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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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164 | 164 |
165 // Test detection at the end of a DTMF tone. | 165 // Test detection at the end of a DTMF tone. |
166 // EXPECT_EQ(0, module2->SetTelephoneEventForwardToDecoder(true)); | 166 // EXPECT_EQ(0, module2->SetTelephoneEventForwardToDecoder(true)); |
167 | 167 |
168 EXPECT_EQ(0, module1->SetSendingStatus(true)); | 168 EXPECT_EQ(0, module1->SetSendingStatus(true)); |
169 | 169 |
170 // Start basic RTP test. | 170 // Start basic RTP test. |
171 | 171 |
172 // Send an empty RTP packet. | 172 // Send an empty RTP packet. |
173 // Should fail since we have not registered the payload type. | 173 // Should fail since we have not registered the payload type. |
174 EXPECT_EQ(-1, module1->SendOutgoingData(webrtc::kAudioFrameSpeech, | 174 EXPECT_FALSE(module1->SendOutgoingData(webrtc::kAudioFrameSpeech, 96, 0, -1, |
175 96, 0, -1, NULL, 0)); | 175 nullptr, 0, nullptr, nullptr, |
| 176 nullptr)); |
176 | 177 |
177 CodecInst voice_codec; | 178 CodecInst voice_codec; |
178 memset(&voice_codec, 0, sizeof(voice_codec)); | 179 memset(&voice_codec, 0, sizeof(voice_codec)); |
179 voice_codec.pltype = 96; | 180 voice_codec.pltype = 96; |
180 voice_codec.plfreq = 8000; | 181 voice_codec.plfreq = 8000; |
181 memcpy(voice_codec.plname, "PCMU", 5); | 182 memcpy(voice_codec.plname, "PCMU", 5); |
182 | 183 |
183 EXPECT_EQ(0, module1->RegisterSendPayload(voice_codec)); | 184 EXPECT_EQ(0, module1->RegisterSendPayload(voice_codec)); |
184 EXPECT_EQ(0, rtp_receiver1_->RegisterReceivePayload( | 185 EXPECT_EQ(0, rtp_receiver1_->RegisterReceivePayload( |
185 voice_codec.plname, | 186 voice_codec.plname, |
186 voice_codec.pltype, | 187 voice_codec.pltype, |
187 voice_codec.plfreq, | 188 voice_codec.plfreq, |
188 voice_codec.channels, | 189 voice_codec.channels, |
189 (voice_codec.rate < 0) ? 0 : voice_codec.rate)); | 190 (voice_codec.rate < 0) ? 0 : voice_codec.rate)); |
190 EXPECT_EQ(0, module2->RegisterSendPayload(voice_codec)); | 191 EXPECT_EQ(0, module2->RegisterSendPayload(voice_codec)); |
191 voice_codec.rate = test_rate; | 192 voice_codec.rate = test_rate; |
192 EXPECT_EQ(0, rtp_receiver2_->RegisterReceivePayload( | 193 EXPECT_EQ(0, rtp_receiver2_->RegisterReceivePayload( |
193 voice_codec.plname, | 194 voice_codec.plname, |
194 voice_codec.pltype, | 195 voice_codec.pltype, |
195 voice_codec.plfreq, | 196 voice_codec.plfreq, |
196 voice_codec.channels, | 197 voice_codec.channels, |
197 (voice_codec.rate < 0) ? 0 : voice_codec.rate)); | 198 (voice_codec.rate < 0) ? 0 : voice_codec.rate)); |
198 | 199 |
199 const uint8_t test[5] = "test"; | 200 const uint8_t test[5] = "test"; |
200 EXPECT_EQ(0, module1->SendOutgoingData(webrtc::kAudioFrameSpeech, 96, | 201 EXPECT_EQ(true, |
201 0, -1, test, 4)); | 202 module1->SendOutgoingData(webrtc::kAudioFrameSpeech, 96, 0, -1, |
| 203 test, 4, nullptr, nullptr, nullptr)); |
202 | 204 |
203 EXPECT_EQ(test_ssrc, rtp_receiver2_->SSRC()); | 205 EXPECT_EQ(test_ssrc, rtp_receiver2_->SSRC()); |
204 uint32_t timestamp; | 206 uint32_t timestamp; |
205 EXPECT_TRUE(rtp_receiver2_->Timestamp(×tamp)); | 207 EXPECT_TRUE(rtp_receiver2_->Timestamp(×tamp)); |
206 EXPECT_EQ(test_timestamp, timestamp); | 208 EXPECT_EQ(test_timestamp, timestamp); |
207 } | 209 } |
208 | 210 |
209 TEST_F(RtpRtcpAudioTest, RED) { | 211 TEST_F(RtpRtcpAudioTest, RED) { |
210 CodecInst voice_codec; | 212 CodecInst voice_codec; |
211 memset(&voice_codec, 0, sizeof(voice_codec)); | 213 memset(&voice_codec, 0, sizeof(voice_codec)); |
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264 fragmentation.fragmentationOffset[1] = 4; | 266 fragmentation.fragmentationOffset[1] = 4; |
265 fragmentation.fragmentationTimeDiff = new uint16_t[2]; | 267 fragmentation.fragmentationTimeDiff = new uint16_t[2]; |
266 fragmentation.fragmentationTimeDiff[0] = 0; | 268 fragmentation.fragmentationTimeDiff[0] = 0; |
267 fragmentation.fragmentationTimeDiff[1] = 0; | 269 fragmentation.fragmentationTimeDiff[1] = 0; |
268 fragmentation.fragmentationPlType = new uint8_t[2]; | 270 fragmentation.fragmentationPlType = new uint8_t[2]; |
269 fragmentation.fragmentationPlType[0] = 96; | 271 fragmentation.fragmentationPlType[0] = 96; |
270 fragmentation.fragmentationPlType[1] = 96; | 272 fragmentation.fragmentationPlType[1] = 96; |
271 | 273 |
272 const uint8_t test[5] = "test"; | 274 const uint8_t test[5] = "test"; |
273 // Send a RTP packet. | 275 // Send a RTP packet. |
274 EXPECT_EQ(0, module1->SendOutgoingData(webrtc::kAudioFrameSpeech, | 276 EXPECT_TRUE(module1->SendOutgoingData(webrtc::kAudioFrameSpeech, 96, 160, -1, |
275 96, 160, -1, test, 4, | 277 test, 4, &fragmentation, nullptr, |
276 &fragmentation)); | 278 nullptr)); |
277 | 279 |
278 EXPECT_EQ(0, module1->SetSendREDPayloadType(-1)); | 280 EXPECT_EQ(0, module1->SetSendREDPayloadType(-1)); |
279 EXPECT_EQ(-1, module1->SendREDPayloadType(&red)); | 281 EXPECT_EQ(-1, module1->SendREDPayloadType(&red)); |
280 } | 282 } |
281 | 283 |
282 TEST_F(RtpRtcpAudioTest, DTMF) { | 284 TEST_F(RtpRtcpAudioTest, DTMF) { |
283 CodecInst voice_codec; | 285 CodecInst voice_codec; |
284 memset(&voice_codec, 0, sizeof(voice_codec)); | 286 memset(&voice_codec, 0, sizeof(voice_codec)); |
285 voice_codec.pltype = 96; | 287 voice_codec.pltype = 96; |
286 voice_codec.plfreq = 8000; | 288 voice_codec.plfreq = 8000; |
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326 for (int i = 0; i < 16; i++) { | 328 for (int i = 0; i < 16; i++) { |
327 EXPECT_EQ(0, module1->SendTelephoneEventOutband(i, timeStamp, 10)); | 329 EXPECT_EQ(0, module1->SendTelephoneEventOutband(i, timeStamp, 10)); |
328 } | 330 } |
329 timeStamp += 160; // Prepare for next packet. | 331 timeStamp += 160; // Prepare for next packet. |
330 | 332 |
331 const uint8_t test[9] = "test"; | 333 const uint8_t test[9] = "test"; |
332 | 334 |
333 // Send RTP packets for 16 tones a 160 ms 100ms | 335 // Send RTP packets for 16 tones a 160 ms 100ms |
334 // pause between = 2560ms + 1600ms = 4160ms | 336 // pause between = 2560ms + 1600ms = 4160ms |
335 for (; timeStamp <= 250 * 160; timeStamp += 160) { | 337 for (; timeStamp <= 250 * 160; timeStamp += 160) { |
336 EXPECT_EQ(0, module1->SendOutgoingData(webrtc::kAudioFrameSpeech, 96, | 338 EXPECT_TRUE(module1->SendOutgoingData(webrtc::kAudioFrameSpeech, 96, |
337 timeStamp, -1, test, 4)); | 339 timeStamp, -1, test, 4, nullptr, |
| 340 nullptr, nullptr)); |
338 fake_clock.AdvanceTimeMilliseconds(20); | 341 fake_clock.AdvanceTimeMilliseconds(20); |
339 module1->Process(); | 342 module1->Process(); |
340 } | 343 } |
341 EXPECT_EQ(0, module1->SendTelephoneEventOutband(32, 9000, 10)); | 344 EXPECT_EQ(0, module1->SendTelephoneEventOutband(32, 9000, 10)); |
342 | 345 |
343 for (; timeStamp <= 740 * 160; timeStamp += 160) { | 346 for (; timeStamp <= 740 * 160; timeStamp += 160) { |
344 EXPECT_EQ(0, module1->SendOutgoingData(webrtc::kAudioFrameSpeech, 96, | 347 EXPECT_TRUE(module1->SendOutgoingData(webrtc::kAudioFrameSpeech, 96, |
345 timeStamp, -1, test, 4)); | 348 timeStamp, -1, test, 4, nullptr, |
| 349 nullptr, nullptr)); |
346 fake_clock.AdvanceTimeMilliseconds(20); | 350 fake_clock.AdvanceTimeMilliseconds(20); |
347 module1->Process(); | 351 module1->Process(); |
348 } | 352 } |
349 } | 353 } |
350 | 354 |
351 } // namespace | 355 } // namespace |
352 } // namespace webrtc | 356 } // namespace webrtc |
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