| /external/v8/src/ |
| bit-vector.h | 17 uintptr_t* ptr_; // valid if data_length_ > 1
18 uintptr_t inline_; // valid if data_length_ == 1
36 bool Done() const { return current_index_ >= target_->data_length_; }
73 BitVector() : length_(0), data_length_(kDataLengthForInline), data_(0) {}
76 : length_(length), data_length_(SizeFor(length)), data_(0) {
79 data_.ptr_ = zone->NewArray<uintptr_t>(data_length_);
87 data_length_(other.data_length_),
90 data_.ptr_ = zone->NewArray<uintptr_t>(data_length_);
91 for (int i = 0; i < other.data_length_; i++)
279 int data_length_; member in class:v8::internal::BitVector
[all...] |
| bit-vector.cc | 46 if (data_length_ == 0) {
50 for (int i = 0; i < data_length_; i++) {
|
| /external/webrtc/webrtc/test/testsupport/ |
| packet_reader.cc | 31 data_length_ = data_length_in_bytes;
44 currentIndex_ = std::min(currentIndex_ + packet_size_, data_length_);
|
| packet_reader.h | 45 size_t data_length_; member in class:webrtc::test::PacketReader
|
| /external/webrtc/webrtc/modules/audio_processing/transient/ |
| wpd_tree.h | 83 size_t data_length_; member in class:webrtc::WPDTree
|
| transient_suppressor.cc | 48 : data_length_(0),
102 data_length_ = sample_rate_hz * ts::kChunkSizeMs / 1000;
103 if (data_length_ > analysis_length_) {
107 buffer_delay_ = analysis_length_ - data_length_;
174 if (!data || data_length != data_length_ || num_channels != num_channels_ ||
221 memcpy(&data[i * data_length_],
224 data_length_ * sizeof(*data));
341 &in_buffer_[data_length_],
347 &data[i * data_length_],
348 data_length_ * sizeof(*data))
[all...] |
| wpd_tree.cc | 26 : data_length_(data_length),
77 if (!data || data_length != data_length_) {
|
| transient_suppressor.h | 37 // |data_length| must be equal to |data_length_|.
76 size_t data_length_; member in class:webrtc::TransientSuppressor
|
| /external/webrtc/webrtc/base/ |
| stream.cc | 525 : buffer_(NULL), buffer_length_(0), data_length_(0),
535 if (seek_position_ >= data_length_) {
538 size_t available = data_length_ - seek_position_;
573 if (data_length_ < seek_position_) {
574 data_length_ = seek_position_;
587 if (position > data_length_)
601 *size = data_length_;
607 *size = data_length_ - seek_position_;
640 data_length_ = buffer_length_ = length;
644 memcpy(buffer_, data, data_length_);
[all...] |
| stream.h | 454 // Invariant: 0 <= seek_position <= data_length_ <= buffer_length_
457 size_t data_length_; member in class:rtc::MemoryStreamBase
507 // Resizes the buffer to the specified capacity. Fails if data_length_ > size
555 size_t data_length_; // amount of readable data in the buffer member in class:rtc::FifoBuffer
|
| /frameworks/av/media/libeffects/loudness/dsp/core/ |
| interpolator_base.h | 91 int data_length_; member in class:le_fx::sigmod::InterpolatorBase
92 // Index of the last element `data_length_ - 1` kept here for optimization
|
| interpolator_base-inl.h | 38 data_length_ = 0;
102 data_length_ = 0;
110 data_length_ = data_length;
131 if (cached_index_ < 0 || cached_index_ > data_length_ - 2) {
133 "[0, %d, %d]", cached_index_, data_length_ - 2);
|
| interpolator_linear.h | 57 using BaseClass::data_length_;
|
| /external/webrtc/webrtc/modules/audio_coding/neteq/ |
| audio_decoder_unittest.cc | 105 data_length_(0),
117 ASSERT_GT(data_length_, 0u) << "The test must set data_length_ > 0";
119 encoded_ = new uint8_t[data_length_ * 2];
165 data_length_ * 2, output);
186 while (processed_samples + frame_size_ <= data_length_) {
278 size_t data_length_; member in namespace:webrtc
291 data_length_ = 10 * frame_size_;
304 data_length_ = 10 * frame_size_;
318 data_length_ = 10 * frame_size_
[all...] |
| /external/webrtc/webrtc/test/ |
| fake_network_pipe.cc | 44 data_length_(length),
55 size_t data_length() const { return data_length_; }
66 size_t data_length_; member in class:webrtc::NetworkPacket
|
