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Side by Side Diff: webrtc/modules/rtp_rtcp/test/testAPI/test_api_audio.cc

Issue 2066473002: RtpRtcp: Remove the SetSendREDPayloadType and SendREDPayloadType methods (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@remove-red3
Patch Set: rebase Created 4 years, 4 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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201 EXPECT_EQ(true, 201 EXPECT_EQ(true,
202 module1->SendOutgoingData(webrtc::kAudioFrameSpeech, 96, 0, -1, 202 module1->SendOutgoingData(webrtc::kAudioFrameSpeech, 96, 0, -1,
203 test, 4, nullptr, nullptr, nullptr)); 203 test, 4, nullptr, nullptr, nullptr));
204 204
205 EXPECT_EQ(test_ssrc, rtp_receiver2_->SSRC()); 205 EXPECT_EQ(test_ssrc, rtp_receiver2_->SSRC());
206 uint32_t timestamp; 206 uint32_t timestamp;
207 EXPECT_TRUE(rtp_receiver2_->Timestamp(&timestamp)); 207 EXPECT_TRUE(rtp_receiver2_->Timestamp(&timestamp));
208 EXPECT_EQ(test_timestamp, timestamp); 208 EXPECT_EQ(test_timestamp, timestamp);
209 } 209 }
210 210
211 TEST_F(RtpRtcpAudioTest, RED) {
212 CodecInst voice_codec;
213 memset(&voice_codec, 0, sizeof(voice_codec));
214 voice_codec.pltype = 96;
215 voice_codec.plfreq = 8000;
216 memcpy(voice_codec.plname, "PCMU", 5);
217
218 EXPECT_EQ(0, module1->RegisterSendPayload(voice_codec));
219 EXPECT_EQ(0, rtp_receiver1_->RegisterReceivePayload(
220 voice_codec.plname,
221 voice_codec.pltype,
222 voice_codec.plfreq,
223 voice_codec.channels,
224 (voice_codec.rate < 0) ? 0 : voice_codec.rate));
225 EXPECT_EQ(0, module2->RegisterSendPayload(voice_codec));
226 voice_codec.rate = test_rate;
227 EXPECT_EQ(0, rtp_receiver2_->RegisterReceivePayload(
228 voice_codec.plname,
229 voice_codec.pltype,
230 voice_codec.plfreq,
231 voice_codec.channels,
232 (voice_codec.rate < 0) ? 0 : voice_codec.rate));
233
234 module1->SetSSRC(test_ssrc);
235 module1->SetStartTimestamp(test_timestamp);
236 EXPECT_EQ(0, module1->SetSendingStatus(true));
237
238 voice_codec.pltype = 127;
239 voice_codec.plfreq = 8000;
240 memcpy(voice_codec.plname, "RED", 4);
241
242 EXPECT_EQ(0, module1->SetSendREDPayloadType(voice_codec.pltype));
243 int8_t red = 0;
244 EXPECT_EQ(0, module1->SendREDPayloadType(&red));
245 EXPECT_EQ(voice_codec.pltype, red);
246 EXPECT_EQ(0, rtp_receiver1_->RegisterReceivePayload(
247 voice_codec.plname,
248 voice_codec.pltype,
249 voice_codec.plfreq,
250 voice_codec.channels,
251 (voice_codec.rate < 0) ? 0 : voice_codec.rate));
252 EXPECT_EQ(0, rtp_receiver2_->RegisterReceivePayload(
253 voice_codec.plname,
254 voice_codec.pltype,
255 voice_codec.plfreq,
256 voice_codec.channels,
257 (voice_codec.rate < 0) ? 0 : voice_codec.rate));
258
259 RTPFragmentationHeader fragmentation;
260 fragmentation.fragmentationVectorSize = 2;
261 fragmentation.fragmentationLength = new size_t[2];
262 fragmentation.fragmentationLength[0] = 4;
263 fragmentation.fragmentationLength[1] = 4;
264 fragmentation.fragmentationOffset = new size_t[2];
265 fragmentation.fragmentationOffset[0] = 0;
266 fragmentation.fragmentationOffset[1] = 4;
267 fragmentation.fragmentationTimeDiff = new uint16_t[2];
268 fragmentation.fragmentationTimeDiff[0] = 0;
269 fragmentation.fragmentationTimeDiff[1] = 0;
270 fragmentation.fragmentationPlType = new uint8_t[2];
271 fragmentation.fragmentationPlType[0] = 96;
272 fragmentation.fragmentationPlType[1] = 96;
273
274 const uint8_t test[5] = "test";
275 // Send a RTP packet.
276 EXPECT_TRUE(module1->SendOutgoingData(webrtc::kAudioFrameSpeech, 96, 160, -1,
277 test, 4, &fragmentation, nullptr,
278 nullptr));
279
280 EXPECT_EQ(0, module1->SetSendREDPayloadType(-1));
281 EXPECT_EQ(-1, module1->SendREDPayloadType(&red));
282 }
283
284 TEST_F(RtpRtcpAudioTest, DTMF) { 211 TEST_F(RtpRtcpAudioTest, DTMF) {
285 CodecInst voice_codec; 212 CodecInst voice_codec;
286 memset(&voice_codec, 0, sizeof(voice_codec)); 213 memset(&voice_codec, 0, sizeof(voice_codec));
287 voice_codec.pltype = 96; 214 voice_codec.pltype = 96;
288 voice_codec.plfreq = 8000; 215 voice_codec.plfreq = 8000;
289 memcpy(voice_codec.plname, "PCMU", 5); 216 memcpy(voice_codec.plname, "PCMU", 5);
290 217
291 EXPECT_EQ(0, module1->RegisterSendPayload(voice_codec)); 218 EXPECT_EQ(0, module1->RegisterSendPayload(voice_codec));
292 EXPECT_EQ(0, rtp_receiver1_->RegisterReceivePayload( 219 EXPECT_EQ(0, rtp_receiver1_->RegisterReceivePayload(
293 voice_codec.plname, 220 voice_codec.plname,
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347 EXPECT_TRUE(module1->SendOutgoingData(webrtc::kAudioFrameSpeech, 96, 274 EXPECT_TRUE(module1->SendOutgoingData(webrtc::kAudioFrameSpeech, 96,
348 timeStamp, -1, test, 4, nullptr, 275 timeStamp, -1, test, 4, nullptr,
349 nullptr, nullptr)); 276 nullptr, nullptr));
350 fake_clock.AdvanceTimeMilliseconds(20); 277 fake_clock.AdvanceTimeMilliseconds(20);
351 module1->Process(); 278 module1->Process();
352 } 279 }
353 } 280 }
354 281
355 } // namespace 282 } // namespace
356 } // namespace webrtc 283 } // namespace webrtc
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