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Side by Side Diff: webrtc/voice_engine/test/auto_test/standard/mixing_test.cc

Issue 1335353005: Remove channel ids from various interfaces. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: rebase Created 5 years, 3 months ago
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1 /* 1 /*
2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2012 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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28 const CodecInst kCodecOpus = {kPayloadType, "opus", 48000, 960, 1, 32000}; 28 const CodecInst kCodecOpus = {kPayloadType, "opus", 48000, 960, 1, 32000};
29 29
30 } // namespace 30 } // namespace
31 31
32 class MixingTest : public AfterInitializationFixture { 32 class MixingTest : public AfterInitializationFixture {
33 protected: 33 protected:
34 MixingTest() 34 MixingTest()
35 : output_filename_(test::OutputPath() + "mixing_test_output.pcm") { 35 : output_filename_(test::OutputPath() + "mixing_test_output.pcm") {
36 } 36 }
37 void SetUp() { 37 void SetUp() {
38 transport_ = new LoopBackTransport(voe_network_); 38 transport_ = new LoopBackTransport(voe_network_, 0);
39 } 39 }
40 void TearDown() { 40 void TearDown() {
41 delete transport_; 41 delete transport_;
42 } 42 }
43 43
44 // Creates and mixes |num_remote_streams| which play a file "as microphone" 44 // Creates and mixes |num_remote_streams| which play a file "as microphone"
45 // with |num_local_streams| which play a file "locally", using a constant 45 // with |num_local_streams| which play a file "locally", using a constant
46 // amplitude of |input_value|. The local streams manifest as "anonymous" 46 // amplitude of |input_value|. The local streams manifest as "anonymous"
47 // mixing participants, meaning they will be mixed regardless of the number 47 // mixing participants, meaning they will be mixed regardless of the number
48 // of participants. (A stream is a VoiceEngine "channel"). 48 // of participants. (A stream is a VoiceEngine "channel").
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175 codec_inst_stereo.pltype++; 175 codec_inst_stereo.pltype++;
176 for (size_t i = num_remote_streams_using_mono; i < streams.size(); ++i) { 176 for (size_t i = num_remote_streams_using_mono; i < streams.size(); ++i) {
177 StartRemoteStream(streams[i], codec_inst_stereo, 1234 + 2 * i); 177 StartRemoteStream(streams[i], codec_inst_stereo, 1234 + 2 * i);
178 } 178 }
179 } 179 }
180 180
181 // Start up a single remote stream. 181 // Start up a single remote stream.
182 void StartRemoteStream(int stream, const CodecInst& codec_inst, int port) { 182 void StartRemoteStream(int stream, const CodecInst& codec_inst, int port) {
183 EXPECT_EQ(0, voe_codec_->SetRecPayloadType(stream, codec_inst)); 183 EXPECT_EQ(0, voe_codec_->SetRecPayloadType(stream, codec_inst));
184 EXPECT_EQ(0, voe_network_->RegisterExternalTransport(stream, *transport_)); 184 EXPECT_EQ(0, voe_network_->RegisterExternalTransport(stream, *transport_));
185 EXPECT_EQ(0, voe_rtp_rtcp_->SetLocalSSRC(
186 stream, static_cast<unsigned int>(stream)));
187 transport_->AddChannel(stream, stream);
185 EXPECT_EQ(0, voe_base_->StartReceive(stream)); 188 EXPECT_EQ(0, voe_base_->StartReceive(stream));
186 EXPECT_EQ(0, voe_base_->StartPlayout(stream)); 189 EXPECT_EQ(0, voe_base_->StartPlayout(stream));
187 EXPECT_EQ(0, voe_codec_->SetSendCodec(stream, codec_inst)); 190 EXPECT_EQ(0, voe_codec_->SetSendCodec(stream, codec_inst));
188 EXPECT_EQ(0, voe_base_->StartSend(stream)); 191 EXPECT_EQ(0, voe_base_->StartSend(stream));
189 EXPECT_EQ(0, voe_file_->StartPlayingFileAsMicrophone(stream, 192 EXPECT_EQ(0, voe_file_->StartPlayingFileAsMicrophone(stream,
190 input_filename_.c_str(), true)); 193 input_filename_.c_str(), true));
191 } 194 }
192 195
193 void StopRemoteStreams(const std::vector<int>& streams) { 196 void StopRemoteStreams(const std::vector<int>& streams) {
194 for (size_t i = 0; i < streams.size(); ++i) { 197 for (size_t i = 0; i < streams.size(); ++i) {
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276 279
277 TEST_F(MixingTest, VerifyStereoAndMonoMixing) { 280 TEST_F(MixingTest, VerifyStereoAndMonoMixing) {
278 const int16_t kInputValue = 1000; 281 const int16_t kInputValue = 1000;
279 const int16_t kExpectedOutput = kInputValue * 2; 282 const int16_t kExpectedOutput = kInputValue * 2;
280 RunMixingTest(2, 0, 1, false, kInputValue, 1.1 * kExpectedOutput, 283 RunMixingTest(2, 0, 1, false, kInputValue, 1.1 * kExpectedOutput,
281 // Lower than 0.9 due to observed flakiness on bots. 284 // Lower than 0.9 due to observed flakiness on bots.
282 0.8 * kExpectedOutput, kCodecL16); 285 0.8 * kExpectedOutput, kCodecL16);
283 } 286 }
284 287
285 } // namespace webrtc 288 } // namespace webrtc
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