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Remove USE_WEBRTC_DEV_BRANCH.

Browse Source 
talk/ and webrtc/ are hosted in the same repository and it no longer
makes sense to support building talk/ without the corresponding webrtc/
catalog.

R=bjornv@webrtc.org, juberti@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/39849004

Cr-Commit-Position: refs/heads/master@{#8291}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8291 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
pbos@webrtc.org
2015-02-09 10:16:33 +00:00
parent 2b69eab077
commit 8cf9bdb3fa
14 changed files with 0 additions and 157 deletions
Download Patch File Download Diff File Expand all files Collapse all files

8
talk/app/webrtc/test/fakeaudiocapturemodule.cc
View File

@@ -704,20 +704,12 @@ void FakeAudioCaptureModule::ReceiveFrameP() {
}
ResetRecBuffer();
uint32_t nSamplesOut = 0;
#ifdef USE_WEBRTC_DEV_BRANCH
int64_t elapsed_time_ms = 0;
#else
uint32_t rtp_timestamp = 0;
#endif
int64_t ntp_time_ms = 0;
if (audio_callback_->NeedMorePlayData(kNumberSamples, kNumberBytesPerSample,
kNumberOfChannels, kSamplesPerSecond,
rec_buffer_, nSamplesOut,
#ifdef USE_WEBRTC_DEV_BRANCH
&elapsed_time_ms, &ntp_time_ms) != 0) {
#else
&rtp_timestamp, &ntp_time_ms) != 0) {
#endif
ASSERT(false);
}
ASSERT(nSamplesOut == kNumberSamples);

8
talk/app/webrtc/test/fakeaudiocapturemodule_unittest.cc
View File

@@ -85,11 +85,7 @@ class FakeAdmTest : public testing::Test,
const uint32_t samplesPerSec,
void* audioSamples,
uint32_t& nSamplesOut,
#ifdef USE_WEBRTC_DEV_BRANCH
int64_t* elapsed_time_ms,
#else
uint32_t* rtp_timestamp,
#endif
int64_t* ntp_time_ms) {
++pull_iterations_;
const uint32_t audio_buffer_size = nSamples * nBytesPerSample;
@@ -97,11 +93,7 @@ class FakeAdmTest : public testing::Test,
CopyFromRecBuffer(audioSamples, audio_buffer_size):
GenerateZeroBuffer(audioSamples, audio_buffer_size);
nSamplesOut = bytes_out / nBytesPerSample;
#ifdef USE_WEBRTC_DEV_BRANCH
*elapsed_time_ms = 0;
#else
*rtp_timestamp = 0;
#endif
*ntp_time_ms = 0;
return 0;
}

1
talk/build/common.gypi
View File

@@ -67,7 +67,6 @@
'HAVE_SRTP',
'HAVE_WEBRTC_VIDEO',
'HAVE_WEBRTC_VOICE',
'USE_WEBRTC_DEV_BRANCH',
],
'conditions': [
# TODO(ronghuawu): Support dynamic library build.

2
talk/media/webrtc/fakewebrtccommon.h
View File

@@ -41,10 +41,8 @@ namespace cricket {
#define WEBRTC_BOOL_STUB(method, args) \
virtual bool method args OVERRIDE { return true; }
#ifdef USE_WEBRTC_DEV_BRANCH
#define WEBRTC_BOOL_STUB_CONST(method, args) \
virtual bool method args const OVERRIDE { return true; }
#endif
#define WEBRTC_VOID_STUB(method, args) \
virtual void method args OVERRIDE {}

16
talk/media/webrtc/fakewebrtcvoiceengine.h
View File

@@ -40,9 +40,7 @@
#include "webrtc/base/basictypes.h"
#include "webrtc/base/gunit.h"
#include "webrtc/base/stringutils.h"
#ifdef USE_WEBRTC_DEV_BRANCH
#include "webrtc/modules/audio_processing/include/audio_processing.h"
#endif
#include "webrtc/video_engine/include/vie_network.h"
namespace cricket {
@@ -76,7 +74,6 @@ static const int kOpusBandwidthFb = 20000;
} \
} while (0);
#ifdef USE_WEBRTC_DEV_BRANCH
class FakeAudioProcessing : public webrtc::AudioProcessing {
public:
FakeAudioProcessing() : experimental_ns_enabled_(false) {}
@@ -156,7 +153,6 @@ class FakeAudioProcessing : public webrtc::AudioProcessing {
private:
bool experimental_ns_enabled_;
};
#endif
class FakeWebRtcVoiceEngine
: public webrtc::VoEAudioProcessing,
@@ -442,11 +438,7 @@ class FakeWebRtcVoiceEngine
return 0;
}
virtual webrtc::AudioProcessing* audio_processing() OVERRIDE {
#ifdef USE_WEBRTC_DEV_BRANCH
return &audio_processing_;
#else
return NULL;
#endif
}
WEBRTC_FUNC(CreateChannel, ()) {
return AddChannel();
@@ -628,7 +620,6 @@ class FakeWebRtcVoiceEngine
WEBRTC_STUB(GetVADStatus, (int channel, bool& enabled,
webrtc::VadModes& mode, bool& disabledDTX));
#ifdef USE_WEBRTC_DEV_BRANCH
WEBRTC_FUNC(SetFECStatus, (int channel, bool enable)) {
WEBRTC_CHECK_CHANNEL(channel);
if (_stricmp(channels_[channel]->send_codec.plname, "opus") != 0) {
@@ -663,7 +654,6 @@ class FakeWebRtcVoiceEngine
channels_[channel]->max_encoding_bandwidth = kOpusBandwidthFb;
return 0;
}
#endif // USE_WEBRTC_DEV_BRANCH
// webrtc::VoEDtmf
WEBRTC_FUNC(SendTelephoneEvent, (int channel, int event_code,
@@ -970,11 +960,9 @@ class FakeWebRtcVoiceEngine
stats.packetsReceived = kIntStatValue;
return 0;
}
#ifdef USE_WEBRTC_DEV_BRANCH
WEBRTC_FUNC(SetREDStatus, (int channel, bool enable, int redPayloadtype)) {
return SetFECStatus(channel, enable, redPayloadtype);
}
#endif
// TODO(minyue): remove the below function when transition to SetREDStatus
// is finished.
WEBRTC_FUNC(SetFECStatus, (int channel, bool enable, int redPayloadtype)) {
@@ -983,11 +971,9 @@ class FakeWebRtcVoiceEngine
channels_[channel]->red_type = redPayloadtype;
return 0;
}
#ifdef USE_WEBRTC_DEV_BRANCH
WEBRTC_FUNC(GetREDStatus, (int channel, bool& enable, int& redPayloadtype)) {
return GetFECStatus(channel, enable, redPayloadtype);
}
#endif
// TODO(minyue): remove the below function when transition to GetREDStatus
// is finished.
WEBRTC_FUNC(GetFECStatus, (int channel, bool& enable, int& redPayloadtype)) {
@@ -1310,9 +1296,7 @@ class FakeWebRtcVoiceEngine
int playout_sample_rate_;
DtmfInfo dtmf_info_;
webrtc::VoEMediaProcess* media_processor_;
#ifdef USE_WEBRTC_DEV_BRANCH
FakeAudioProcessing audio_processing_;
#endif
};
#undef WEBRTC_CHECK_HEADER_EXTENSION_ID

18
talk/media/webrtc/webrtcvideoengine_unittest.cc
View File

@@ -1213,11 +1213,9 @@ TEST_F(WebRtcVideoEngineTestFake, SetCpuOveruseOptionsWithEncodeUsageMethod) {
EXPECT_EQ(20, cpu_option.high_encode_usage_threshold_percent);
EXPECT_FALSE(cpu_option.enable_capture_jitter_method);
EXPECT_TRUE(cpu_option.enable_encode_usage_method);
#ifdef USE_WEBRTC_DEV_BRANCH
// Verify that optional encode rsd thresholds are not set.
EXPECT_EQ(-1, cpu_option.low_encode_time_rsd_threshold);
EXPECT_EQ(-1, cpu_option.high_encode_time_rsd_threshold);
#endif
// Add a new send stream and verify that cpu options are set from start.
EXPECT_TRUE(channel_->AddSendStream(cricket::StreamParams::CreateLegacy(3)));
@@ -1228,11 +1226,9 @@ TEST_F(WebRtcVideoEngineTestFake, SetCpuOveruseOptionsWithEncodeUsageMethod) {
EXPECT_EQ(20, cpu_option.high_encode_usage_threshold_percent);
EXPECT_FALSE(cpu_option.enable_capture_jitter_method);
EXPECT_TRUE(cpu_option.enable_encode_usage_method);
#ifdef USE_WEBRTC_DEV_BRANCH
// Verify that optional encode rsd thresholds are not set.
EXPECT_EQ(-1, cpu_option.low_encode_time_rsd_threshold);
EXPECT_EQ(-1, cpu_option.high_encode_time_rsd_threshold);
#endif
}
TEST_F(WebRtcVideoEngineTestFake, SetCpuOveruseOptionsWithEncodeRsdThresholds) {
@@ -1255,10 +1251,8 @@ TEST_F(WebRtcVideoEngineTestFake, SetCpuOveruseOptionsWithEncodeRsdThresholds) {
EXPECT_EQ(20, cpu_option.high_encode_usage_threshold_percent);
EXPECT_FALSE(cpu_option.enable_capture_jitter_method);
EXPECT_TRUE(cpu_option.enable_encode_usage_method);
#ifdef USE_WEBRTC_DEV_BRANCH
EXPECT_EQ(30, cpu_option.low_encode_time_rsd_threshold);
EXPECT_EQ(40, cpu_option.high_encode_time_rsd_threshold);
#endif
// Add a new send stream and verify that cpu options are set from start.
EXPECT_TRUE(channel_->AddSendStream(cricket::StreamParams::CreateLegacy(3)));
@@ -1269,10 +1263,8 @@ TEST_F(WebRtcVideoEngineTestFake, SetCpuOveruseOptionsWithEncodeRsdThresholds) {
EXPECT_EQ(20, cpu_option.high_encode_usage_threshold_percent);
EXPECT_FALSE(cpu_option.enable_capture_jitter_method);
EXPECT_TRUE(cpu_option.enable_encode_usage_method);
#ifdef USE_WEBRTC_DEV_BRANCH
EXPECT_EQ(30, cpu_option.low_encode_time_rsd_threshold);
EXPECT_EQ(40, cpu_option.high_encode_time_rsd_threshold);
#endif
}
// Test that AddRecvStream doesn't create new channel for 1:1 call.
@@ -2105,7 +2097,6 @@ TEST_F(WebRtcVideoEngineTestFake, ExternalCodecIgnored) {
EXPECT_NE("VP8", codecs[codecs.size() - 1].name);
}
#ifdef USE_WEBRTC_DEV_BRANCH
TEST_F(WebRtcVideoEngineTestFake, SetSendCodecsWithExternalH264) {
encoder_factory_.AddSupportedVideoCodecType(webrtc::kVideoCodecH264, "H264");
engine_.SetExternalEncoderFactory(&encoder_factory_);
@@ -2241,7 +2232,6 @@ TEST_F(WebRtcVideoEngineTestFake, SetRecvCodecsWithVP8AndExternalH264) {
// The RTX payload type should have been set.
EXPECT_EQ(rtx_codec.id, vie_.GetRtxRecvPayloadType(channel_num));
}
#endif
// Tests that OnReadyToSend will be propagated into ViE.
TEST_F(WebRtcVideoEngineTestFake, OnReadyToSend) {
@@ -2445,19 +2435,11 @@ TEST_F(WebRtcVideoMediaChannelTest, DISABLED_SendVp8HdAndReceiveAdaptedVp8Vga) {
EXPECT_FRAME_WAIT(1, codec.width, codec.height, kTimeout);
}
#ifdef USE_WEBRTC_DEV_BRANCH
TEST_F(WebRtcVideoMediaChannelTest, GetStats) {
#else
TEST_F(WebRtcVideoMediaChannelTest, DISABLED_GetStats) {
#endif
Base::GetStats();
}
#ifdef USE_WEBRTC_DEV_BRANCH
TEST_F(WebRtcVideoMediaChannelTest, GetStatsMultipleRecvStreams) {
#else
TEST_F(WebRtcVideoMediaChannelTest, DISABLED_GetStatsMultipleRecvStreams) {
#endif
Base::GetStatsMultipleRecvStreams();
}

38
talk/media/webrtc/webrtcvoiceengine.cc
View File

@@ -956,7 +956,6 @@ bool WebRtcVoiceEngine::ApplyOptions(const AudioOptions& options_in) {
new webrtc::DelayCorrection(experimental_aec));
}
#ifdef USE_WEBRTC_DEV_BRANCH
experimental_ns_.SetFrom(options.experimental_ns);
bool experimental_ns;
if (experimental_ns_.Get(&experimental_ns)) {
@@ -964,7 +963,6 @@ bool WebRtcVoiceEngine::ApplyOptions(const AudioOptions& options_in) {
config.Set<webrtc::ExperimentalNs>(
new webrtc::ExperimentalNs(experimental_ns));
}
#endif
// We check audioproc for the benefit of tests, since FakeWebRtcVoiceEngine
// returns NULL on audio_processing().
@@ -973,24 +971,6 @@ bool WebRtcVoiceEngine::ApplyOptions(const AudioOptions& options_in) {
audioproc->SetExtraOptions(config);
}
#ifndef USE_WEBRTC_DEV_BRANCH
bool experimental_ns;
if (options.experimental_ns.Get(&experimental_ns)) {
LOG(LS_INFO) << "Experimental ns is enabled? " << experimental_ns;
// We check audioproc for the benefit of tests, since FakeWebRtcVoiceEngine
// returns NULL on audio_processing().
if (audioproc) {
if (audioproc->EnableExperimentalNs(experimental_ns) == -1) {
LOG_RTCERR1(EnableExperimentalNs, experimental_ns);
return false;
}
} else {
LOG(LS_VERBOSE) << "Experimental noise suppression set to "
<< experimental_ns;
}
}
#endif
uint32 recording_sample_rate;
if (options.recording_sample_rate.Get(&recording_sample_rate)) {
LOG(LS_INFO) << "Recording sample rate is " << recording_sample_rate;
@@ -2073,13 +2053,8 @@ bool WebRtcVoiceMediaChannel::SetSendCodecs(
// Disable VAD, FEC, and RED unless we know the other side wants them.
engine()->voe()->codec()->SetVADStatus(channel, false);
engine()->voe()->rtp()->SetNACKStatus(channel, false, 0);
#ifdef USE_WEBRTC_DEV_BRANCH
engine()->voe()->rtp()->SetREDStatus(channel, false);
engine()->voe()->codec()->SetFECStatus(channel, false);
#else
// TODO(minyue): Remove code under #else case after new WebRTC roll.
engine()->voe()->rtp()->SetFECStatus(channel, false);
#endif // USE_WEBRTC_DEV_BRANCH
// Scan through the list to figure out the codec to use for sending, along
// with the proper configuration for VAD and DTMF.
@@ -2121,16 +2096,9 @@ bool WebRtcVoiceMediaChannel::SetSendCodecs(
// Enable redundant encoding of the specified codec. Treat any
// failure as a fatal internal error.
#ifdef USE_WEBRTC_DEV_BRANCH
LOG(LS_INFO) << "Enabling RED on channel " << channel;
if (engine()->voe()->rtp()->SetREDStatus(channel, true, it->id) == -1) {
LOG_RTCERR3(SetREDStatus, channel, true, it->id);
#else
// TODO(minyue): Remove code under #else case after new WebRTC roll.
LOG(LS_INFO) << "Enabling FEC";
if (engine()->voe()->rtp()->SetFECStatus(channel, true, it->id) == -1) {
LOG_RTCERR3(SetFECStatus, channel, true, it->id);
#endif // USE_WEBRTC_DEV_BRANCH
return false;
}
} else {
@@ -2166,13 +2134,11 @@ bool WebRtcVoiceMediaChannel::SetSendCodecs(
if (enable_codec_fec) {
LOG(LS_INFO) << "Attempt to enable codec internal FEC on channel "
<< channel;
#ifdef USE_WEBRTC_DEV_BRANCH
if (engine()->voe()->codec()->SetFECStatus(channel, true) == -1) {
// Enable codec internal FEC. Treat any failure as fatal internal error.
LOG_RTCERR2(SetFECStatus, channel, true);
return false;
}
#endif // USE_WEBRTC_DEV_BRANCH
}
// maxplaybackrate should be set after SetSendCodec.
@@ -2183,12 +2149,10 @@ bool WebRtcVoiceMediaChannel::SetSendCodecs(
<< opus_max_playback_rate
<< " Hz on channel "
<< channel;
#ifdef USE_WEBRTC_DEV_BRANCH
if (engine()->voe()->codec()->SetOpusMaxPlaybackRate(
channel, opus_max_playback_rate) == -1) {
LOG(LS_WARNING) << "Could not set maximum playback rate.";
}
#endif
}
// Always update the |send_codec_| to the currently set send codec.
@@ -3432,9 +3396,7 @@ bool WebRtcVoiceMediaChannel::GetStats(VoiceMediaInfo* info) {
rinfo.fraction_lost = static_cast<float>(cs.fractionLost) / (1 << 8);
rinfo.packets_lost = cs.cumulativeLost;
rinfo.ext_seqnum = cs.extendedMax;
#ifdef USE_WEBRTC_DEV_BRANCH
rinfo.capture_start_ntp_time_ms = cs.capture_start_ntp_time_ms_;
#endif
if (codec.pltype != -1) {
rinfo.codec_name = codec.plname;
}

2
talk/media/webrtc/webrtcvoiceengine_unittest.cc
View File

@@ -1159,7 +1159,6 @@ TEST_F(WebRtcVoiceEngineTestFake, AddRecvStreamEnableNack) {
EXPECT_TRUE(voe_.GetNACK(channel_num));
}
#ifdef USE_WEBRTC_DEV_BRANCH
// Test that without useinbandfec, Opus FEC is off.
TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecNoOpusFec) {
EXPECT_TRUE(SetupEngine());
@@ -1410,7 +1409,6 @@ TEST_F(WebRtcVoiceEngineTestFake, SetOpusMaxPlaybackRateOnTwoStreams) {
EXPECT_EQ(cricket::kOpusBandwidthNb,
voe_.GetMaxEncodingBandwidth(channel_num));
}
#endif // USE_WEBRTC_DEV_BRANCH
// Test that we can apply CELT with stereo mode but fail with mono mode.
TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsCelt) {

2
webrtc/base/BUILD.gn
View File

@@ -19,7 +19,6 @@ config("rtc_base_config") {
defines = [
"FEATURE_ENABLE_SSL",
"LOGGING=1",
"USE_WEBRTC_DEV_BRANCH",
]
# TODO(henrike): issue 3307, make rtc_base build without disabling
@@ -149,7 +148,6 @@ static_library("rtc_base") {
defines = [
"LOGGING=1",
"USE_WEBRTC_DEV_BRANCH",
]
sources = [

1
webrtc/base/base.gyp
View File

@@ -62,7 +62,6 @@
'defines': [
'FEATURE_ENABLE_SSL',
'LOGGING=1',
'USE_WEBRTC_DEV_BRANCH',
],
'sources': [
'arraysize.h',

13
webrtc/p2p/client/basicportallocator.cc
View File

@@ -414,11 +414,7 @@ void BasicPortAllocatorSession::DoAllocate() {
}
if (!(sequence_flags & PORTALLOCATOR_ENABLE_IPV6) &&
#ifdef USE_WEBRTC_DEV_BRANCH
networks[i]->GetBestIP().family() == AF_INET6) {
#else // USE_WEBRTC_DEV_BRANCH
networks[i]->ip().family() == AF_INET6) {
#endif // USE_WEBRTC_DEV_BRANCH
// Skip IPv6 networks unless the flag's been set.
continue;
}
@@ -718,12 +714,7 @@ AllocationSequence::AllocationSequence(BasicPortAllocatorSession* session,
uint32 flags)
: session_(session),
network_(network),
#ifdef USE_WEBRTC_DEV_BRANCH
ip_(network->GetBestIP()),
#else // USE_WEBRTC_DEV_BRANCH
ip_(network->ip()),
#endif // USE_WEBRTC_DEV_BRANCH
config_(config),
state_(kInit),
flags_(flags),
@@ -766,11 +757,7 @@ AllocationSequence::~AllocationSequence() {
void AllocationSequence::DisableEquivalentPhases(rtc::Network* network,
PortConfiguration* config, uint32* flags) {
#ifdef USE_WEBRTC_DEV_BRANCH
if (!((network == network_) && (ip_ == network->GetBestIP()))) {
#else // USE_WEBRTC_DEV_BRANCH
if (!((network == network_) && (ip_ == network->ip()))) {
#endif // USE_WEBRTC_DEV_BRANCH
// Different network setup; nothing is equivalent.
return;
}

36
webrtc/p2p/client/connectivitychecker.cc
View File

@@ -222,11 +222,7 @@ void ConnectivityChecker::OnRequestDone(rtc::AsyncHttpRequest* request) {
}
rtc::ProxyInfo proxy_info = request->proxy();
NicMap::iterator i =
#ifdef USE_WEBRTC_DEV_BRANCH
nics_.find(NicId(networks[0]->GetBestIP(), proxy_info.address));
#else // USE_WEBRTC_DEV_BRANCH
nics_.find(NicId(networks[0]->ip(), proxy_info.address));
#endif // USE_WEBRTC_DEV_BRANCH
if (i != nics_.end()) {
int port = request->port();
uint32 now = rtc::Time();
@@ -257,11 +253,7 @@ void ConnectivityChecker::OnRelayPortComplete(Port* port) {
ASSERT(worker_ == rtc::Thread::Current());
RelayPort* relay_port = reinterpret_cast<RelayPort*>(port);
const ProtocolAddress* address = relay_port->ServerAddress(0);
#ifdef USE_WEBRTC_DEV_BRANCH
rtc::IPAddress ip = port->Network()->GetBestIP();
#else // USE_WEBRTC_DEV_BRANCH
rtc::IPAddress ip = port->Network()->ip();
#endif // USE_WEBRTC_DEV_BRANCH
NicMap::iterator i = nics_.find(NicId(ip, port->proxy().address));
if (i != nics_.end()) {
// We have it already, add the new information.
@@ -295,11 +287,7 @@ void ConnectivityChecker::OnStunPortComplete(Port* port) {
ASSERT(worker_ == rtc::Thread::Current());
const std::vector<Candidate> candidates = port->Candidates();
Candidate c = candidates[0];
#ifdef USE_WEBRTC_DEV_BRANCH
rtc::IPAddress ip = port->Network()->GetBestIP();
#else // USE_WEBRTC_DEV_BRANCH
rtc::IPAddress ip = port->Network()->ip();
#endif // USE_WEBRTC_DEV_BRANCH
NicMap::iterator i = nics_.find(NicId(ip, port->proxy().address));
if (i != nics_.end()) {
// We have it already, add the new information.
@@ -318,11 +306,7 @@ void ConnectivityChecker::OnStunPortComplete(Port* port) {
void ConnectivityChecker::OnStunPortError(Port* port) {
ASSERT(worker_ == rtc::Thread::Current());
LOG(LS_ERROR) << "Stun address error.";
#ifdef USE_WEBRTC_DEV_BRANCH
rtc::IPAddress ip = port->Network()->GetBestIP();
#else // USE_WEBRTC_DEV_BRANCH
rtc::IPAddress ip = port->Network()->ip();
#endif // USE_WEBRTC_DEV_BRANCH
NicMap::iterator i = nics_.find(NicId(ip, port->proxy().address));
if (i != nics_.end()) {
// We have it already, add the new information.
@@ -362,11 +346,7 @@ StunPort* ConnectivityChecker::CreateStunPort(
return StunPort::Create(worker_,
socket_factory_.get(),
network,
#ifdef USE_WEBRTC_DEV_BRANCH
network->GetBestIP(),
#else // USE_WEBRTC_DEV_BRANCH
network->ip(),
#endif // USE_WEBRTC_DEV_BRANCH
0,
0,
username,
@@ -381,11 +361,7 @@ RelayPort* ConnectivityChecker::CreateRelayPort(
return RelayPort::Create(worker_,
socket_factory_.get(),
network,
#ifdef USE_WEBRTC_DEV_BRANCH
network->GetBestIP(),
#else // USE_WEBRTC_DEV_BRANCH
network->ip(),
#endif // USE_WEBRTC_DEV_BRANCH
port_allocator_->min_port(),
port_allocator_->max_port(),
username,
@@ -406,11 +382,7 @@ void ConnectivityChecker::CreateRelayPorts(
relay != config->relays.end(); ++relay) {
for (uint32 i = 0; i < networks.size(); ++i) {
NicMap::iterator iter =
#ifdef USE_WEBRTC_DEV_BRANCH
nics_.find(NicId(networks[i]->GetBestIP(), proxy_info.address));
#else // USE_WEBRTC_DEV_BRANCH
nics_.find(NicId(networks[i]->ip(), proxy_info.address));
#endif // USE_WEBRTC_DEV_BRANCH
if (iter != nics_.end()) {
// TODO: Now setting the same start time for all protocols.
// This might affect accuracy, but since we are mainly looking for
@@ -467,11 +439,7 @@ void ConnectivityChecker::AllocatePorts() {
rtc::ProxyInfo proxy_info = GetProxyInfo();
bool allocate_relay_ports = false;
for (uint32 i = 0; i < networks.size(); ++i) {
#ifdef USE_WEBRTC_DEV_BRANCH
if (AddNic(networks[i]->GetBestIP(), proxy_info.address)) {
#else // USE_WEBRTC_DEV_BRANCH
if (AddNic(networks[i]->ip(), proxy_info.address)) {
#endif // USE_WEBRTC_DEV_BRANCH
Port* port = CreateStunPort(username, password, &config, networks[i]);
if (port) {
@@ -547,11 +515,7 @@ void ConnectivityChecker::RegisterHttpStart(int port) {
}
rtc::ProxyInfo proxy_info = GetProxyInfo();
NicMap::iterator i =
#ifdef USE_WEBRTC_DEV_BRANCH
nics_.find(NicId(networks[0]->GetBestIP(), proxy_info.address));
#else // USE_WEBRTC_DEV_BRANCH
nics_.find(NicId(networks[0]->ip(), proxy_info.address));
#endif // USE_WEBRTC_DEV_BRANCH
if (i != nics_.end()) {
uint32 now = rtc::Time();
NicInfo* nic_info = &i->second;

8
webrtc/p2p/client/connectivitychecker_unittest.cc
View File

@@ -221,11 +221,7 @@ class ConnectivityCheckerForTest : public ConnectivityChecker {
return new FakeStunPort(worker(),
socket_factory_,
network,
#ifdef USE_WEBRTC_DEV_BRANCH
network->GetBestIP(),
#else // USE_WEBRTC_DEV_BRANCH
network->ip(),
#endif // USE_WEBRTC_DEV_BRANCH
kMinPort,
kMaxPort,
username,
@@ -238,11 +234,7 @@ class ConnectivityCheckerForTest : public ConnectivityChecker {
return new FakeRelayPort(worker(),
socket_factory_,
network,
#ifdef USE_WEBRTC_DEV_BRANCH
network->GetBestIP(),
#else // USE_WEBRTC_DEV_BRANCH
network->ip(),
#endif // USE_WEBRTC_DEV_BRANCH
kMinPort,
kMaxPort,
username,

4
webrtc/p2p/client/fakeportallocator.h
View File

@@ -48,11 +48,7 @@ class FakePortAllocatorSession : public PortAllocatorSession {
port_.reset(cricket::UDPPort::Create(worker_thread_,
factory_,
&network_,
#ifdef USE_WEBRTC_DEV_BRANCH
network_.GetBestIP(),
#else // USE_WEBRTC_DEV_BRANCH
network_.ip(),
#endif // USE_WEBRTC_DEV_BRANCH
0,
0,
username(),