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1 /* | 1 /* |
2 * Copyright 2009 The WebRTC project authors. All Rights Reserved. | 2 * Copyright 2009 The WebRTC project authors. All Rights Reserved. |
3 * | 3 * |
4 * Use of this source code is governed by a BSD-style license | 4 * Use of this source code is governed by a BSD-style license |
5 * that can be found in the LICENSE file in the root of the source | 5 * that can be found in the LICENSE file in the root of the source |
6 * tree. An additional intellectual property rights grant can be found | 6 * tree. An additional intellectual property rights grant can be found |
7 * in the file PATENTS. All contributing project authors may | 7 * in the file PATENTS. All contributing project authors may |
8 * be found in the AUTHORS file in the root of the source tree. | 8 * be found in the AUTHORS file in the root of the source tree. |
9 */ | 9 */ |
10 | 10 |
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2350 } | 2350 } |
2351 | 2351 |
2352 TEST_F(VoiceChannelTest, SendBundleToBundleWithRtcpMux) { | 2352 TEST_F(VoiceChannelTest, SendBundleToBundleWithRtcpMux) { |
2353 Base::SendBundleToBundle(kAudioPts, arraysize(kAudioPts), true, false); | 2353 Base::SendBundleToBundle(kAudioPts, arraysize(kAudioPts), true, false); |
2354 } | 2354 } |
2355 | 2355 |
2356 TEST_F(VoiceChannelTest, SendBundleToBundleWithRtcpMuxSecure) { | 2356 TEST_F(VoiceChannelTest, SendBundleToBundleWithRtcpMuxSecure) { |
2357 Base::SendBundleToBundle(kAudioPts, arraysize(kAudioPts), true, true); | 2357 Base::SendBundleToBundle(kAudioPts, arraysize(kAudioPts), true, true); |
2358 } | 2358 } |
2359 | 2359 |
2360 TEST_F(VoiceChannelTest, GetRtpParametersIsNotImplemented) { | 2360 TEST_F(VoiceChannelTest, DefaultMaxBitrateIsUnlimited) { |
2361 // These tests verify that the Get/SetRtpParameters methods for VoiceChannel | 2361 Base::DefaultMaxBitrateIsUnlimited(); |
2362 // always fail as they are not implemented. | |
2363 // TODO(skvlad): Replace with full tests when support for bitrate limiting | |
2364 // for audio RtpSenders is added. | |
2365 CreateChannels(0, 0); | |
2366 EXPECT_TRUE( | |
2367 channel1_->SetLocalContent(&local_media_content1_, CA_OFFER, NULL)); | |
2368 webrtc::RtpParameters voice_parameters = channel1_->GetRtpParameters(kSsrc1); | |
2369 EXPECT_EQ(0UL, voice_parameters.encodings.size()); | |
2370 } | 2362 } |
2371 | 2363 |
2372 TEST_F(VoiceChannelTest, SetRtpParametersIsNotImplemented) { | 2364 TEST_F(VoiceChannelTest, CanChangeMaxBitrate) { |
2373 CreateChannels(0, 0); | 2365 Base::CanChangeMaxBitrate(); |
2374 EXPECT_TRUE( | |
2375 channel1_->SetLocalContent(&local_media_content1_, CA_OFFER, NULL)); | |
2376 EXPECT_FALSE( | |
2377 channel1_->SetRtpParameters(kSsrc1, BitrateLimitedParameters(1000))); | |
2378 } | 2366 } |
2379 | 2367 |
2380 // VideoChannelTest | 2368 // VideoChannelTest |
2381 TEST_F(VideoChannelTest, TestInit) { | 2369 TEST_F(VideoChannelTest, TestInit) { |
2382 Base::TestInit(); | 2370 Base::TestInit(); |
2383 } | 2371 } |
2384 | 2372 |
2385 TEST_F(VideoChannelTest, TestSetContents) { | 2373 TEST_F(VideoChannelTest, TestSetContents) { |
2386 Base::TestSetContents(); | 2374 Base::TestSetContents(); |
2387 } | 2375 } |
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2822 }; | 2810 }; |
2823 rtc::CopyOnWriteBuffer payload(data, 3); | 2811 rtc::CopyOnWriteBuffer payload(data, 3); |
2824 cricket::SendDataResult result; | 2812 cricket::SendDataResult result; |
2825 ASSERT_TRUE(media_channel1_->SendData(params, payload, &result)); | 2813 ASSERT_TRUE(media_channel1_->SendData(params, payload, &result)); |
2826 EXPECT_EQ(params.ssrc, | 2814 EXPECT_EQ(params.ssrc, |
2827 media_channel1_->last_sent_data_params().ssrc); | 2815 media_channel1_->last_sent_data_params().ssrc); |
2828 EXPECT_EQ("foo", media_channel1_->last_sent_data()); | 2816 EXPECT_EQ("foo", media_channel1_->last_sent_data()); |
2829 } | 2817 } |
2830 | 2818 |
2831 // TODO(pthatcher): TestSetReceiver? | 2819 // TODO(pthatcher): TestSetReceiver? |
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