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Issue 1438663003: modules/audio_coding: Remove some codec include dirs (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Rebase again Created 5 years, 1 month ago
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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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21 #include <algorithm> 21 #include <algorithm>
22 #include <set> 22 #include <set>
23 #include <string> 23 #include <string>
24 #include <vector> 24 #include <vector>
25 25
26 #include "gflags/gflags.h" 26 #include "gflags/gflags.h"
27 #include "testing/gtest/include/gtest/gtest.h" 27 #include "testing/gtest/include/gtest/gtest.h"
28 #include "webrtc/base/scoped_ptr.h" 28 #include "webrtc/base/scoped_ptr.h"
29 #include "webrtc/modules/audio_coding/neteq/tools/audio_loop.h" 29 #include "webrtc/modules/audio_coding/neteq/tools/audio_loop.h"
30 #include "webrtc/modules/audio_coding/neteq/tools/rtp_file_source.h" 30 #include "webrtc/modules/audio_coding/neteq/tools/rtp_file_source.h"
31 #include "webrtc/modules/audio_coding/codecs/pcm16b/include/pcm16b.h" 31 #include "webrtc/modules/audio_coding/codecs/pcm16b/pcm16b.h"
32 #include "webrtc/test/testsupport/fileutils.h" 32 #include "webrtc/test/testsupport/fileutils.h"
33 #include "webrtc/test/testsupport/gtest_disable.h" 33 #include "webrtc/test/testsupport/gtest_disable.h"
34 #include "webrtc/typedefs.h" 34 #include "webrtc/typedefs.h"
35 35
36 DEFINE_bool(gen_ref, false, "Generate reference files."); 36 DEFINE_bool(gen_ref, false, "Generate reference files.");
37 37
38 namespace webrtc { 38 namespace webrtc {
39 39
40 static bool IsAllZero(const int16_t* buf, size_t buf_length) { 40 static bool IsAllZero(const int16_t* buf, size_t buf_length) {
41 bool all_zero = true; 41 bool all_zero = true;
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1534 // Pull audio once. 1534 // Pull audio once.
1535 ASSERT_EQ(0, neteq_->GetAudio(kMaxBlockSize, out_data_, &out_len, 1535 ASSERT_EQ(0, neteq_->GetAudio(kMaxBlockSize, out_data_, &out_len,
1536 &num_channels, &type)); 1536 &num_channels, &type));
1537 ASSERT_EQ(kBlockSize16kHz, out_len); 1537 ASSERT_EQ(kBlockSize16kHz, out_len);
1538 } 1538 }
1539 // Verify speech output. 1539 // Verify speech output.
1540 EXPECT_EQ(kOutputNormal, type); 1540 EXPECT_EQ(kOutputNormal, type);
1541 } 1541 }
1542 1542
1543 } // namespace webrtc 1543 } // namespace webrtc
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