Commit 1b5df0c3 authored by nicholss's avatar nicholss Committed by Commit bot

Removing the dependency on //third_party/webrtc:webrtc and replacing with rtc_base.

R=sergeyu@chromium.org

BUG=653612

Review-Url: https://codereview.chromium.org/2437453004
Cr-Commit-Position: refs/heads/master@{#427371}
parent a28343f8
......@@ -140,6 +140,7 @@ test("remoting_unittests") {
deps = [
":test_support",
"//base",
"//base/test:test_support",
"//google_apis",
"//remoting/base:unit_tests",
"//remoting/client:unit_tests",
......@@ -148,7 +149,6 @@ test("remoting_unittests") {
"//remoting/test:unit_tests",
"//testing/gmock",
"//testing/gtest",
"//third_party/webrtc",
]
if (enable_remoting_host) {
......
......@@ -98,7 +98,7 @@ source_set("opengl_renderer") {
deps = [
"//remoting/proto",
"//third_party/libyuv",
"//third_party/webrtc",
"//third_party/webrtc/base:rtc_base",
]
configs += [ "//third_party/khronos:khronos_headers" ]
......@@ -111,6 +111,13 @@ source_set("opengl_renderer") {
libs = [ "OpenGL.framework" ]
}
if (is_ios) {
libs = [
"GLKit.framework",
"OpenGLES.framework",
]
}
if (is_win) {
deps += [ "//third_party/angle:libGLESv2" ]
}
......@@ -150,7 +157,7 @@ source_set("unit_tests") {
"//remoting/proto",
"//testing/gmock",
"//testing/gtest",
"//third_party/webrtc",
"//third_party/webrtc/base:rtc_base",
]
if (!is_win) {
......
......@@ -142,4 +142,8 @@ source_set("unit_tests") {
"//testing/gtest",
"//third_party/webrtc/modules/desktop_capture",
]
if (is_ios) {
sources -= [ "audio_encoder_opus_unittest.cc" ]
}
}
......@@ -32,11 +32,17 @@ namespace {
std::unique_ptr<AudioEncoder> CreateAudioEncoder(
const protocol::SessionConfig& config) {
#if defined(OS_IOS)
// TODO(nicholss): iOS should not use Opus. This is to prevent us from
// depending on //media. In the future we will use webrtc for conneciton
// and this will be a non-issue.
return nullptr;
#else
const protocol::ChannelConfig& audio_config = config.audio_config();
if (audio_config.codec == protocol::ChannelConfig::CODEC_OPUS) {
return base::WrapUnique(new AudioEncoderOpus());
}
#endif
NOTREACHED();
return nullptr;
......
......@@ -488,12 +488,13 @@ int32_t WebrtcAudioModule::EnableBuiltInNS(bool enable) {
#if defined(WEBRTC_IOS)
int WebrtcAudioModule::GetPlayoutAudioParameters(
AudioParameters* params) const {
webrtc::AudioParameters* params) const {
NOTREACHED();
return -1;
}
int WebrtcAudioModule::GetRecordAudioParameters(AudioParameters* params) const {
int WebrtcAudioModule::GetRecordAudioParameters(
webrtc::AudioParameters* params) const {
NOTREACHED();
return -1;
}
......
......@@ -135,8 +135,8 @@ class WebrtcAudioModule : public webrtc::AudioDeviceModule {
// Only supported on iOS.
#if defined(WEBRTC_IOS)
int GetPlayoutAudioParameters(AudioParameters* params) const override;
int GetRecordAudioParameters(AudioParameters* params) const override;
int GetPlayoutAudioParameters(webrtc::AudioParameters* params) const override;
int GetRecordAudioParameters(webrtc::AudioParameters* params) const override;
#endif // WEBRTC_IOS
private:
......
......@@ -180,9 +180,15 @@ int main(int argc, char** argv) {
PrintUsage();
PrintJsonFileInfo();
PrintAuthCodeInfo();
#if defined(OS_IOS)
return base::LaunchUnitTests(
argc, argv,
base::Bind(&base::TestSuite::Run, base::Unretained(&test_suite)));
#else
return base::LaunchUnitTestsSerially(
argc, argv,
base::Bind(&base::TestSuite::Run, base::Unretained(&test_suite)));
#endif
}
remoting::test::AppRemotingTestDriverEnvironment::EnvironmentOptions options;
......@@ -249,7 +255,13 @@ int main(int argc, char** argv) {
// Because many tests may access the same remoting host(s), we need to run
// the tests sequentially so they do not interfere with each other.
#if defined(OS_IOS)
return base::LaunchUnitTests(
argc, argv,
base::Bind(&base::TestSuite::Run, base::Unretained(&test_suite)));
#else
return base::LaunchUnitTestsSerially(
argc, argv,
base::Bind(&base::TestSuite::Run, base::Unretained(&test_suite)));
#endif
}
......@@ -164,9 +164,15 @@ int main(int argc, char* argv[]) {
PrintUsage();
PrintJsonFileInfo();
PrintAuthCodeInfo();
#if defined(OS_IOS)
return base::LaunchUnitTests(
argc, argv,
base::Bind(&base::TestSuite::Run, base::Unretained(&test_suite)));
#else
return base::LaunchUnitTestsSerially(
argc, argv,
base::Bind(&base::TestSuite::Run, base::Unretained(&test_suite)));
#endif
}
// Update the logging verbosity level if user specified one.
......@@ -242,7 +248,13 @@ int main(int argc, char* argv[]) {
// Running the tests serially will avoid clients from connecting to the same
// host.
#if defined(OS_IOS)
return base::LaunchUnitTests(
argc, argv,
base::Bind(&base::TestSuite::Run, base::Unretained(&test_suite)));
#else
return base::LaunchUnitTestsSerially(
argc, argv,
base::Bind(&base::TestSuite::Run, base::Unretained(&test_suite)));
#endif
}
......@@ -113,7 +113,6 @@ if (enable_webrtc) {
":libjingle",
"//third_party/libsrtp",
"//third_party/usrsctp",
"//third_party/webrtc",
"//third_party/webrtc/api:libjingle_peerconnection",
"//third_party/webrtc/media:rtc_media",
"//third_party/webrtc/modules/media_file",
......
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