| /system/update_engine/payload_consumer/ |
| bzip_extent_writer.cc | 43 // Copy the input data into |input_buffer_| only if |input_buffer_| already
48 if (!input_buffer_.empty()) {
49 input_buffer_.insert(input_buffer_.end(), input, input_end);
50 input = input_buffer_.data();
51 input_end = input + input_buffer_.size();
76 // Store unconsumed data (if any) in |input_buffer_|.
77 if (stream_.avail_in || !input_buffer_.empty()) {
79 new_input_buffer.swap(input_buffer_);
[all...] |
| xz_extent_writer.cc | 66 // Copy the input data into |input_buffer_| only if |input_buffer_| already
70 if (!input_buffer_.empty()) {
71 input_buffer_.insert(input_buffer_.end(), input, input + count);
72 input = input_buffer_.data();
73 count = input_buffer_.size();
105 // Store unconsumed data (if any) in |input_buffer_|. Since |input| can point
106 // to the existing |input_buffer_| we create a new one before assigning it.
109 input_buffer_ = std::move(new_input_buffer)
[all...] |
| bzip_extent_writer.h | 52 brillo::Blob input_buffer_; member in class:chromeos_update_engine::BzipExtentWriter
|
| xz_extent_writer.h | 53 brillo::Blob input_buffer_; member in class:chromeos_update_engine::XzExtentWriter
|
| /external/webrtc/webrtc/modules/audio_coding/codecs/opus/ |
| audio_encoder_opus.cc | 138 if (input_buffer_.empty())
141 input_buffer_.insert(input_buffer_.end(), audio.cbegin(), audio.cend());
142 if (input_buffer_.size() <
146 RTC_CHECK_EQ(input_buffer_.size(),
149 inst_, &input_buffer_[0],
150 rtc::CheckedDivExact(input_buffer_.size(), config_.num_channels),
153 input_buffer_.clear();
231 input_buffer_.clear();
232 input_buffer_.reserve(Num10msFramesPerPacket() * SamplesPer10msFrame())
[all...] |
| audio_encoder_opus.h | 94 std::vector<int16_t> input_buffer_; member in class:webrtc::final
|
| /external/webrtc/webrtc/common_audio/resampler/ |
| sinc_resampler.cc | 49 // r0_ = input_buffer_ + kKernelSize / 2
50 // r1_ = input_buffer_
76 // r0_ = r0_ + kKernelSize / 2 = input_buffer_ + kKernelSize
165 input_buffer_(static_cast<float*>(
170 r1_(input_buffer_.get()),
171 r2_(input_buffer_.get() + kKernelSize / 2) {
195 r0_ = input_buffer_.get() + (second_load ? kKernelSize : kKernelSize / 2);
201 assert(r1_ == input_buffer_.get());
334 memcpy(r1_, r3_, sizeof(*input_buffer_.get()) * kKernelSize);
354 memset(input_buffer_.get(), 0
[all...] |
| sinc_resampler.h | 145 rtc::scoped_ptr<float[], AlignedFreeDeleter> input_buffer_; member in class:webrtc::SincResampler
157 // Pointers to the various regions inside |input_buffer_|. See the diagram at
|
| /external/webrtc/webrtc/common_audio/ |
| blocker.cc | 114 input_buffer_(num_input_channels_, chunk_size_ + initial_delay_),
125 input_buffer_.MoveReadPositionBackward(initial_delay_);
176 input_buffer_.Write(input, num_input_channels, chunk_size_);
181 input_buffer_.Read(input_block_.channels(), num_input_channels,
183 input_buffer_.MoveReadPositionBackward(block_size_ - shift_amount_);
|
| blocker.h | 103 AudioRingBuffer input_buffer_; member in class:webrtc::Blocker
|
| /external/webrtc/webrtc/modules/audio_coding/codecs/ilbc/ |
| audio_encoder_ilbc.h | 57 int16_t input_buffer_[kMaxSamplesPerPacket]; member in class:webrtc::final
|
| audio_encoder_ilbc.cc | 106 input_buffer_ + kSampleRateHz / 100 * num_10ms_frames_buffered_);
119 input_buffer_,
|
| /art/runtime/jdwp/ |
| jdwp_adb.cc | 430 readCount = read(clientSock, input_buffer_ + input_count_, sizeof(input_buffer_) - input_count_);
463 if (memcmp(input_buffer_, kMagicHandshake, kMagicHandshakeLen) != 0) {
464 LOG(ERROR) << StringPrintf("ERROR: bad handshake '%.14s'", input_buffer_);
469 int cc = TEMP_FAILURE_RETRY(write(clientSock, input_buffer_, kMagicHandshakeLen));
|
| jdwp_socket.cc | 470 read(clientSock, input_buffer_ + input_count_, sizeof(input_buffer_) - input_count_);
503 if (memcmp(input_buffer_, kMagicHandshake, kMagicHandshakeLen) != 0) {
504 LOG(ERROR) << android::base::StringPrintf("ERROR: bad handshake '%.14s'", input_buffer_);
509 int cc = TEMP_FAILURE_RETRY(write(clientSock, input_buffer_, kMagicHandshakeLen));
|
| jdwp_priv.h | 102 uint8_t input_buffer_[8192]; member in class:art::JDWP::JdwpNetStateBase
|
| jdwp_main.cc | 88 memmove(input_buffer_, input_buffer_ + count, input_count_ - count);
99 uint32_t length = Get4BE(input_buffer_);
405 JDWP::Request request(netStateBase->input_buffer_, netStateBase->input_count_);
|
| /external/libbrillo/brillo/streams/ |
| fake_stream.cc | 96 input_buffer_.clear();
182 return input_ptr_ >= input_buffer_.size();
190 input_buffer_ = std::move(packet.data);
218 std::string message{input_buffer_.begin(), input_buffer_.end()};
221 input_buffer_.clear();
227 size_to_read = std::min(size_to_read, input_buffer_.size() - input_ptr_);
228 std::memcpy(buffer, input_buffer_.data() + input_ptr_, size_to_read);
|
| /external/webrtc/webrtc/modules/audio_processing/ |
| audio_buffer.cc | 114 if (need_to_downmix && !input_buffer_) {
115 input_buffer_.reset(
127 input_buffer_->fbuf()->channels()[0]);
128 data_ptr = input_buffer_->fbuf_const()->channels();
375 if ((input_num_frames_ != proc_num_frames_) && !input_buffer_) {
376 input_buffer_.reset(
385 deinterleaved = input_buffer_->ibuf()->channels();
402 input_resamplers_[i]->Resample(input_buffer_->fbuf_const()->channels()[i],
|
| audio_buffer.h | 154 rtc::scoped_ptr<IFChannelBuffer> input_buffer_; member in class:webrtc::AudioBuffer
|
| /external/protobuf/src/google/protobuf/io/ |
| gzip_stream.h | 191 void* input_buffer_; member in class:google::protobuf::io::GzipOutputStream
|
| gzip_stream.cc | 215 input_buffer_ = operator new(input_buffer_length_);
216 GOOGLE_CHECK(input_buffer_ != NULL);
244 operator delete(input_buffer_);
287 zcontext_.next_in = static_cast<Bytef*>(input_buffer_);
289 *data = input_buffer_;
|
| /prebuilts/misc/darwin-x86_64/protobuf2.5/include/google/protobuf/io/ |
| gzip_stream.h | 191 void* input_buffer_; member in class:google::protobuf::io::GzipOutputStream
|
| /prebuilts/misc/linux-x86_64/protobuf2.5/include/google/protobuf/io/ |
| gzip_stream.h | 191 void* input_buffer_; member in class:google::protobuf::io::GzipOutputStream
|
| /prebuilts/tools/darwin-x86_64/protoc/include/google/protobuf/io/ |
| gzip_stream.h | 191 void* input_buffer_; member in class:google::protobuf::io::GzipOutputStream
|
| /prebuilts/tools/linux-x86_64/protoc/include/google/protobuf/io/ |
| gzip_stream.h | 191 void* input_buffer_; member in class:google::protobuf::io::GzipOutputStream
|
