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Side by Side Diff: webrtc/pc/channel_unittest.cc

Issue 1847353004: Allow applications to control audio send bitrate through RtpParameters. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Redesigned to keep the logic inside WebRtcVoiceMediaChannel Created 4 years, 8 months ago
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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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