/dports/security/vault/vault-1.8.2/vendor/github.com/apache/arrow/cpp/src/arrow/util/ |
H A D | compression_lz4.cc | 96 auto dst_capacity = static_cast<size_t>(output_len); in Decompress() local 106 static_cast<int64_t>(dst_capacity), in Decompress() 164 auto dst_capacity = static_cast<size_t>(output_len); in Compress() local 168 BEGIN_COMPRESS(dst, dst_capacity, (CompressResult{0, 0})); in Compress() 170 if (dst_capacity < LZ4F_compressBound(src_size, &prefs_)) { in Compress() 186 auto dst_capacity = static_cast<size_t>(output_len); in Flush() local 190 BEGIN_COMPRESS(dst, dst_capacity, (FlushResult{0, true})); in Flush() 192 if (dst_capacity < LZ4F_compressBound(0, &prefs_)) { in Flush() 208 auto dst_capacity = static_cast<size_t>(output_len); in End() local 212 BEGIN_COMPRESS(dst, dst_capacity, (EndResult{0, true})); in End() [all …]
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/dports/www/grafana8/grafana-8.3.6/vendor/github.com/apache/arrow/cpp/src/arrow/util/ |
H A D | compression_lz4.cc | 99 auto dst_capacity = static_cast<size_t>(output_len); in Decompress() local 109 static_cast<int64_t>(dst_capacity), in Decompress() 167 auto dst_capacity = static_cast<size_t>(output_len); in Compress() local 171 BEGIN_COMPRESS(dst, dst_capacity, (CompressResult{0, 0})); in Compress() 173 if (dst_capacity < LZ4F_compressBound(src_size, &prefs_)) { in Compress() 189 auto dst_capacity = static_cast<size_t>(output_len); in Flush() local 193 BEGIN_COMPRESS(dst, dst_capacity, (FlushResult{0, true})); in Flush() 195 if (dst_capacity < LZ4F_compressBound(0, &prefs_)) { in Flush() 211 auto dst_capacity = static_cast<size_t>(output_len); in End() local 215 BEGIN_COMPRESS(dst, dst_capacity, (EndResult{0, true})); in End() [all …]
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/dports/databases/arrow/apache-arrow-6.0.1/cpp/src/arrow/util/ |
H A D | compression_lz4.cc | 99 auto dst_capacity = static_cast<size_t>(output_len); in Decompress() local 109 static_cast<int64_t>(dst_capacity), in Decompress() 167 auto dst_capacity = static_cast<size_t>(output_len); in Compress() local 171 BEGIN_COMPRESS(dst, dst_capacity, (CompressResult{0, 0})); in Compress() 173 if (dst_capacity < LZ4F_compressBound(src_size, &prefs_)) { in Compress() 189 auto dst_capacity = static_cast<size_t>(output_len); in Flush() local 193 BEGIN_COMPRESS(dst, dst_capacity, (FlushResult{0, true})); in Flush() 195 if (dst_capacity < LZ4F_compressBound(0, &prefs_)) { in Flush() 211 auto dst_capacity = static_cast<size_t>(output_len); in End() local 215 BEGIN_COMPRESS(dst, dst_capacity, (EndResult{0, true})); in End() [all …]
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/dports/multimedia/mswebrtc/mswebrtc-1.1.1/webrtc/webrtc/common_audio/ |
H A D | audio_converter.cc | 33 size_t dst_capacity) override { in Convert() argument 34 CheckSizes(src_size, dst_capacity); in Convert() 50 size_t dst_capacity) override { in Convert() argument 51 CheckSizes(src_size, dst_capacity); in Convert() 69 size_t dst_capacity) override { in Convert() argument 70 CheckSizes(src_size, dst_capacity); in Convert() 93 size_t dst_capacity) override { in Convert() argument 94 CheckSizes(src_size, dst_capacity); in Convert() 118 size_t dst_capacity) override { in Convert() argument 130 buffers_.back()->size(), dst, dst_capacity); in Convert() [all …]
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/dports/www/firefox-legacy/firefox-52.8.0esr/media/webrtc/trunk/webrtc/common_audio/ |
H A D | audio_converter.cc | 33 size_t dst_capacity) override { 34 CheckSizes(src_size, dst_capacity); 50 size_t dst_capacity) override { 51 CheckSizes(src_size, dst_capacity); 69 size_t dst_capacity) override { 70 CheckSizes(src_size, dst_capacity); 93 size_t dst_capacity) override { 94 CheckSizes(src_size, dst_capacity); 118 size_t dst_capacity) override { 130 buffers_.back()->size(), dst, dst_capacity); [all …]
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/dports/audio/webrtc-audio-processing0/webrtc-audio-processing-0.3.1/webrtc/common_audio/ |
H A D | audio_converter.cc | 33 size_t dst_capacity) override { in Convert() argument 34 CheckSizes(src_size, dst_capacity); in Convert() 50 size_t dst_capacity) override { in Convert() argument 51 CheckSizes(src_size, dst_capacity); in Convert() 69 size_t dst_capacity) override { in Convert() argument 70 CheckSizes(src_size, dst_capacity); in Convert() 93 size_t dst_capacity) override { in Convert() argument 94 CheckSizes(src_size, dst_capacity); in Convert() 118 size_t dst_capacity) override { in Convert() argument 130 buffers_.back()->size(), dst, dst_capacity); in Convert() [all …]
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/dports/www/qt5-webengine/qtwebengine-everywhere-src-5.15.2/src/3rdparty/chromium/third_party/webrtc/common_audio/ |
H A D | audio_converter.cc | 37 size_t dst_capacity) override { in Convert() argument 38 CheckSizes(src_size, dst_capacity); in Convert() 58 size_t dst_capacity) override { in Convert() argument 59 CheckSizes(src_size, dst_capacity); in Convert() 80 size_t dst_capacity) override { in Convert() argument 81 CheckSizes(src_size, dst_capacity); in Convert() 109 size_t dst_capacity) override { in Convert() argument 110 CheckSizes(src_size, dst_capacity); in Convert() 138 size_t dst_capacity) override { in Convert() argument 148 buffers_.back()->size(), dst, dst_capacity); in Convert() [all …]
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/dports/net-im/tg_owt/tg_owt-d578c76/src/common_audio/ |
H A D | audio_converter.cc | 37 size_t dst_capacity) override { in Convert() argument 38 CheckSizes(src_size, dst_capacity); in Convert() 58 size_t dst_capacity) override { in Convert() argument 59 CheckSizes(src_size, dst_capacity); in Convert() 80 size_t dst_capacity) override { in Convert() argument 81 CheckSizes(src_size, dst_capacity); in Convert() 109 size_t dst_capacity) override { in Convert() argument 110 CheckSizes(src_size, dst_capacity); in Convert() 138 size_t dst_capacity) override { in Convert() argument 148 buffers_.back()->size(), dst, dst_capacity); in Convert() [all …]
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/dports/www/chromium-legacy/chromium-88.0.4324.182/third_party/webrtc/common_audio/ |
H A D | audio_converter.cc | 37 size_t dst_capacity) override { in Convert() argument 38 CheckSizes(src_size, dst_capacity); in Convert() 58 size_t dst_capacity) override { in Convert() argument 59 CheckSizes(src_size, dst_capacity); in Convert() 80 size_t dst_capacity) override { in Convert() argument 81 CheckSizes(src_size, dst_capacity); in Convert() 109 size_t dst_capacity) override { in Convert() argument 110 CheckSizes(src_size, dst_capacity); in Convert() 138 size_t dst_capacity) override { in Convert() argument 148 buffers_.back()->size(), dst, dst_capacity); in Convert() [all …]
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/dports/www/firefox/firefox-99.0/third_party/libwebrtc/common_audio/ |
H A D | audio_converter.cc | 37 size_t dst_capacity) override { in Convert() argument 38 CheckSizes(src_size, dst_capacity); in Convert() 58 size_t dst_capacity) override { in Convert() argument 59 CheckSizes(src_size, dst_capacity); in Convert() 80 size_t dst_capacity) override { in Convert() argument 81 CheckSizes(src_size, dst_capacity); in Convert() 109 size_t dst_capacity) override { in Convert() argument 110 CheckSizes(src_size, dst_capacity); in Convert() 138 size_t dst_capacity) override { in Convert() argument 148 buffers_.back()->size(), dst, dst_capacity); in Convert() [all …]
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/dports/audio/webrtc-audio-processing/webrtc-audio-processing-1.0/webrtc/common_audio/ |
H A D | audio_converter.cc | 37 size_t dst_capacity) override { in Convert() argument 38 CheckSizes(src_size, dst_capacity); in Convert() 58 size_t dst_capacity) override { in Convert() argument 59 CheckSizes(src_size, dst_capacity); in Convert() 80 size_t dst_capacity) override { in Convert() argument 81 CheckSizes(src_size, dst_capacity); in Convert() 109 size_t dst_capacity) override { in Convert() argument 110 CheckSizes(src_size, dst_capacity); in Convert() 138 size_t dst_capacity) override { in Convert() argument 148 buffers_.back()->size(), dst, dst_capacity); in Convert() [all …]
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/dports/mail/thunderbird/thunderbird-91.8.0/third_party/libwebrtc/webrtc/common_audio/ |
H A D | audio_converter.cc | 35 size_t dst_capacity) override { in Convert() argument 36 CheckSizes(src_size, dst_capacity); in Convert() 52 size_t dst_capacity) override { in Convert() argument 53 CheckSizes(src_size, dst_capacity); in Convert() 71 size_t dst_capacity) override { in Convert() argument 72 CheckSizes(src_size, dst_capacity); in Convert() 96 size_t dst_capacity) override { in Convert() argument 97 CheckSizes(src_size, dst_capacity); in Convert() 123 size_t dst_capacity) override { in Convert() argument 135 buffers_.back()->size(), dst, dst_capacity); in Convert() [all …]
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/dports/net-im/telegram-desktop/tdesktop-3.2.5-full/Telegram/ThirdParty/libtgvoip/webrtc_dsp/common_audio/ |
H A D | audio_converter.cc | 39 size_t dst_capacity) override { in Convert() argument 40 CheckSizes(src_size, dst_capacity); in Convert() 60 size_t dst_capacity) override { in Convert() argument 61 CheckSizes(src_size, dst_capacity); in Convert() 82 size_t dst_capacity) override { in Convert() argument 83 CheckSizes(src_size, dst_capacity); in Convert() 111 size_t dst_capacity) override { in Convert() argument 112 CheckSizes(src_size, dst_capacity); in Convert() 140 size_t dst_capacity) override { in Convert() argument 150 buffers_.back()->size(), dst, dst_capacity); in Convert() [all …]
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/dports/lang/spidermonkey60/firefox-60.9.0/media/webrtc/trunk/webrtc/common_audio/ |
H A D | audio_converter.cc | 35 size_t dst_capacity) override { in Convert() argument 36 CheckSizes(src_size, dst_capacity); in Convert() 52 size_t dst_capacity) override { in Convert() argument 53 CheckSizes(src_size, dst_capacity); in Convert() 71 size_t dst_capacity) override { in Convert() argument 72 CheckSizes(src_size, dst_capacity); in Convert() 96 size_t dst_capacity) override { in Convert() argument 97 CheckSizes(src_size, dst_capacity); in Convert() 122 size_t dst_capacity) override { in Convert() argument 134 buffers_.back()->size(), dst, dst_capacity); in Convert() [all …]
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/dports/www/firefox-esr/firefox-91.8.0/third_party/libwebrtc/webrtc/common_audio/ |
H A D | audio_converter.cc | 35 size_t dst_capacity) override { in Convert() argument 36 CheckSizes(src_size, dst_capacity); in Convert() 52 size_t dst_capacity) override { in Convert() argument 53 CheckSizes(src_size, dst_capacity); in Convert() 71 size_t dst_capacity) override { in Convert() argument 72 CheckSizes(src_size, dst_capacity); in Convert() 96 size_t dst_capacity) override { in Convert() argument 97 CheckSizes(src_size, dst_capacity); in Convert() 123 size_t dst_capacity) override { in Convert() argument 135 buffers_.back()->size(), dst, dst_capacity); in Convert() [all …]
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/dports/lang/spidermonkey78/firefox-78.9.0/media/webrtc/trunk/webrtc/common_audio/ |
H A D | audio_converter.cc | 35 size_t dst_capacity) override { in Convert() argument 36 CheckSizes(src_size, dst_capacity); in Convert() 52 size_t dst_capacity) override { in Convert() argument 53 CheckSizes(src_size, dst_capacity); in Convert() 71 size_t dst_capacity) override { in Convert() argument 72 CheckSizes(src_size, dst_capacity); in Convert() 96 size_t dst_capacity) override { in Convert() argument 97 CheckSizes(src_size, dst_capacity); in Convert() 123 size_t dst_capacity) override { in Convert() argument 135 buffers_.back()->size(), dst, dst_capacity); in Convert() [all …]
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/dports/biology/plink/plink-ng-79b2df8c/2.0/ |
H A D | plink2_decompress.cc | 26 const uint32_t dst_capacity = RoundUpPow2(max_line_blen + kDecompressChunkSize, kCacheline); in InitTextStreamEx() local 27 if (unlikely(dst_capacity > bigstack_left())) { in InitTextStreamEx() 32 dst = S_CAST(char*, bigstack_alloc_raw(dst_capacity)); in InitTextStreamEx() 34 dst = S_CAST(char*, bigstack_end_alloc_raw(dst_capacity)); in InitTextStreamEx() 36 …return TextStreamOpenEx(fname, enforced_max_line_blen, dst_capacity, decompress_thread_ct, nullptr… in InitTextStreamEx()
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/dports/lang/spidermonkey60/firefox-60.9.0/media/webrtc/trunk/webrtc/common_audio/resampler/ |
H A D | push_resampler.cc | 40 size_t dst_capacity, in CheckExpectedBufferSizes() argument 52 RTC_DCHECK_GE(dst_capacity, dst_size_10ms); in CheckExpectedBufferSizes() 110 size_t dst_capacity) { in Resample() argument 111 CheckExpectedBufferSizes(src_length, dst_capacity, num_channels_, in Resample() 122 const size_t dst_capacity_mono = dst_capacity / num_channels_; in Resample() 138 sinc_resampler_->Resample(src, src_length, dst, dst_capacity)); in Resample()
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/dports/mail/thunderbird/thunderbird-91.8.0/third_party/libwebrtc/webrtc/common_audio/resampler/ |
H A D | push_resampler.cc | 41 size_t dst_capacity, in CheckExpectedBufferSizes() argument 53 RTC_DCHECK_GE(dst_capacity, dst_size_10ms); in CheckExpectedBufferSizes() 111 size_t dst_capacity) { in Resample() argument 112 CheckExpectedBufferSizes(src_length, dst_capacity, num_channels_, in Resample() 123 const size_t dst_capacity_mono = dst_capacity / num_channels_; in Resample() 139 sinc_resampler_->Resample(src, src_length, dst, dst_capacity)); in Resample()
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/dports/www/firefox-esr/firefox-91.8.0/third_party/libwebrtc/webrtc/common_audio/resampler/ |
H A D | push_resampler.cc | 41 size_t dst_capacity, in CheckExpectedBufferSizes() argument 53 RTC_DCHECK_GE(dst_capacity, dst_size_10ms); in CheckExpectedBufferSizes() 111 size_t dst_capacity) { in Resample() argument 112 CheckExpectedBufferSizes(src_length, dst_capacity, num_channels_, in Resample() 123 const size_t dst_capacity_mono = dst_capacity / num_channels_; in Resample() 139 sinc_resampler_->Resample(src, src_length, dst, dst_capacity)); in Resample()
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/dports/lang/spidermonkey78/firefox-78.9.0/media/webrtc/trunk/webrtc/common_audio/resampler/ |
H A D | push_resampler.cc | 41 size_t dst_capacity, in CheckExpectedBufferSizes() argument 53 RTC_DCHECK_GE(dst_capacity, dst_size_10ms); in CheckExpectedBufferSizes() 111 size_t dst_capacity) { in Resample() argument 112 CheckExpectedBufferSizes(src_length, dst_capacity, num_channels_, in Resample() 123 const size_t dst_capacity_mono = dst_capacity / num_channels_; in Resample() 139 sinc_resampler_->Resample(src, src_length, dst, dst_capacity)); in Resample()
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/dports/www/qt5-webengine/qtwebengine-everywhere-src-5.15.2/src/3rdparty/chromium/third_party/webrtc/common_audio/resampler/ |
H A D | push_resampler.cc | 43 size_t dst_capacity, in CheckExpectedBufferSizes() argument 55 RTC_DCHECK_GE(dst_capacity, dst_size_10ms); in CheckExpectedBufferSizes() 111 size_t dst_capacity) { in Resample() argument 112 CheckExpectedBufferSizes(src_length, dst_capacity, num_channels_, in Resample() 123 const size_t dst_capacity_mono = dst_capacity / num_channels_; in Resample()
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/dports/net-im/tg_owt/tg_owt-d578c76/src/common_audio/resampler/ |
H A D | push_resampler.cc | 43 size_t dst_capacity, in CheckExpectedBufferSizes() argument 55 RTC_DCHECK_GE(dst_capacity, dst_size_10ms); in CheckExpectedBufferSizes() 111 size_t dst_capacity) { in Resample() argument 112 CheckExpectedBufferSizes(src_length, dst_capacity, num_channels_, in Resample() 123 const size_t dst_capacity_mono = dst_capacity / num_channels_; in Resample()
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/dports/net-im/telegram-desktop/tdesktop-3.2.5-full/Telegram/ThirdParty/libtgvoip/webrtc_dsp/common_audio/resampler/ |
H A D | push_resampler.cc | 43 size_t dst_capacity, in CheckExpectedBufferSizes() argument 55 RTC_DCHECK_GE(dst_capacity, dst_size_10ms); in CheckExpectedBufferSizes() 109 size_t dst_capacity) { in Resample() argument 110 CheckExpectedBufferSizes(src_length, dst_capacity, num_channels_, in Resample() 121 const size_t dst_capacity_mono = dst_capacity / num_channels_; in Resample()
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/dports/www/chromium-legacy/chromium-88.0.4324.182/third_party/webrtc/common_audio/resampler/ |
H A D | push_resampler.cc | 43 size_t dst_capacity, in CheckExpectedBufferSizes() argument 55 RTC_DCHECK_GE(dst_capacity, dst_size_10ms); in CheckExpectedBufferSizes() 111 size_t dst_capacity) { in Resample() argument 112 CheckExpectedBufferSizes(src_length, dst_capacity, num_channels_, in Resample() 123 const size_t dst_capacity_mono = dst_capacity / num_channels_; in Resample()
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