/dports/www/qt5-webengine/qtwebengine-everywhere-src-5.15.2/src/3rdparty/chromium/third_party/webrtc/modules/audio_coding/neteq/ |
H A D | merge_unittest.cc | 53 sync_buffer_(num_channels_, in MergeTest() 56 &sync_buffer_, in MergeTest() 61 merge_(test_sample_rate_hz_, num_channels_, &expand_, &sync_buffer_) { in MergeTest() 73 std::unique_ptr<int16_t[]> temp(new int16_t[sync_buffer_.Size()]); in SetUp() 74 ASSERT_TRUE(input_file_.Read(sync_buffer_.Size(), temp.get())); in SetUp() 75 sync_buffer_.Channel(0).OverwriteAt(temp.get(), sync_buffer_.Size(), 0); in SetUp() 77 sync_buffer_.set_next_index(sync_buffer_.next_index() - in SetUp() 80 ASSERT_GT(sync_buffer_.FutureLength(), 0u); in SetUp() 87 SyncBuffer sync_buffer_; member in webrtc::MergeTest
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H A D | neteq_impl.cc | 428 assert(sync_buffer_.get()); in FlushBuffers() 430 sync_buffer_->Flush(); in FlushBuffers() 431 sync_buffer_->set_next_index(sync_buffer_->next_index() - in FlushBuffers() 476 return sync_buffer_.get(); in sync_buffer_for_test() 926 sync_buffer_->set_next_index(sync_buffer_->next_index() - in GetAudioInternal() 961 sync_buffer_->set_dtmf_index(sync_buffer_->Size()); in GetAudioInternal() 1008 assert(sync_buffer_.get()); in GetDecision() 1066 assert(sync_buffer_.get()); in GetDecision() 1898 if (sync_buffer_->dtmf_index() > sync_buffer_->next_index()) { in DtmfOverdub() 1901 std::min(sync_buffer_->dtmf_index() - sync_buffer_->next_index(), in DtmfOverdub() [all …]
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H A D | expand_unittest.cc | 78 sync_buffer_(num_channels_, in ExpandTest() 81 &sync_buffer_, in ExpandTest() 97 std::unique_ptr<int16_t[]> temp(new int16_t[sync_buffer_.Size()]); in SetUp() 98 ASSERT_TRUE(input_file_.Read(sync_buffer_.Size(), temp.get())); in SetUp() 99 sync_buffer_.Channel(0).OverwriteAt(temp.get(), sync_buffer_.Size(), 0); in SetUp() 107 SyncBuffer sync_buffer_; member in webrtc::ExpandTest
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H A D | merge.cc | 39 sync_buffer_(sync_buffer), in Merge() 152 sync_buffer_->ReplaceAtIndex(*output, old_length, sync_buffer_->next_index()); in Process() 163 *old_length = sync_buffer_->FutureLength(); in GetExpandedSignal() 177 sync_buffer_->InsertZerosAtIndex(length_diff, sync_buffer_->next_index()); in GetExpandedSignal() 190 expanded_.PushBackFromIndex(*sync_buffer_, sync_buffer_->next_index()); in GetExpandedSignal()
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/dports/net-im/tg_owt/tg_owt-d578c76/src/modules/audio_coding/neteq/ |
H A D | merge_unittest.cc | 53 sync_buffer_(num_channels_, in MergeTest() 56 &sync_buffer_, in MergeTest() 61 merge_(test_sample_rate_hz_, num_channels_, &expand_, &sync_buffer_) { in MergeTest() 73 std::unique_ptr<int16_t[]> temp(new int16_t[sync_buffer_.Size()]); in SetUp() 74 ASSERT_TRUE(input_file_.Read(sync_buffer_.Size(), temp.get())); in SetUp() 75 sync_buffer_.Channel(0).OverwriteAt(temp.get(), sync_buffer_.Size(), 0); in SetUp() 77 sync_buffer_.set_next_index(sync_buffer_.next_index() - in SetUp() 80 ASSERT_GT(sync_buffer_.FutureLength(), 0u); in SetUp() 87 SyncBuffer sync_buffer_; member in webrtc::MergeTest
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H A D | neteq_impl.cc | 503 assert(sync_buffer_.get()); in FlushBuffers() 505 sync_buffer_->Flush(); in FlushBuffers() 506 sync_buffer_->set_next_index(sync_buffer_->next_index() - in FlushBuffers() 551 return sync_buffer_.get(); in sync_buffer_for_test() 1017 sync_buffer_->set_next_index(sync_buffer_->next_index() - in GetAudioInternal() 1052 sync_buffer_->set_dtmf_index(sync_buffer_->Size()); in GetAudioInternal() 1099 assert(sync_buffer_.get()); in GetDecision() 1157 assert(sync_buffer_.get()); in GetDecision() 1987 if (sync_buffer_->dtmf_index() > sync_buffer_->next_index()) { in DtmfOverdub() 1990 std::min(sync_buffer_->dtmf_index() - sync_buffer_->next_index(), in DtmfOverdub() [all …]
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H A D | expand_unittest.cc | 78 sync_buffer_(num_channels_, in ExpandTest() 81 &sync_buffer_, in ExpandTest() 97 std::unique_ptr<int16_t[]> temp(new int16_t[sync_buffer_.Size()]); in SetUp() 98 ASSERT_TRUE(input_file_.Read(sync_buffer_.Size(), temp.get())); in SetUp() 99 sync_buffer_.Channel(0).OverwriteAt(temp.get(), sync_buffer_.Size(), 0); in SetUp() 107 SyncBuffer sync_buffer_; member in webrtc::ExpandTest
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H A D | merge.cc | 39 sync_buffer_(sync_buffer), in Merge() 155 sync_buffer_->ReplaceAtIndex(*output, old_length, sync_buffer_->next_index()); in Process() 166 *old_length = sync_buffer_->FutureLength(); in GetExpandedSignal() 180 sync_buffer_->InsertZerosAtIndex(length_diff, sync_buffer_->next_index()); in GetExpandedSignal() 193 expanded_.PushBackFromIndex(*sync_buffer_, sync_buffer_->next_index()); in GetExpandedSignal()
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/dports/www/firefox/firefox-99.0/third_party/libwebrtc/modules/audio_coding/neteq/ |
H A D | merge_unittest.cc | 53 sync_buffer_(num_channels_, in MergeTest() 56 &sync_buffer_, in MergeTest() 61 merge_(test_sample_rate_hz_, num_channels_, &expand_, &sync_buffer_) { in MergeTest() 73 std::unique_ptr<int16_t[]> temp(new int16_t[sync_buffer_.Size()]); in SetUp() 74 ASSERT_TRUE(input_file_.Read(sync_buffer_.Size(), temp.get())); in SetUp() 75 sync_buffer_.Channel(0).OverwriteAt(temp.get(), sync_buffer_.Size(), 0); in SetUp() 77 sync_buffer_.set_next_index(sync_buffer_.next_index() - in SetUp() 80 ASSERT_GT(sync_buffer_.FutureLength(), 0u); in SetUp() 87 SyncBuffer sync_buffer_; member in webrtc::MergeTest
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H A D | neteq_impl.cc | 503 assert(sync_buffer_.get()); in FlushBuffers() 505 sync_buffer_->Flush(); in FlushBuffers() 506 sync_buffer_->set_next_index(sync_buffer_->next_index() - in FlushBuffers() 551 return sync_buffer_.get(); in sync_buffer_for_test() 1001 sync_buffer_->set_next_index(sync_buffer_->next_index() - in GetAudioInternal() 1036 sync_buffer_->set_dtmf_index(sync_buffer_->Size()); in GetAudioInternal() 1083 assert(sync_buffer_.get()); in GetDecision() 1141 assert(sync_buffer_.get()); in GetDecision() 1966 if (sync_buffer_->dtmf_index() > sync_buffer_->next_index()) { in DtmfOverdub() 1969 std::min(sync_buffer_->dtmf_index() - sync_buffer_->next_index(), in DtmfOverdub() [all …]
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H A D | expand_unittest.cc | 78 sync_buffer_(num_channels_, in ExpandTest() 81 &sync_buffer_, in ExpandTest() 97 std::unique_ptr<int16_t[]> temp(new int16_t[sync_buffer_.Size()]); in SetUp() 98 ASSERT_TRUE(input_file_.Read(sync_buffer_.Size(), temp.get())); in SetUp() 99 sync_buffer_.Channel(0).OverwriteAt(temp.get(), sync_buffer_.Size(), 0); in SetUp() 107 SyncBuffer sync_buffer_; member in webrtc::ExpandTest
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/dports/www/chromium-legacy/chromium-88.0.4324.182/third_party/webrtc/modules/audio_coding/neteq/ |
H A D | merge_unittest.cc | 53 sync_buffer_(num_channels_, in MergeTest() 56 &sync_buffer_, in MergeTest() 61 merge_(test_sample_rate_hz_, num_channels_, &expand_, &sync_buffer_) { in MergeTest() 73 std::unique_ptr<int16_t[]> temp(new int16_t[sync_buffer_.Size()]); in SetUp() 74 ASSERT_TRUE(input_file_.Read(sync_buffer_.Size(), temp.get())); in SetUp() 75 sync_buffer_.Channel(0).OverwriteAt(temp.get(), sync_buffer_.Size(), 0); in SetUp() 77 sync_buffer_.set_next_index(sync_buffer_.next_index() - in SetUp() 80 ASSERT_GT(sync_buffer_.FutureLength(), 0u); in SetUp() 87 SyncBuffer sync_buffer_; member in webrtc::MergeTest
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H A D | neteq_impl.cc | 503 assert(sync_buffer_.get()); in FlushBuffers() 505 sync_buffer_->Flush(); in FlushBuffers() 506 sync_buffer_->set_next_index(sync_buffer_->next_index() - in FlushBuffers() 551 return sync_buffer_.get(); in sync_buffer_for_test() 1006 sync_buffer_->set_next_index(sync_buffer_->next_index() - in GetAudioInternal() 1041 sync_buffer_->set_dtmf_index(sync_buffer_->Size()); in GetAudioInternal() 1088 assert(sync_buffer_.get()); in GetDecision() 1146 assert(sync_buffer_.get()); in GetDecision() 1971 if (sync_buffer_->dtmf_index() > sync_buffer_->next_index()) { in DtmfOverdub() 1974 std::min(sync_buffer_->dtmf_index() - sync_buffer_->next_index(), in DtmfOverdub() [all …]
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H A D | expand_unittest.cc | 78 sync_buffer_(num_channels_, in ExpandTest() 81 &sync_buffer_, in ExpandTest() 97 std::unique_ptr<int16_t[]> temp(new int16_t[sync_buffer_.Size()]); in SetUp() 98 ASSERT_TRUE(input_file_.Read(sync_buffer_.Size(), temp.get())); in SetUp() 99 sync_buffer_.Channel(0).OverwriteAt(temp.get(), sync_buffer_.Size(), 0); in SetUp() 107 SyncBuffer sync_buffer_; member in webrtc::ExpandTest
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/dports/lang/spidermonkey60/firefox-60.9.0/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/ |
H A D | neteq_impl.cc | 522 assert(sync_buffer_.get()); in FlushBuffers() 524 sync_buffer_->Flush(); in FlushBuffers() 525 sync_buffer_->set_next_index(sync_buffer_->next_index() - in FlushBuffers() 565 return sync_buffer_.get(); in sync_buffer_for_test() 969 *sync_buffer_, sync_buffer_->Size() - output_size_samples_); in GetAudioInternal() 1014 sync_buffer_->set_next_index(sync_buffer_->next_index() - in GetAudioInternal() 1049 sync_buffer_->set_dtmf_index(sync_buffer_->Size()); in GetAudioInternal() 1093 assert(sync_buffer_.get()); in GetDecision() 1918 if (sync_buffer_->dtmf_index() > sync_buffer_->next_index()) { in DtmfOverdub() 1921 sync_buffer_->dtmf_index() - sync_buffer_->next_index(), in DtmfOverdub() [all …]
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H A D | expand_unittest.cc | 76 sync_buffer_(num_channels_, in ExpandTest() 79 &sync_buffer_, in ExpandTest() 96 std::unique_ptr<int16_t[]> temp(new int16_t[sync_buffer_.Size()]); in SetUp() 97 ASSERT_TRUE(input_file_.Read(sync_buffer_.Size(), temp.get())); in SetUp() 98 sync_buffer_.Channel(0).OverwriteAt(temp.get(), sync_buffer_.Size(), 0); in SetUp() 106 SyncBuffer sync_buffer_; member in webrtc::ExpandTest
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/dports/multimedia/mswebrtc/mswebrtc-1.1.1/webrtc/webrtc/modules/audio_coding/neteq/ |
H A D | neteq_impl.cc | 399 assert(sync_buffer_.get()); in FlushBuffers() 401 sync_buffer_->Flush(); in FlushBuffers() 402 sync_buffer_->set_next_index(sync_buffer_->next_index() - in FlushBuffers() 442 return sync_buffer_.get(); in sync_buffer_for_test() 837 *sync_buffer_, sync_buffer_->Size() - output_size_samples_); in GetAudioInternal() 882 sync_buffer_->set_next_index(sync_buffer_->next_index() - in GetAudioInternal() 917 sync_buffer_->set_dtmf_index(sync_buffer_->Size()); in GetAudioInternal() 950 assert(sync_buffer_.get()); in GetDecision() 1844 if (sync_buffer_->dtmf_index() > sync_buffer_->next_index()) { in DtmfOverdub() 1847 sync_buffer_->dtmf_index() - sync_buffer_->next_index(), in DtmfOverdub() [all …]
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H A D | expand_unittest.cc | 76 sync_buffer_(num_channels_, in ExpandTest() 79 &sync_buffer_, in ExpandTest() 97 input_file_.Read(sync_buffer_.Size(), &sync_buffer_.Channel(0)[0])); in SetUp() 105 SyncBuffer sync_buffer_; member in webrtc::ExpandTest
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/dports/www/firefox-legacy/firefox-52.8.0esr/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/ |
H A D | neteq_impl.cc | 371 assert(sync_buffer_.get()); in FlushBuffers() 373 sync_buffer_->Flush(); in FlushBuffers() 374 sync_buffer_->set_next_index(sync_buffer_->next_index() - in FlushBuffers() 397 return sync_buffer_.get(); in sync_buffer_for_test() 776 *sync_buffer_, sync_buffer_->Size() - output_size_samples_); in GetAudioInternal() 845 sync_buffer_->set_dtmf_index(sync_buffer_->Size()); in GetAudioInternal() 878 assert(sync_buffer_.get()); in GetDecision() 925 assert(sync_buffer_.get()); in GetDecision() 1734 if (sync_buffer_->dtmf_index() > sync_buffer_->next_index()) { in DtmfOverdub() 1737 sync_buffer_->dtmf_index() - sync_buffer_->next_index(), in DtmfOverdub() [all …]
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/dports/mail/thunderbird/thunderbird-91.8.0/third_party/libwebrtc/webrtc/modules/audio_coding/neteq/ |
H A D | neteq_impl.cc | 479 assert(sync_buffer_.get()); in FlushBuffers() 481 sync_buffer_->Flush(); in FlushBuffers() 482 sync_buffer_->set_next_index(sync_buffer_->next_index() - in FlushBuffers() 533 return sync_buffer_.get(); in sync_buffer_for_test() 955 *sync_buffer_, sync_buffer_->Size() - output_size_samples_); in GetAudioInternal() 1000 sync_buffer_->set_next_index(sync_buffer_->next_index() - in GetAudioInternal() 1036 sync_buffer_->set_dtmf_index(sync_buffer_->Size()); in GetAudioInternal() 1080 assert(sync_buffer_.get()); in GetDecision() 1910 if (sync_buffer_->dtmf_index() > sync_buffer_->next_index()) { in DtmfOverdub() 1913 sync_buffer_->dtmf_index() - sync_buffer_->next_index(), in DtmfOverdub() [all …]
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H A D | expand_unittest.cc | 76 sync_buffer_(num_channels_, in ExpandTest() 79 &sync_buffer_, in ExpandTest() 96 std::unique_ptr<int16_t[]> temp(new int16_t[sync_buffer_.Size()]); in SetUp() 97 ASSERT_TRUE(input_file_.Read(sync_buffer_.Size(), temp.get())); in SetUp() 98 sync_buffer_.Channel(0).OverwriteAt(temp.get(), sync_buffer_.Size(), 0); in SetUp() 106 SyncBuffer sync_buffer_; member in webrtc::ExpandTest
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/dports/www/firefox-esr/firefox-91.8.0/third_party/libwebrtc/webrtc/modules/audio_coding/neteq/ |
H A D | neteq_impl.cc | 479 assert(sync_buffer_.get()); in FlushBuffers() 481 sync_buffer_->Flush(); in FlushBuffers() 482 sync_buffer_->set_next_index(sync_buffer_->next_index() - in FlushBuffers() 533 return sync_buffer_.get(); in sync_buffer_for_test() 955 *sync_buffer_, sync_buffer_->Size() - output_size_samples_); in GetAudioInternal() 1000 sync_buffer_->set_next_index(sync_buffer_->next_index() - in GetAudioInternal() 1036 sync_buffer_->set_dtmf_index(sync_buffer_->Size()); in GetAudioInternal() 1080 assert(sync_buffer_.get()); in GetDecision() 1910 if (sync_buffer_->dtmf_index() > sync_buffer_->next_index()) { in DtmfOverdub() 1913 sync_buffer_->dtmf_index() - sync_buffer_->next_index(), in DtmfOverdub() [all …]
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H A D | expand_unittest.cc | 76 sync_buffer_(num_channels_, in ExpandTest() 79 &sync_buffer_, in ExpandTest() 96 std::unique_ptr<int16_t[]> temp(new int16_t[sync_buffer_.Size()]); in SetUp() 97 ASSERT_TRUE(input_file_.Read(sync_buffer_.Size(), temp.get())); in SetUp() 98 sync_buffer_.Channel(0).OverwriteAt(temp.get(), sync_buffer_.Size(), 0); in SetUp() 106 SyncBuffer sync_buffer_; member in webrtc::ExpandTest
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/dports/lang/spidermonkey78/firefox-78.9.0/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/ |
H A D | neteq_impl.cc | 479 assert(sync_buffer_.get()); in FlushBuffers() 481 sync_buffer_->Flush(); in FlushBuffers() 482 sync_buffer_->set_next_index(sync_buffer_->next_index() - in FlushBuffers() 533 return sync_buffer_.get(); in sync_buffer_for_test() 955 *sync_buffer_, sync_buffer_->Size() - output_size_samples_); in GetAudioInternal() 1000 sync_buffer_->set_next_index(sync_buffer_->next_index() - in GetAudioInternal() 1036 sync_buffer_->set_dtmf_index(sync_buffer_->Size()); in GetAudioInternal() 1080 assert(sync_buffer_.get()); in GetDecision() 1910 if (sync_buffer_->dtmf_index() > sync_buffer_->next_index()) { in DtmfOverdub() 1913 sync_buffer_->dtmf_index() - sync_buffer_->next_index(), in DtmfOverdub() [all …]
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H A D | expand_unittest.cc | 76 sync_buffer_(num_channels_, in ExpandTest() 79 &sync_buffer_, in ExpandTest() 96 std::unique_ptr<int16_t[]> temp(new int16_t[sync_buffer_.Size()]); in SetUp() 97 ASSERT_TRUE(input_file_.Read(sync_buffer_.Size(), temp.get())); in SetUp() 98 sync_buffer_.Channel(0).OverwriteAt(temp.get(), sync_buffer_.Size(), 0); in SetUp() 106 SyncBuffer sync_buffer_; member in webrtc::ExpandTest
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