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Allow WebRTC to offer receiving capability for 120ms Opus packets.

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TEST=Build Chromium for receiving + a special AppRTCDemo built with 120ms Opus sending capability. Call went well.

BUG=
R=solenberg@webrtc.org

Review URL: https://codereview.webrtc.org/1957963002 .

Cr-Commit-Position: refs/heads/master@{#12673}
This commit is contained in:
minyuel 2016-05-10 15:35:21 +02:00
parent f3995f71ce
commit b031a2e862
2 changed files with 1 additions and 3 deletions
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2
webrtc/media/base/mediaconstants.cc
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@ -64,7 +64,7 @@ const int kOpusDefaultUseInbandFec = 0;
const int kOpusDefaultUseDtx = 0;
const int kOpusDefaultMaxPlaybackRate = 48000;
const int kPreferredMaxPTime = 60;
const int kPreferredMaxPTime = 120;
const int kPreferredMinPTime = 10;
const int kPreferredSPropStereo = 0;
const int kPreferredStereo = 0;

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

@ -3610,8 +3610,6 @@ TEST(WebRtcVoiceEngineTest, HasCorrectCodecs) {
EXPECT_EQ(111, it->id);
ASSERT_TRUE(it->params.find("minptime") != it->params.end());
EXPECT_EQ("10", it->params.find("minptime")->second);
ASSERT_TRUE(it->params.find("maxptime") != it->params.end());
EXPECT_EQ("60", it->params.find("maxptime")->second);
ASSERT_TRUE(it->params.find("useinbandfec") != it->params.end());
EXPECT_EQ("1", it->params.find("useinbandfec")->second);
}