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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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1292 EXPECT_FALSE(voice_activity); | 1292 EXPECT_FALSE(voice_activity); |
1293 } | 1293 } |
1294 | 1294 |
1295 // As RFC4733, named telephone events are carried as part of the audio stream | 1295 // As RFC4733, named telephone events are carried as part of the audio stream |
1296 // and must use the same sequence number and timestamp base as the regular | 1296 // and must use the same sequence number and timestamp base as the regular |
1297 // audio channel. | 1297 // audio channel. |
1298 // This test checks the marker bit for the first packet and the consequent | 1298 // This test checks the marker bit for the first packet and the consequent |
1299 // packets of the same telephone event. Since it is specifically for DTMF | 1299 // packets of the same telephone event. Since it is specifically for DTMF |
1300 // events, ignoring audio packets and sending kEmptyFrame instead of those. | 1300 // events, ignoring audio packets and sending kEmptyFrame instead of those. |
1301 TEST_F(RtpSenderAudioTest, CheckMarkerBitForTelephoneEvents) { | 1301 TEST_F(RtpSenderAudioTest, CheckMarkerBitForTelephoneEvents) { |
1302 char payload_name[RTP_PAYLOAD_NAME_SIZE] = "telephone-event"; | 1302 const char* kDtmfPayloadName = "telephone-event"; |
1303 uint8_t payload_type = 126; | 1303 const uint32_t kPayloadFrequency = 8000; |
1304 ASSERT_EQ(0, | 1304 const uint8_t kPayloadType = 126; |
1305 rtp_sender_->RegisterPayload(payload_name, payload_type, 0, 0, 0)); | 1305 ASSERT_EQ(0, rtp_sender_->RegisterPayload(kDtmfPayloadName, kPayloadType, |
| 1306 kPayloadFrequency, 0, 0)); |
1306 // For Telephone events, payload is not added to the registered payload list, | 1307 // For Telephone events, payload is not added to the registered payload list, |
1307 // it will register only the payload used for audio stream. | 1308 // it will register only the payload used for audio stream. |
1308 // Registering the payload again for audio stream with different payload name. | 1309 // Registering the payload again for audio stream with different payload name. |
1309 const char kPayloadName[] = "payload_name"; | 1310 const char* kPayloadName = "payload_name"; |
1310 ASSERT_EQ( | 1311 ASSERT_EQ(0, rtp_sender_->RegisterPayload(kPayloadName, kPayloadType, |
1311 0, rtp_sender_->RegisterPayload(kPayloadName, payload_type, 8000, 1, 0)); | 1312 kPayloadFrequency, 1, 0)); |
1312 int64_t capture_time_ms = fake_clock_.TimeInMilliseconds(); | 1313 int64_t capture_time_ms = fake_clock_.TimeInMilliseconds(); |
1313 // DTMF event key=9, duration=500 and attenuationdB=10 | 1314 // DTMF event key=9, duration=500 and attenuationdB=10 |
1314 rtp_sender_->SendTelephoneEvent(9, 500, 10); | 1315 rtp_sender_->SendTelephoneEvent(9, 500, 10); |
1315 // During start, it takes the starting timestamp as last sent timestamp. | 1316 // During start, it takes the starting timestamp as last sent timestamp. |
1316 // The duration is calculated as the difference of current and last sent | 1317 // The duration is calculated as the difference of current and last sent |
1317 // timestamp. So for first call it will skip since the duration is zero. | 1318 // timestamp. So for first call it will skip since the duration is zero. |
1318 ASSERT_TRUE(rtp_sender_->SendOutgoingData(kEmptyFrame, payload_type, | 1319 ASSERT_TRUE(rtp_sender_->SendOutgoingData(kEmptyFrame, kPayloadType, |
1319 capture_time_ms, 0, nullptr, 0, | 1320 capture_time_ms, 0, nullptr, 0, |
1320 nullptr, nullptr, nullptr)); | 1321 nullptr, nullptr, nullptr)); |
1321 // DTMF Sample Length is (Frequency/1000) * Duration. | 1322 // DTMF Sample Length is (Frequency/1000) * Duration. |
1322 // So in this case, it is (8000/1000) * 500 = 4000. | 1323 // So in this case, it is (8000/1000) * 500 = 4000. |
1323 // Sending it as two packets. | 1324 // Sending it as two packets. |
1324 ASSERT_TRUE(rtp_sender_->SendOutgoingData( | 1325 ASSERT_TRUE(rtp_sender_->SendOutgoingData( |
1325 kEmptyFrame, payload_type, capture_time_ms + 2000, 0, | 1326 kEmptyFrame, kPayloadType, capture_time_ms + 2000, 0, |
1326 nullptr, 0, nullptr, nullptr, nullptr)); | 1327 nullptr, 0, nullptr, nullptr, nullptr)); |
1327 | 1328 |
1328 // Marker Bit should be set to 1 for first packet. | 1329 // Marker Bit should be set to 1 for first packet. |
1329 EXPECT_TRUE(transport_.last_sent_packet().Marker()); | 1330 EXPECT_TRUE(transport_.last_sent_packet().Marker()); |
1330 | 1331 |
1331 ASSERT_TRUE(rtp_sender_->SendOutgoingData( | 1332 ASSERT_TRUE(rtp_sender_->SendOutgoingData( |
1332 kEmptyFrame, payload_type, capture_time_ms + 4000, 0, | 1333 kEmptyFrame, kPayloadType, capture_time_ms + 4000, 0, |
1333 nullptr, 0, nullptr, nullptr, nullptr)); | 1334 nullptr, 0, nullptr, nullptr, nullptr)); |
1334 // Marker Bit should be set to 0 for rest of the packets. | 1335 // Marker Bit should be set to 0 for rest of the packets. |
1335 EXPECT_FALSE(transport_.last_sent_packet().Marker()); | 1336 EXPECT_FALSE(transport_.last_sent_packet().Marker()); |
1336 } | 1337 } |
1337 | 1338 |
1338 TEST_F(RtpSenderTestWithoutPacer, BytesReportedCorrectly) { | 1339 TEST_F(RtpSenderTestWithoutPacer, BytesReportedCorrectly) { |
1339 const char* kPayloadName = "GENERIC"; | 1340 const char* kPayloadName = "GENERIC"; |
1340 const uint8_t kPayloadType = 127; | 1341 const uint8_t kPayloadType = 127; |
1341 rtp_sender_->SetSSRC(1234); | 1342 rtp_sender_->SetSSRC(1234); |
1342 rtp_sender_->SetRtxSsrc(4321); | 1343 rtp_sender_->SetRtxSsrc(4321); |
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1537 EXPECT_CALL(mock_overhead_observer, OnOverheadChanged(_)).Times(1); | 1538 EXPECT_CALL(mock_overhead_observer, OnOverheadChanged(_)).Times(1); |
1538 rtp_sender_->SetTransportOverhead(28); | 1539 rtp_sender_->SetTransportOverhead(28); |
1539 rtp_sender_->SetTransportOverhead(28); | 1540 rtp_sender_->SetTransportOverhead(28); |
1540 | 1541 |
1541 EXPECT_CALL(mock_overhead_observer, OnOverheadChanged(_)).Times(1); | 1542 EXPECT_CALL(mock_overhead_observer, OnOverheadChanged(_)).Times(1); |
1542 SendGenericPayload(); | 1543 SendGenericPayload(); |
1543 SendGenericPayload(); | 1544 SendGenericPayload(); |
1544 } | 1545 } |
1545 | 1546 |
1546 } // namespace webrtc | 1547 } // namespace webrtc |
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