Roll chromium_revision 288251:289723
Mainly to pick up the libvpx.gyp change in r288724 to unblock https://webrtc-codereview.appspot.com/16229005/ Overview of changes in Chrome DEPS: $ svn diff http://src.chromium.org/chrome/trunk/src/DEPS -r 288251:289723 which can be compared with the output of: $ svn cat http://webrtc.googlecode.com/svn/trunk/DEPS | grep chromium_deps | sed 's/^ *//' | sort | uniq In a WebRTC checkout, that sums up to the following relevant changes: * src/buildtools 59b932:567f0a * testing/gtest 643:692 * testing/gmock 410:485 * third_party/boringssl/src 533cbe:c3d796 * third_party/libvpx 287125:289332 * third_party/libyuv 1035:1038 * third_party/nss 287121:289430 * third_party/opus/src 256783:289085 * tools/gyp 1959:1964 BUG=2863, chromium:339647 TEST=Local testing as trybots currently cannot handle DEPS changes properly due to http://crbug.com/385594 R=tommi@webrtc.org Review URL: https://webrtc-codereview.appspot.com/22119004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6913 4adac7df-926f-26a2-2b94-8c16560cd09d
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2
DEPS
2
DEPS
@ -13,7 +13,7 @@ vars = {
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"chromium_trunk" : "http://src.chromium.org/svn/trunk",
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# chrome://version/ for revision of canary Chrome.
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# http://chromium-status.appspot.com/lkgr is a last known good revision.
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"chromium_revision": "288251",
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"chromium_revision": "289723",
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# A small subset of WebKit is needed for the Android Python test framework.
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"webkit_trunk": "http://src.chromium.org/blink/trunk",
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@ -13,8 +13,6 @@
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#include "webrtc/modules/audio_coding/codecs/tools/audio_codec_speed_test.h"
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using ::std::string;
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using ::std::tr1::make_tuple;
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using ::testing::ValuesIn;
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namespace webrtc {
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@ -101,10 +99,10 @@ TEST_P(IsacSpeedTest, IsacEncodeDecodeTest) {
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}
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const coding_param param_set[] =
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{make_tuple(1, 32000, string("audio_coding/speech_mono_16kHz"),
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string("pcm"), true)};
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{::std::tr1::make_tuple(1, 32000, string("audio_coding/speech_mono_16kHz"),
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string("pcm"), true)};
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INSTANTIATE_TEST_CASE_P(AllTest, IsacSpeedTest,
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ValuesIn(param_set));
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::testing::ValuesIn(param_set));
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} // namespace webrtc
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@ -15,10 +15,8 @@
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using ::std::string;
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using ::std::tr1::tuple;
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using ::std::tr1::make_tuple;
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using ::std::tr1::get;
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using ::testing::TestWithParam;
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using ::testing::ValuesIn;
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namespace webrtc {
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@ -226,15 +224,15 @@ TEST_P(OpusFecTest, RandomPacketLossTest) {
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}
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const coding_param param_set[] =
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{make_tuple(1, 64000, string("audio_coding/testfile32kHz"),
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string("pcm")),
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make_tuple(1, 32000, string("audio_coding/testfile32kHz"),
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string("pcm")),
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make_tuple(2, 64000, string("audio_coding/teststereo32kHz"),
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string("pcm"))};
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{::std::tr1::make_tuple(1, 64000, string("audio_coding/testfile32kHz"),
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string("pcm")),
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::std::tr1::make_tuple(1, 32000, string("audio_coding/testfile32kHz"),
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string("pcm")),
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::std::tr1::make_tuple(2, 64000, string("audio_coding/teststereo32kHz"),
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string("pcm"))};
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// 64 kbps, stereo
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INSTANTIATE_TEST_CASE_P(AllTest, OpusFecTest,
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ValuesIn(param_set));
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::testing::ValuesIn(param_set));
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} // namespace webrtc
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@ -12,8 +12,6 @@
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#include "webrtc/modules/audio_coding/codecs/tools/audio_codec_speed_test.h"
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using ::std::string;
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using ::std::tr1::make_tuple;
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using ::testing::ValuesIn;
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namespace webrtc {
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@ -105,14 +103,17 @@ ADD_TEST(0);
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// List all test cases: (channel, bit rat, filename, extension).
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const coding_param param_set[] =
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{make_tuple(1, 64000, string("audio_coding/speech_mono_32_48kHz"),
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string("pcm"), true),
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make_tuple(1, 32000, string("audio_coding/speech_mono_32_48kHz"),
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string("pcm"), true),
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make_tuple(2, 64000, string("audio_coding/music_stereo_48kHz"),
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string("pcm"), true)};
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{::std::tr1::make_tuple(1, 64000,
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string("audio_coding/speech_mono_32_48kHz"),
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string("pcm"), true),
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::std::tr1::make_tuple(1, 32000,
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string("audio_coding/speech_mono_32_48kHz"),
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string("pcm"), true),
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::std::tr1::make_tuple(2, 64000,
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string("audio_coding/music_stereo_48kHz"),
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string("pcm"), true)};
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INSTANTIATE_TEST_CASE_P(AllTest, OpusSpeedTest,
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ValuesIn(param_set));
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::testing::ValuesIn(param_set));
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} // namespace webrtc
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