/external/compiler-rt/test/asan/TestCases/ |
H A D | debug_stacks.cc | 29 size_t num_frames = 100; local 31 num_frames = __asan_get_alloc_stack(mem, trace, num_frames, &thread_id); 33 fprintf(stderr, "alloc stack retval %s\n", (num_frames > 0 && num_frames < 10) 43 num_frames = 100; 44 num_frames = __asan_get_free_stack(mem, trace, num_frames, &thread_id); 46 fprintf(stderr, "free stack retval %s\n", (num_frames > 0 && num_frames < 1 [all...] |
/external/webrtc/webrtc/common_audio/ |
H A D | audio_util.cc | 44 size_t num_frames, 47 DownmixInterleavedToMonoImpl<int16_t, int32_t>(interleaved, num_frames, 43 DownmixInterleavedToMono(const int16_t* interleaved, size_t num_frames, int num_channels, int16_t* deinterleaved) argument
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H A D | channel_buffer.cc | 15 IFChannelBuffer::IFChannelBuffer(size_t num_frames, argument 19 ibuf_(num_frames, num_channels, num_bands), 21 fbuf_(num_frames, num_channels, num_bands) {} 51 for (size_t j = 0; j < ibuf_.num_frames(); ++j) { 66 ibuf_.num_frames(),
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H A D | lapped_transform.cc | 23 size_t num_frames, 29 RTC_CHECK_EQ(parent_->block_length_, num_frames); 33 num_frames * sizeof(*input[0])); 39 RealFourier::FftOrder(num_frames)); 51 num_frames * sizeof(*input[0])); 22 ProcessBlock(const float* const* input, size_t num_frames, size_t num_input_channels, size_t num_output_channels, float* const* output) argument
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H A D | blocker.cc | 24 size_t num_frames, 29 for (size_t j = 0; j < num_frames; ++j) { 39 size_t num_frames, 46 num_frames * sizeof(dst[i][dst_start_index])); 53 size_t num_frames, 60 num_frames * sizeof(dst[i][dst_start_index])); 66 size_t num_frames, 70 num_frames * sizeof(buffer[i][starting_idx])); 77 size_t num_frames, 81 for (size_t j = 0; j < num_frames; 20 AddFrames(const float* const* a, size_t a_start_index, const float* const* b, int b_start_index, size_t num_frames, size_t num_channels, float* const* result, size_t result_start_index) argument 37 CopyFrames(const float* const* src, size_t src_start_index, size_t num_frames, size_t num_channels, float* const* dst, size_t dst_start_index) argument 51 MoveFrames(const float* const* src, size_t src_start_index, size_t num_frames, size_t num_channels, float* const* dst, size_t dst_start_index) argument 64 ZeroOut(float* const* buffer, size_t starting_idx, size_t num_frames, size_t num_channels) argument 76 ApplyWindow(const float* window, size_t num_frames, size_t num_channels, float* const* frames) argument [all...] |
H A D | blocker_unittest.cc | 22 size_t num_frames, 27 for (size_t j = 0; j < num_frames; ++j) { 38 size_t num_frames, 43 for (size_t j = 0; j < num_frames; ++j) { 61 size_t num_frames, 70 while (end < num_frames) { 87 size_t num_frames) { 89 for (size_t j = 0; j < num_frames; ++j) { 97 size_t num_frames, 100 for (size_t j = 0; j < num_frames; 59 RunTest(Blocker* blocker, size_t chunk_size, size_t num_frames, const float* const* input, float* const* input_chunk, float* const* output, float* const* output_chunk, size_t num_input_channels, size_t num_output_channels) argument 84 ValidateSignalEquality(const float* const* expected, const float* const* actual, size_t num_channels, size_t num_frames) argument 95 ValidateInitialDelay(const float* const* output, size_t num_channels, size_t num_frames, size_t initial_delay) argument 110 CopyTo(float* const* dst, size_t start_index_dst, size_t start_index_src, size_t num_channels, size_t num_frames, const float* const* src) argument [all...] |
H A D | channel_buffer.h | 42 ChannelBuffer(size_t num_frames, argument 45 : data_(new T[num_frames * num_channels]()), 48 num_frames_(num_frames), 49 num_frames_per_band_(num_frames / num_bands), 116 size_t num_frames() const { return num_frames_; } function in class:webrtc::ChannelBuffer 145 IFChannelBuffer(size_t num_frames, size_t num_channels, size_t num_bands = 1); 152 size_t num_frames() const { return ibuf_.num_frames(); } function in class:webrtc::IFChannelBuffer
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/external/webrtc/webrtc/modules/audio_processing/ |
H A D | splitting_filter.cc | 21 size_t num_frames) 28 three_band_filter_banks_.push_back(new ThreeBandFilterBank(num_frames)); 37 RTC_DCHECK_EQ(data->num_frames(), 50 RTC_DCHECK_EQ(data->num_frames(), 64 data->num_frames(), 90 data->num_frames(), 19 SplittingFilter(size_t num_channels, size_t num_bands, size_t num_frames) argument
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H A D | audio_buffer.cc | 35 size_t NumBandsFromSamplesPerChannel(size_t num_frames) { argument 37 if (num_frames == kSamplesPer32kHzChannel || 38 num_frames == kSamplesPer48kHzChannel) { 39 num_bands = rtc::CheckedDivExact(num_frames, kSamplesPer16kHzChannel); 107 assert(stream_config.num_frames() == input_num_frames_); 152 assert(stream_config.num_frames() == output_num_frames_); 352 size_t AudioBuffer::num_frames() const { function in class:webrtc::AudioBuffer
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/external/webrtc/webrtc/modules/audio_processing/vad/ |
H A D | common.h | 23 size_t num_frames; member in struct:AudioFeatures
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H A D | standalone_vad_unittest.cc | 81 int num_frames = 0; local 84 num_frames++; 85 if (num_frames == kNumVadFramesToProcess) { 86 num_frames = 0;
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H A D | vad_audio_proc_unittest.cc | 49 if (features.num_frames > 0) { 50 ASSERT_LT(features.num_frames, kMaxNumFrames); 52 const size_t num_frames = features.num_frames; local 53 ASSERT_EQ(num_frames, fread(sp, sizeof(sp[0]), num_frames, peak_file)); 54 for (size_t n = 0; n < features.num_frames; n++)
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H A D | standalone_vad.cc | 64 const size_t num_frames = index_ / kLength10Ms; local 65 if (num_frames > length_p) 76 for (size_t n = 1; n < num_frames; n++)
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/external/webrtc/webrtc/modules/audio_coding/codecs/ilbc/ |
H A D | audio_decoder_ilbc.cc | 44 size_t AudioDecoderIlbc::DecodePlc(size_t num_frames, int16_t* decoded) { argument 45 return WebRtcIlbcfix_NetEqPlc(dec_state_, decoded, num_frames);
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/external/webrtc/webrtc/modules/audio_coding/codecs/ |
H A D | audio_decoder.cc | 56 size_t AudioDecoder::DecodePlc(size_t num_frames, int16_t* decoded) { argument
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/external/webrtc/webrtc/modules/audio_coding/codecs/isac/ |
H A D | audio_decoder_isac_t_impl.h | 66 size_t AudioDecoderIsacT<T>::DecodePlc(size_t num_frames, int16_t* decoded) { argument 67 return T::DecodePlc(isac_state_, decoded, num_frames);
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/external/autotest/client/site_tests/graphics_SanAngeles/src/ |
H A D | app-linux.c | 227 int num_frames = 0; local 255 num_frames++; 261 fprintf(stdout, "frame_rate = %.1f\n", num_frames / total_time);
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/external/webrtc/webrtc/common_audio/include/ |
H A D | audio_util.h | 73 int num_frames, 78 std::copy(src[i], src[i] + num_frames, dest[i]); 125 int num_frames, 129 for (int i = 0; i < num_frames; ++i) { 138 size_t num_frames, 141 for (size_t i = 0; i < num_frames; ++i) { 154 size_t num_frames, 158 RTC_DCHECK_GT(num_frames, 0u); 160 const T* const end = interleaved + num_frames * num_channels; 176 size_t num_frames, 72 CopyAudioIfNeeded(const T* const* src, int num_frames, int num_channels, T* const* dest) argument 124 UpmixMonoToInterleaved(const T* mono, int num_frames, int num_channels, T* interleaved) argument 137 DownmixToMono(const T* const* input_channels, size_t num_frames, int num_channels, T* out) argument 153 DownmixInterleavedToMonoImpl(const T* interleaved, size_t num_frames, int num_channels, T* deinterleaved) argument [all...] |
/external/webp/include/webp/ |
H A D | demux.h | 140 int num_frames; // equivalent to WEBP_FF_FRAME_COUNT. member in struct:WebPIterator
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/external/webp/src/webp/ |
H A D | demux.h | 140 int num_frames; // equivalent to WEBP_FF_FRAME_COUNT. member in struct:WebPIterator
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/external/webrtc/webrtc/modules/audio_coding/test/ |
H A D | delay_test.cc | 181 int num_frames = 0; local 187 while (num_frames < (duration_sec * 100)) { 193 if ((num_frames & 0x3F) == 0x3F) { 217 if (num_frames > 10) 220 ++num_frames;
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/external/webrtc/webrtc/modules/video_coding/codecs/vp8/ |
H A D | vp8_sequence_coder.cc | 125 // Get range of frames: will encode num_frames following start_frame). 127 int num_frames = strtol((parser->GetFlag("num_frames")).c_str(), NULL, 10); local 159 // num_frames = -1 implies unlimited encoding (entire sequence). 165 (num_frames == -1 || frames_processed < num_frames)) { 214 " - num_frames - Number of frames to be processed. " 229 parser.SetFlag("num_frames", "-1");
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/external/webrtc/webrtc/tools/agc/ |
H A D | activity_metric.cc | 111 if (features.num_frames > 0) { 119 for (size_t n = 0; n < features.num_frames; n++) { 128 for (size_t n = 0; n < features.num_frames; n++) { 142 return static_cast<int>(features.num_frames); 236 int num_frames = 0; local 277 num_frames++;
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/external/webrtc/webrtc/video/ |
H A D | overuse_frame_detector_unittest.cc | 76 int num_frames, int interval_ms, int width, int height, int delay_ms) { 77 while (num_frames-- > 0) { 75 InsertAndSendFramesWithInterval( int num_frames, int interval_ms, int width, int height, int delay_ms) argument
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/external/libyuv/files/util/ |
H A D | convert.cc | 37 int num_frames = 0; // Number of frames to convert. variable 102 num_frames = atoi(argv[++c]); // NOLINT 124 if (num_frames < 0) { 268 if (num_frames && number_of_frames >= num_frames)
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