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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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| 27 #include "webrtc/base/stringencode.h" | 27 #include "webrtc/base/stringencode.h" |
| 28 #include "webrtc/modules/audio_coding/codecs/pcm16b/pcm16b.h" | 28 #include "webrtc/modules/audio_coding/codecs/pcm16b/pcm16b.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/include/module_common_types.h" | 31 #include "webrtc/modules/include/module_common_types.h" |
| 32 #include "webrtc/test/gtest.h" | 32 #include "webrtc/test/gtest.h" |
| 33 #include "webrtc/test/testsupport/fileutils.h" | 33 #include "webrtc/test/testsupport/fileutils.h" |
| 34 #include "webrtc/typedefs.h" | 34 #include "webrtc/typedefs.h" |
| 35 | 35 |
| 36 #ifdef WEBRTC_NETEQ_UNITTEST_BITEXACT | 36 #ifdef WEBRTC_NETEQ_UNITTEST_BITEXACT |
| 37 #include "webrtc/base/protobuf_utils.h" |
| 37 RTC_PUSH_IGNORING_WUNDEF() | 38 RTC_PUSH_IGNORING_WUNDEF() |
| 38 #ifdef WEBRTC_ANDROID_PLATFORM_BUILD | 39 #ifdef WEBRTC_ANDROID_PLATFORM_BUILD |
| 39 #include "external/webrtc/webrtc/modules/audio_coding/neteq/neteq_unittest.pb.h" | 40 #include "external/webrtc/webrtc/modules/audio_coding/neteq/neteq_unittest.pb.h" |
| 40 #else | 41 #else |
| 41 #include "webrtc/modules/audio_coding/neteq/neteq_unittest.pb.h" | 42 #include "webrtc/modules/audio_coding/neteq/neteq_unittest.pb.h" |
| 42 #endif | 43 #endif |
| 43 RTC_POP_IGNORING_WUNDEF() | 44 RTC_POP_IGNORING_WUNDEF() |
| 44 #endif | 45 #endif |
| 45 | 46 |
| 46 DEFINE_bool(gen_ref, false, "Generate reference files."); | 47 DEFINE_bool(gen_ref, false, "Generate reference files."); |
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| 187 ASSERT_EQ(length, fwrite(&test_results, sizeof(T), length, output_fp_)); | 188 ASSERT_EQ(length, fwrite(&test_results, sizeof(T), length, output_fp_)); |
| 188 } | 189 } |
| 189 digest_->Update(&test_results, sizeof(T) * length); | 190 digest_->Update(&test_results, sizeof(T) * length); |
| 190 } | 191 } |
| 191 | 192 |
| 192 void ResultSink::AddResult(const NetEqNetworkStatistics& stats_raw) { | 193 void ResultSink::AddResult(const NetEqNetworkStatistics& stats_raw) { |
| 193 #ifdef WEBRTC_NETEQ_UNITTEST_BITEXACT | 194 #ifdef WEBRTC_NETEQ_UNITTEST_BITEXACT |
| 194 neteq_unittest::NetEqNetworkStatistics stats; | 195 neteq_unittest::NetEqNetworkStatistics stats; |
| 195 Convert(stats_raw, &stats); | 196 Convert(stats_raw, &stats); |
| 196 | 197 |
| 197 std::string stats_string; | 198 ProtoString stats_string; |
| 198 ASSERT_TRUE(stats.SerializeToString(&stats_string)); | 199 ASSERT_TRUE(stats.SerializeToString(&stats_string)); |
| 199 AddMessage(output_fp_, digest_.get(), stats_string); | 200 AddMessage(output_fp_, digest_.get(), stats_string); |
| 200 #else | 201 #else |
| 201 FAIL() << "Writing to reference file requires Proto Buffer."; | 202 FAIL() << "Writing to reference file requires Proto Buffer."; |
| 202 #endif // WEBRTC_NETEQ_UNITTEST_BITEXACT | 203 #endif // WEBRTC_NETEQ_UNITTEST_BITEXACT |
| 203 } | 204 } |
| 204 | 205 |
| 205 void ResultSink::AddResult(const RtcpStatistics& stats_raw) { | 206 void ResultSink::AddResult(const RtcpStatistics& stats_raw) { |
| 206 #ifdef WEBRTC_NETEQ_UNITTEST_BITEXACT | 207 #ifdef WEBRTC_NETEQ_UNITTEST_BITEXACT |
| 207 neteq_unittest::RtcpStatistics stats; | 208 neteq_unittest::RtcpStatistics stats; |
| 208 Convert(stats_raw, &stats); | 209 Convert(stats_raw, &stats); |
| 209 | 210 |
| 210 std::string stats_string; | 211 ProtoString stats_string; |
| 211 ASSERT_TRUE(stats.SerializeToString(&stats_string)); | 212 ASSERT_TRUE(stats.SerializeToString(&stats_string)); |
| 212 AddMessage(output_fp_, digest_.get(), stats_string); | 213 AddMessage(output_fp_, digest_.get(), stats_string); |
| 213 #else | 214 #else |
| 214 FAIL() << "Writing to reference file requires Proto Buffer."; | 215 FAIL() << "Writing to reference file requires Proto Buffer."; |
| 215 #endif // WEBRTC_NETEQ_UNITTEST_BITEXACT | 216 #endif // WEBRTC_NETEQ_UNITTEST_BITEXACT |
| 216 } | 217 } |
| 217 | 218 |
| 218 void ResultSink::VerifyChecksum(const std::string& checksum) { | 219 void ResultSink::VerifyChecksum(const std::string& checksum) { |
| 219 std::vector<char> buffer; | 220 std::vector<char> buffer; |
| 220 buffer.resize(digest_->Size()); | 221 buffer.resize(digest_->Size()); |
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| 1581 if (muted) { | 1582 if (muted) { |
| 1582 EXPECT_TRUE(AudioFramesEqualExceptData(out_frame1, out_frame2)); | 1583 EXPECT_TRUE(AudioFramesEqualExceptData(out_frame1, out_frame2)); |
| 1583 } else { | 1584 } else { |
| 1584 EXPECT_TRUE(AudioFramesEqual(out_frame1, out_frame2)); | 1585 EXPECT_TRUE(AudioFramesEqual(out_frame1, out_frame2)); |
| 1585 } | 1586 } |
| 1586 } | 1587 } |
| 1587 EXPECT_FALSE(muted); | 1588 EXPECT_FALSE(muted); |
| 1588 } | 1589 } |
| 1589 | 1590 |
| 1590 } // namespace webrtc | 1591 } // namespace webrtc |
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