t kMaxMonoDataSizeSamples = 1920;
-
 // VolumeControl
 enum { kMinVolumeLevel = 0 };
 enum { kMaxVolumeLevel = 255 };
@@ -42,21 +34,8 @@ const float kMinOutputVolumePanning = 0.0f;
 // Max scale factor for output volume panning
 const float kMaxOutputVolumePanning = 1.0f;
 
-// DTMF
-enum { kMinDtmfEventCode = 0 };         // DTMF digit "0"
-enum { kMaxDtmfEventCode = 15 };        // DTMF digit "D"
-enum { kMinTelephoneEventCode = 0 };    // RFC4733 (Section 2.3.1)
-enum { kMaxTelephoneEventCode = 255 };  // RFC4733 (Section 2.3.1)
-enum { kMinTelephoneEventDuration = 100 };
-enum { kMaxTelephoneEventDuration = 60000 };       // Actual limit is 2^16
-enum { kMinTelephoneEventAttenuation = 0 };        // 0 dBm0
-enum { kMaxTelephoneEventAttenuation = 36 };       // -36 dBm0
-enum { kMinTelephoneEventSeparationMs = 100 };     // Min delta time between two
-                                                   // telephone events
 enum { kVoiceEngineMaxIpPacketSizeBytes = 1500 };  // assumes Ethernet
 
-enum { kVoiceEngineMaxModuleVersionSize = 960 };
-
 // Audio processing
 const NoiseSuppression::Level kDefaultNsMode = NoiseSuppression::kModerate;
 const GainControl::Mode kDefaultAgcMode =
@@ -73,48 +52,12 @@ const bool kDefaultAgcState =
 #endif
 const GainControl::Mode kDefaultRxAgcMode = GainControl::kAdaptiveDigital;
 
-// Codec
-// Min init target rate for iSAC-wb
-enum { kVoiceEngineMinIsacInitTargetRateBpsWb = 10000 };
-// Max init target rate for iSAC-wb
-enum { kVoiceEngineMaxIsacInitTargetRateBpsWb = 32000 };
-// Min init target rate for iSAC-swb
-enum { kVoiceEngineMinIsacInitTargetRateBpsSwb = 10000 };
-// Max init target rate for iSAC-swb
-enum { kVoiceEngineMaxIsacInitTargetRateBpsSwb = 56000 };
-// Lowest max rate for iSAC-wb
-enum { kVoiceEngineMinIsacMaxRateBpsWb = 32000 };
-// Highest max rate for iSAC-wb
-enum { kVoiceEngineMaxIsacMaxRateBpsWb = 53400 };
-// Lowest max rate for iSAC-swb
-enum { kVoiceEngineMinIsacMaxRateBpsSwb = 32000 };
-// Highest max rate for iSAC-swb
-enum { kVoiceEngineMaxIsacMaxRateBpsSwb = 107000 };
-// Lowest max payload size for iSAC-wb
-enum { kVoiceEngineMinIsacMaxPayloadSizeBytesWb = 120 };
-// Highest max payload size for iSAC-wb
-enum { kVoiceEngineMaxIsacMaxPayloadSizeBytesWb = 400 };
-// Lowest max payload size for iSAC-swb
-enum { kVoiceEngineMinIsacMaxPayloadSizeBytesSwb = 120 };
-// Highest max payload size for iSAC-swb
-enum { kVoiceEngineMaxIsacMaxPayloadSizeBytesSwb = 600 };
-
 // VideoSync
 // Lowest minimum playout delay
 enum { kVoiceEngineMinMinPlayoutDelayMs = 0 };
 // Highest minimum playout delay
 enum { kVoiceEngineMaxMinPlayoutDelayMs = 10000 };
 
-// Network
-// Min packet-timeout time for received RTP packets
-enum { kVoiceEngineMinPacketTimeoutSec = 1 };
-// Max packet-timeout time for received RTP packets
-enum { kVoiceEngineMaxPacketTimeoutSec = 150 };
-// Min sample time for dead-or-alive detection
-enum { kVoiceEngineMinSampleTimeSec = 1 };
-// Max sample time for dead-or-alive detection
-enum { kVoiceEngineMaxSampleTimeSec = 150 };
-
 // RTP/RTCP
 // Min 4-bit ID for RTP extension (see section 4.2 in RFC 5285)
 enum { kVoiceEngineMinRtpExtensionId = 1 };
@@ -123,37 +66,11 @@ enum { kVoiceEngineMaxRtpExtensionId = 14 };
 
 }  // namespace webrtc
 
-// ----------------------------------------------------------------------------
-//  Macros
-// ----------------------------------------------------------------------------
-
 #define NOT_SUPPORTED(stat)                 \
   LOG_F(LS_ERROR) << "not supported";       \
   stat.SetLastError(VE_FUNC_NOT_SUPPORTED); \
   return -1;
 
-#if (!defined(NDEBUG) && defined(_WIN32) && (_MSC_VER >= 1400))
-#include <windows.h>
-#include <stdio.h>
-#define DEBUG_PRINT(...)       \
-  {                            \
-    char msg[256];             \
-    sprintf(msg, __VA_ARGS__); \
-    OutputDebugStringA(msg);   \
-  }
-#else
-// special fix for visual 2003
-#define DEBUG_PRINT(exp) ((void)0)
-#endif  // !defined(NDEBUG) && defined(_WIN32)
-
-#define CHECK_CHANNEL(channel)     \
-  if (CheckChannel(channel) == -1) \
-    return -1;
-
-// ----------------------------------------------------------------------------
-//  Inline functions
-// ----------------------------------------------------------------------------
-
 namespace webrtc {
 
 inline int VoEId(int veId, int chId) {
@@ -175,143 +92,11 @@ inline int VoEChannelId(int moduleId) {
 
 }  // namespace webrtc
 
-// ----------------------------------------------------------------------------
-//  Platform settings
-// ----------------------------------------------------------------------------
-
-// *** WINDOWS ***
-
 #if defined(_WIN32)
-
-#include <windows.h>
-
-#pragma comment(lib, "winmm.lib")
-
-#ifndef WEBRTC_EXTERNAL_TRANSPORT
-#pragma comment(lib, "ws2_32.lib")
-#endif
-
-// ----------------------------------------------------------------------------
-//  Defines
-// ----------------------------------------------------------------------------
-
-// Default device for Windows PC
 #define WEBRTC_VOICE_ENGINE_DEFAULT_DEVICE \
   AudioDeviceModule::kDefaultCommunicationDevice
-
+#else
+#define WEBRTC_VOICE_ENGINE_DEFAULT_DEVICE 0
 #endif  // #if (defined(_WIN32)
 
-// *** LINUX ***
-
-#ifdef WEBRTC_LINUX
-
-#include <arpa/inet.h>
-#include <netinet/in.h>
-#include <pthread.h>
-#include <sys/socket.h>
-#include <sys/types.h>
-#ifndef QNX
-#include <linux/net.h>
-#ifndef ANDROID
-#include <sys/soundcard.h>
-#endif  // ANDROID
-#endif  // QNX
-#include <errno.h>
-#include <fcntl.h>
-#include <sched.h>
-#include <stdio.h>
-#include <stdlib.h>
-#include <string.h>
-#include <sys/ioctl.h>
-#include <sys/stat.h>
-#include <sys/time.h>
-#include <time.h>
-#include <unistd.h>
-
-#define DWORD unsigned long int
-#define WINAPI
-#define LPVOID void *
-#define FALSE 0
-#define TRUE 1
-#define UINT unsigned int
-#define UCHAR unsigned char
-#define TCHAR char
-#ifdef QNX
-#define _stricmp stricmp
-#else
-#define _stricmp strcasecmp
-#endif
-#define GetLastError() errno
-#define WSAGetLastError() errno
-#define LPCTSTR const char *
-#define LPCSTR const char *
-#define wsprintf sprintf
-#define TEXT(a) a
-#define _ftprintf fprintf
-#define _tcslen strlen
-#define FAR
-#define __cdecl
-#define LPSOCKADDR struct sockaddr *
-
-// Default device for Linux and Android
-#define WEBRTC_VOICE_ENGINE_DEFAULT_DEVICE 0
-
-#endif  // #ifdef WEBRTC_LINUX
-
-// *** WEBRTC_MAC ***
-// including iPhone
-
-#ifdef WEBRTC_MAC
-
-#include <AudioUnit/AudioUnit.h>
-#include <arpa/inet.h>
-#include <errno.h>
-#include <fcntl.h>
-#include <netinet/in.h>
-#include <pthread.h>
-#include <sched.h>
-#include <stdio.h>
-#include <stdlib.h>
-#include <string.h>
-#include <sys/socket.h>
-#include <sys/stat.h>
-#include <sys/time.h>
-#include <sys/types.h>
-#include <time.h>
-#include <unistd.h>
-#if !defined(WEBRTC_IOS)
-#include <CoreServices/CoreServices.h>
-#include <CoreAudio/CoreAudio.h>
-#include <AudioToolbox/DefaultAudioOutput.h>
-#include <AudioToolbox/AudioConverter.h>
-#include <CoreAudio/HostTime.h>
-#endif
-
-#define DWORD unsigned long int
-#define WINAPI
-#define LPVOID void *
-#define FALSE 0
-#define TRUE 1
-#define SOCKADDR_IN struct sockaddr_in
-#define UINT unsigned int
-#define UCHAR unsigned char
-#define TCHAR char
-#define _stricmp strcasecmp
-#define GetLastError() errno
-#define WSAGetLastError() errno
-#define LPCTSTR const char *
-#define wsprintf sprintf
-#define TEXT(a) a
-#define _ftprintf fprintf
-#define _tcslen strlen
-#define FAR
-#define __cdecl
-#define LPSOCKADDR struct sockaddr *
-#define LPCSTR const char *
-#define ULONG unsigned long
-
-// Default device for Mac and iPhone
-#define WEBRTC_VOICE_ENGINE_DEFAULT_DEVICE 0
-#endif  // #ifdef WEBRTC_MAC
-
 #endif  // WEBRTC_VOICE_ENGINE_VOICE_ENGINE_DEFINES_H