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Side by Side Diff: webrtc/audio/audio_receive_stream_unittest.cc

Issue 2005313003: Propagate probing cluster id to SendTimeHistory. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Feedback fixes. Created 4 years, 6 months ago
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1 /* 1 /*
2 * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2015 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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262 std::vector<uint8_t> rtp_packet = CreateRtpHeaderWithOneByteExtension( 262 std::vector<uint8_t> rtp_packet = CreateRtpHeaderWithOneByteExtension(
263 kTransportSequenceNumberId, kTransportSequenceNumberValue, 2); 263 kTransportSequenceNumberId, kTransportSequenceNumberValue, 2);
264 PacketTime packet_time(5678000, 0); 264 PacketTime packet_time(5678000, 0);
265 const size_t kExpectedHeaderLength = 20; 265 const size_t kExpectedHeaderLength = 20;
266 RTPHeaderExtension expected_extension; 266 RTPHeaderExtension expected_extension;
267 expected_extension.hasTransportSequenceNumber = true; 267 expected_extension.hasTransportSequenceNumber = true;
268 expected_extension.transportSequenceNumber = kTransportSequenceNumberValue; 268 expected_extension.transportSequenceNumber = kTransportSequenceNumberValue;
269 EXPECT_CALL(*helper.remote_bitrate_estimator(), 269 EXPECT_CALL(*helper.remote_bitrate_estimator(),
270 IncomingPacket(packet_time.timestamp / 1000, 270 IncomingPacket(packet_time.timestamp / 1000,
271 rtp_packet.size() - kExpectedHeaderLength, 271 rtp_packet.size() - kExpectedHeaderLength,
272 VerifyHeaderExtension(expected_extension), false)) 272 VerifyHeaderExtension(expected_extension),
273 PacketInfo::kNotAProbe))
273 .Times(1); 274 .Times(1);
274 EXPECT_CALL(*helper.channel_proxy(), 275 EXPECT_CALL(*helper.channel_proxy(),
275 ReceivedRTPPacket(&rtp_packet[0], 276 ReceivedRTPPacket(&rtp_packet[0],
276 rtp_packet.size(), 277 rtp_packet.size(),
277 _)) 278 _))
278 .WillOnce(Return(true)); 279 .WillOnce(Return(true));
279 EXPECT_TRUE( 280 EXPECT_TRUE(
280 recv_stream.DeliverRtp(&rtp_packet[0], rtp_packet.size(), packet_time)); 281 recv_stream.DeliverRtp(&rtp_packet[0], rtp_packet.size(), packet_time));
281 } 282 }
282 283
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331 EXPECT_EQ(kAudioDecodeStats.calls_to_neteq, stats.decoding_calls_to_neteq); 332 EXPECT_EQ(kAudioDecodeStats.calls_to_neteq, stats.decoding_calls_to_neteq);
332 EXPECT_EQ(kAudioDecodeStats.decoded_normal, stats.decoding_normal); 333 EXPECT_EQ(kAudioDecodeStats.decoded_normal, stats.decoding_normal);
333 EXPECT_EQ(kAudioDecodeStats.decoded_plc, stats.decoding_plc); 334 EXPECT_EQ(kAudioDecodeStats.decoded_plc, stats.decoding_plc);
334 EXPECT_EQ(kAudioDecodeStats.decoded_cng, stats.decoding_cng); 335 EXPECT_EQ(kAudioDecodeStats.decoded_cng, stats.decoding_cng);
335 EXPECT_EQ(kAudioDecodeStats.decoded_plc_cng, stats.decoding_plc_cng); 336 EXPECT_EQ(kAudioDecodeStats.decoded_plc_cng, stats.decoding_plc_cng);
336 EXPECT_EQ(kCallStats.capture_start_ntp_time_ms_, 337 EXPECT_EQ(kCallStats.capture_start_ntp_time_ms_,
337 stats.capture_start_ntp_time_ms); 338 stats.capture_start_ntp_time_ms);
338 } 339 }
339 } // namespace test 340 } // namespace test
340 } // namespace webrtc 341 } // namespace webrtc
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