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Side by Side Diff: webrtc/modules/audio_coding/neteq/neteq_external_decoder_unittest.cc

Issue 1172163004: Reformat existing code. There should be no functional effects. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc@master
Patch Set: Resync Created 5 years, 6 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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67 // Method to get the number of calls to the Decode() method of the external 67 // Method to get the number of calls to the Decode() method of the external
68 // decoder. 68 // decoder.
69 virtual int NumExpectedDecodeCalls(int num_loops) = 0; 69 virtual int NumExpectedDecodeCalls(int num_loops) = 0;
70 70
71 // Method to generate packets and return the send time of the packet. 71 // Method to generate packets and return the send time of the packet.
72 int GetNewPacket() { 72 int GetNewPacket() {
73 if (!input_file_->Read(frame_size_samples_, input_)) { 73 if (!input_file_->Read(frame_size_samples_, input_)) {
74 return -1; 74 return -1;
75 } 75 }
76 payload_size_bytes_ = WebRtcPcm16b_Encode(input_, frame_size_samples_, 76 payload_size_bytes_ = WebRtcPcm16b_Encode(input_, frame_size_samples_,
77 encoded_);; 77 encoded_);
78 78
79 int next_send_time = rtp_generator_->GetRtpHeader( 79 int next_send_time = rtp_generator_->GetRtpHeader(
80 kPayloadType, frame_size_samples_, &rtp_header_); 80 kPayloadType, frame_size_samples_, &rtp_header_);
81 return next_send_time; 81 return next_send_time;
82 } 82 }
83 83
84 // Method to decide packet losses. 84 // Method to decide packet losses.
85 virtual bool Lost() { return false; } 85 virtual bool Lost() { return false; }
86 86
87 // Method to calculate packet arrival time. 87 // Method to calculate packet arrival time.
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456 kStartSeqeunceNumber, 456 kStartSeqeunceNumber,
457 kStartTimestamp, 457 kStartTimestamp,
458 kJumpFromTimestamp, 458 kJumpFromTimestamp,
459 kJumpToTimestamp)); 459 kJumpToTimestamp));
460 460
461 RunTest(130); // Run 130 laps @ 10 ms each in the test loop. 461 RunTest(130); // Run 130 laps @ 10 ms each in the test loop.
462 EXPECT_EQ(kRecovered, test_state_); 462 EXPECT_EQ(kRecovered, test_state_);
463 } 463 }
464 464
465 } // namespace webrtc 465 } // namespace webrtc
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