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1 /* | 1 /* |
2 * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2011 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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342 if (rtp_header.header.payloadType != 104) | 342 if (rtp_header.header.payloadType != 104) |
343 #endif | 343 #endif |
344 ASSERT_EQ(0, neteq_->InsertPacket( | 344 ASSERT_EQ(0, neteq_->InsertPacket( |
345 rtp_header, | 345 rtp_header, |
346 rtc::ArrayView<const uint8_t>( | 346 rtc::ArrayView<const uint8_t>( |
347 packet_->payload(), packet_->payload_length_bytes()), | 347 packet_->payload(), packet_->payload_length_bytes()), |
348 static_cast<uint32_t>(packet_->time_ms() * | 348 static_cast<uint32_t>(packet_->time_ms() * |
349 (output_sample_rate_ / 1000)))); | 349 (output_sample_rate_ / 1000)))); |
350 } | 350 } |
351 // Get next packet. | 351 // Get next packet. |
352 packet_.reset(rtp_source_->NextPacket()); | 352 packet_ = rtp_source_->NextPacket(); |
353 } | 353 } |
354 | 354 |
355 // Get audio from NetEq. | 355 // Get audio from NetEq. |
356 bool muted; | 356 bool muted; |
357 ASSERT_EQ(0, neteq_->GetAudio(&out_frame_, &muted)); | 357 ASSERT_EQ(0, neteq_->GetAudio(&out_frame_, &muted)); |
358 ASSERT_FALSE(muted); | 358 ASSERT_FALSE(muted); |
359 ASSERT_TRUE((out_frame_.samples_per_channel_ == kBlockSize8kHz) || | 359 ASSERT_TRUE((out_frame_.samples_per_channel_ == kBlockSize8kHz) || |
360 (out_frame_.samples_per_channel_ == kBlockSize16kHz) || | 360 (out_frame_.samples_per_channel_ == kBlockSize16kHz) || |
361 (out_frame_.samples_per_channel_ == kBlockSize32kHz) || | 361 (out_frame_.samples_per_channel_ == kBlockSize32kHz) || |
362 (out_frame_.samples_per_channel_ == kBlockSize48kHz)); | 362 (out_frame_.samples_per_channel_ == kBlockSize48kHz)); |
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380 ResultSink output(ref_out_file); | 380 ResultSink output(ref_out_file); |
381 | 381 |
382 std::string stat_out_file = | 382 std::string stat_out_file = |
383 gen_ref ? webrtc::test::OutputPath() + "neteq_network_stats.dat" : ""; | 383 gen_ref ? webrtc::test::OutputPath() + "neteq_network_stats.dat" : ""; |
384 ResultSink network_stats(stat_out_file); | 384 ResultSink network_stats(stat_out_file); |
385 | 385 |
386 std::string rtcp_out_file = | 386 std::string rtcp_out_file = |
387 gen_ref ? webrtc::test::OutputPath() + "neteq_rtcp_stats.dat" : ""; | 387 gen_ref ? webrtc::test::OutputPath() + "neteq_rtcp_stats.dat" : ""; |
388 ResultSink rtcp_stats(rtcp_out_file); | 388 ResultSink rtcp_stats(rtcp_out_file); |
389 | 389 |
390 packet_.reset(rtp_source_->NextPacket()); | 390 packet_ = rtp_source_->NextPacket(); |
391 int i = 0; | 391 int i = 0; |
392 while (packet_) { | 392 while (packet_) { |
393 std::ostringstream ss; | 393 std::ostringstream ss; |
394 ss << "Lap number " << i++ << " in DecodeAndCompare while loop"; | 394 ss << "Lap number " << i++ << " in DecodeAndCompare while loop"; |
395 SCOPED_TRACE(ss.str()); // Print out the parameter values on failure. | 395 SCOPED_TRACE(ss.str()); // Print out the parameter values on failure. |
396 ASSERT_NO_FATAL_FAILURE(Process()); | 396 ASSERT_NO_FATAL_FAILURE(Process()); |
397 ASSERT_NO_FATAL_FAILURE(output.AddResult( | 397 ASSERT_NO_FATAL_FAILURE(output.AddResult( |
398 out_frame_.data_, out_frame_.samples_per_channel_)); | 398 out_frame_.data_, out_frame_.samples_per_channel_)); |
399 | 399 |
400 // Query the network statistics API once per second | 400 // Query the network statistics API once per second |
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1769 if (muted) { | 1769 if (muted) { |
1770 EXPECT_TRUE(AudioFramesEqualExceptData(out_frame1, out_frame2)); | 1770 EXPECT_TRUE(AudioFramesEqualExceptData(out_frame1, out_frame2)); |
1771 } else { | 1771 } else { |
1772 EXPECT_TRUE(AudioFramesEqual(out_frame1, out_frame2)); | 1772 EXPECT_TRUE(AudioFramesEqual(out_frame1, out_frame2)); |
1773 } | 1773 } |
1774 } | 1774 } |
1775 EXPECT_FALSE(muted); | 1775 EXPECT_FALSE(muted); |
1776 } | 1776 } |
1777 | 1777 |
1778 } // namespace webrtc | 1778 } // namespace webrtc |
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