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Side by Side Diff: webrtc/modules/audio_coding/acm2/acm_receiver_unittest_oldapi.cc

Issue 1863993002: Move setting of AudioFrame::timestamp_ into NetEq (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@neteq-empty-playout-ts
Patch Set: Created 4 years, 8 months ago
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1 /* 1 /*
2 * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2013 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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319 output_sample_rate_hz > 16000 ? AudioFrame::kVadActive 319 output_sample_rate_hz > 16000 ? AudioFrame::kVadActive
320 : AudioFrame::kVadPassive; 320 : AudioFrame::kVadPassive;
321 321
322 // Expect the first output timestamp to be 5*fs/8000 samples before the 322 // Expect the first output timestamp to be 5*fs/8000 samples before the
323 // first inserted timestamp (because of NetEq's look-ahead). (This value is 323 // first inserted timestamp (because of NetEq's look-ahead). (This value is
324 // defined in Expand::overlap_length_.) 324 // defined in Expand::overlap_length_.)
325 uint32_t expected_output_ts = last_packet_send_timestamp_ - 325 uint32_t expected_output_ts = last_packet_send_timestamp_ -
326 rtc::CheckedDivExact(5 * output_sample_rate_hz, 8000); 326 rtc::CheckedDivExact(5 * output_sample_rate_hz, 8000);
327 327
328 AudioFrame frame; 328 AudioFrame frame;
329 EXPECT_EQ(0, receiver_->GetAudio(output_sample_rate_hz, &frame));
330 // Expect timestamp = 0 before first packet is inserted.
331 EXPECT_EQ(0u, frame.timestamp_);
329 for (int i = 0; i < 5; ++i) { 332 for (int i = 0; i < 5; ++i) {
330 InsertOnePacketOfSilence(codec.id); 333 InsertOnePacketOfSilence(codec.id);
331 for (int k = 0; k < num_10ms_frames; ++k) { 334 for (int k = 0; k < num_10ms_frames; ++k) {
332 EXPECT_EQ(0, receiver_->GetAudio(output_sample_rate_hz, &frame)); 335 EXPECT_EQ(0, receiver_->GetAudio(output_sample_rate_hz, &frame));
333 EXPECT_EQ(expected_output_ts, frame.timestamp_); 336 EXPECT_EQ(expected_output_ts, frame.timestamp_);
334 expected_output_ts += 10 * samples_per_ms; 337 expected_output_ts += 10 * samples_per_ms;
335 EXPECT_EQ(10 * samples_per_ms, frame.samples_per_channel_); 338 EXPECT_EQ(10 * samples_per_ms, frame.samples_per_channel_);
336 EXPECT_EQ(output_sample_rate_hz, frame.sample_rate_hz_); 339 EXPECT_EQ(output_sample_rate_hz, frame.sample_rate_hz_);
337 EXPECT_EQ(output_channels, frame.num_channels_); 340 EXPECT_EQ(output_channels, frame.num_channels_);
338 EXPECT_EQ(AudioFrame::kNormalSpeech, frame.speech_type_); 341 EXPECT_EQ(AudioFrame::kNormalSpeech, frame.speech_type_);
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487 receiver_->last_packet_sample_rate_hz()); 490 receiver_->last_packet_sample_rate_hz());
488 EXPECT_EQ(0, receiver_->LastAudioCodec(&codec)); 491 EXPECT_EQ(0, receiver_->LastAudioCodec(&codec));
489 EXPECT_TRUE(CodecsEqual(c.inst, codec)); 492 EXPECT_TRUE(CodecsEqual(c.inst, codec));
490 } 493 }
491 } 494 }
492 #endif 495 #endif
493 496
494 } // namespace acm2 497 } // namespace acm2
495 498
496 } // namespace webrtc 499 } // namespace webrtc
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