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Issue 2119393002: Fixed time moving backwards in the AudioCodingModule. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Updated Dtx tests that moved the input timestamp backwards to not do that. Created 4 years, 5 months ago
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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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94 in_file.FastForward(50); 94 in_file.FastForward(50);
95 95
96 PCMFile out_file; 96 PCMFile out_file;
97 if (append) { 97 if (append) {
98 out_file.Open(out_filename, kOutputFreqHz, "ab"); 98 out_file.Open(out_filename, kOutputFreqHz, "ab");
99 } else { 99 } else {
100 out_file.Open(out_filename, kOutputFreqHz, "wb"); 100 out_file.Open(out_filename, kOutputFreqHz, "wb");
101 } 101 }
102 102
103 uint16_t frame_size_samples = in_file.PayloadLength10Ms(); 103 uint16_t frame_size_samples = in_file.PayloadLength10Ms();
104 uint32_t time_stamp = 0x12345678;
105 AudioFrame audio_frame; 104 AudioFrame audio_frame;
106 while (!in_file.EndOfFile()) { 105 while (!in_file.EndOfFile()) {
107 in_file.Read10MsData(audio_frame); 106 in_file.Read10MsData(audio_frame);
108 audio_frame.timestamp_ = time_stamp; 107 audio_frame.timestamp_ = time_stamp_;
109 time_stamp += frame_size_samples; 108 time_stamp_ += frame_size_samples;
110 EXPECT_GE(acm_send_->Add10MsData(audio_frame), 0); 109 EXPECT_GE(acm_send_->Add10MsData(audio_frame), 0);
111 bool muted; 110 bool muted;
112 acm_receive_->PlayoutData10Ms(kOutputFreqHz, &audio_frame, &muted); 111 acm_receive_->PlayoutData10Ms(kOutputFreqHz, &audio_frame, &muted);
113 ASSERT_FALSE(muted); 112 ASSERT_FALSE(muted);
114 out_file.Write10MsData(audio_frame); 113 out_file.Write10MsData(audio_frame);
115 } 114 }
116 115
117 in_file.Close(); 116 in_file.Close();
118 out_file.Close(); 117 out_file.Close();
119 118
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269 268
270 EXPECT_EQ(0, acm_send_->EnableOpusDtx()); 269 EXPECT_EQ(0, acm_send_->EnableOpusDtx());
271 270
272 expects[kEmptyFrame] = 1; 271 expects[kEmptyFrame] = 1;
273 Run(webrtc::test::ResourcePath("audio_coding/teststereo32kHz", "pcm"), 272 Run(webrtc::test::ResourcePath("audio_coding/teststereo32kHz", "pcm"),
274 32000, 2, out_filename, true, expects); 273 32000, 2, out_filename, true, expects);
275 #endif 274 #endif
276 } 275 }
277 276
278 } // namespace webrtc 277 } // namespace webrtc
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