| /external/tensorflow/tensorflow/contrib/lite/toco/tflite/ |
| types_test.cc | 46 // Deserialize normally takes a buffer, we need to synthesize one and provide
47 // an index that is non-zero so the buffer is not assumed to be emtpy.
55 /*buffer*/ 1)); // Can't use 0 which means empty.
63 auto* buffer = local
64 flatbuffers::GetRoot<::tflite::Buffer>(buffer_builder.GetBufferPointer());
65 DataBuffer::Deserialize(*tensor, *buffer, &result);
103 auto* buffer = local
104 flatbuffers::GetRoot<::tflite::Buffer>(buffer_builder.GetBufferPointer());
106 DataBuffer::Deserialize(*tensor, *buffer, &array);
107 EXPECT_EQ(nullptr, array.buffer);
126 auto* buffer = flatbuffers::GetRoot<::tflite::Buffer>( local
[all...] |
| /external/tensorflow/tensorflow/core/common_runtime/sycl/ |
| sycl_allocator.cc | 86 const auto& buffer = sycl_device_->get_sycl_buffer(ptr); local
87 return buffer.get_size();
|
| /external/tensorflow/tensorflow/core/distributed_runtime/rpc/ |
| grpc_util.cc | 36 // Use loop instead of if in case buffer contained empty slices.
87 ::grpc::ByteBuffer buffer(&s, 1);
88 dst->Swap(&buffer);
95 ::grpc::ByteBuffer buffer(&s, 1);
96 dst->Swap(&buffer);
110 const ::grpc::ByteBuffer* buffer; member in struct:tensorflow::ByteSource
115 ok = src.Init(*buffer);
120 bs.buffer = &src;
|
| /external/tensorflow/tensorflow/core/util/ |
| memmapped_file_system_writer.cc | 101 char buffer[sizeof(uint64)]; local
103 EncodeUint64LittleEndian(output_file_offset_, buffer)));
|
| /external/tpm2/ |
| ExecCommand.c | 14 // Uncomment this next #include if doing static command/response buffer sizing
22 // a) Parses the command header from input buffer.
31 // 1) unmarshal the command parameters from the command buffer;
33 // 3) marshal the responses into the response buffer.
43 unsigned int requestSize, // IN: command buffer size
44 unsigned char *request, // IN: command buffer
45 unsigned int *responseSize, // OUT: response buffer size
46 unsigned char **response // OUT: response buffer
54 // optional parameter buffer
76 // buffer
77 BYTE *buffer; \/\/ pointer into the buffer being used local
[all...] |
| /external/v8/src/heap/ |
| array-buffer-tracker.cc | 5 #include "src/heap/array-buffer-tracker.h"
6 #include "src/heap/array-buffer-tracker-inl.h"
21 JSArrayBuffer* buffer = reinterpret_cast<JSArrayBuffer*>(it->first); local
22 if ((free_mode == kFreeAll) || ObjectMarking::IsWhite(buffer)) {
24 heap_->isolate()->array_buffer_allocator()->Free(buffer->backing_store(),
127 bool ArrayBufferTracker::IsTracked(JSArrayBuffer* buffer) {
128 Page* page = Page::FromAddress(buffer->address());
133 return tracker->IsTracked(buffer);
|
| /external/v8/src/wasm/ |
| wasm-result.cc | 35 char buffer[256]; local
36 base::OS::VSNPrintF(buffer, 255, format, args);
42 str << buffer; local
|
| /external/vboot_reference/firmware/stub/ |
| vboot_api_stub_sf.c | 37 void *buffer[MAX_STACK_LEVELS]; local
38 int levels = backtrace(buffer, MAX_STACK_LEVELS);
41 backtrace_symbols_fd(buffer + 1, levels - 1, 2);
|
| /external/webrtc/webrtc/modules/audio_coding/acm2/ |
| audio_coding_module_impl.h | 17 #include "webrtc/base/buffer.h"
193 // If a re-mix is required (up or down), this buffer will store a re-mixed
195 int16_t buffer[WEBRTC_10MS_PCM_AUDIO]; member in struct:webrtc::acm2::final::InputData
225 // required, before pushing audio into encoder's buffer.
244 rtc::Buffer encode_buffer_ GUARDED_BY(acm_crit_sect_);
|
| /external/webrtc/webrtc/modules/audio_coding/neteq/ |
| dtmf_buffer_unittest.cc | 57 DtmfBuffer* buffer = new DtmfBuffer(sample_rate_hz); local
58 delete buffer;
91 DtmfBuffer buffer(sample_rate_hz);
92 EXPECT_EQ(DtmfBuffer::kOK, buffer.InsertEvent(event));
93 EXPECT_EQ(1u, buffer.Length());
94 EXPECT_FALSE(buffer.Empty());
97 EXPECT_FALSE(buffer.GetEvent(timestamp - 10, &out_event));
98 EXPECT_EQ(1u, buffer.Length());
99 EXPECT_FALSE(buffer.Empty());
101 EXPECT_TRUE(buffer.GetEvent(timestamp, &out_event))
[all...] |
| /external/webrtc/webrtc/modules/audio_device/ |
| fine_audio_buffer_unittest.cc | 30 // E.g. if there are two buffers of size 3, buffer 1 would contain 0,1,2 and
31 // buffer 2 would contain 3,4,5. Note that SCHAR_MAX is 127 so wrap-around
33 // |buffer| is the audio buffer to verify.
34 bool VerifyBuffer(const int8_t* buffer, int buffer_number, int size) {
37 if (buffer[i] != (i + start_value) % SCHAR_MAX) {
46 // sequence 0,1,..SCHAR_MAX-1,0,1,... to the buffer. Note that this is likely a
47 // buffer of different size than the one VerifyBuffer verifies.
50 // buffer (|arg0|).
52 int8_t* buffer = static_cast<int8_t*>(arg0) local
75 const int8_t* buffer = static_cast<const int8_t*>(arg0); local
[all...] |
| /external/webrtc/webrtc/modules/rtp_rtcp/source/rtcp_packet/ |
| compound_packet_unittest.cc | 42 parser.Parse(packet->Buffer(), packet->Length());
57 parser.Parse(packet->Buffer(), packet->Length());
77 parser.Parse(packet->Buffer(), packet->Length());
111 uint8_t buffer[kBufferSize]; local
112 EXPECT_TRUE(rr.BuildExternalBuffer(buffer, kBufferSize, &verifier));
152 uint8_t buffer[kBufferSize]; local
153 EXPECT_TRUE(rr.BuildExternalBuffer(buffer, kBufferSize, &verifier));
|
| /external/webrtc/webrtc/modules/rtp_rtcp/source/ |
| rtcp_utility_unittest.cc | 79 RtcpParseCommonHeaderTest() { memset(buffer, 0, kBufferCapacityBytes); }
84 uint8_t buffer[kBufferCapacityBytes]; member in class:webrtc::rtcp::RtcpParseCommonHeaderTest
89 // Buffer needs to be able to hold the header.
91 EXPECT_FALSE(RtcpParseCommonHeader(buffer, i, &header));
97 buffer[0] = v << 6;
99 buffer, RtcpCommonHeader::kHeaderSizeBytes, &header));
105 buffer[0] = 2 << 6;
108 ByteWriter<uint16_t>::WriteBigEndian(&buffer[2], kBlockSize);
111 EXPECT_FALSE(RtcpParseCommonHeader(buffer, kSizeInBytes - 1, &header));
112 EXPECT_TRUE(RtcpParseCommonHeader(buffer, kSizeInBytes, &header))
[all...] |
| /external/xmp_toolkit/XMPCore/src/com/adobe/xmp/impl/ |
| ByteBuffer.java | 20 * Byte buffer container including length of valid data.
27 private byte[] buffer; field in class:ByteBuffer
35 * @param initialCapacity the initial capacity for this buffer
39 this.buffer = new byte[initialCapacity];
45 * @param buffer a byte array that will be wrapped with <code>ByteBuffer</code>.
47 public ByteBuffer(byte[] buffer)
49 this.buffer = buffer;
50 this.length = buffer.length;
55 * @param buffer a byte array that will be wrapped with <code>ByteBuffer</code>
[all...] |
| /frameworks/av/cmds/stagefright/ |
| SineSource.cpp | 66 MediaBufferBase *buffer; local
67 status_t err = mGroup->acquire_buffer(&buffer);
74 size_t numFramesPerBuffer = buffer->size() / frameSize;
76 int16_t *ptr = (int16_t *)buffer->data();
92 buffer->meta_data().setInt64(
97 buffer->set_range(0, numFramesPerBuffer * frameSize);
99 *out = buffer;
|
| /frameworks/av/media/libaaudio/tests/ |
| test_attributes.cpp | 37 float *buffer = new float[kNumFrames * kChannelCount]; local
87 AAudioStream_read(aaudioStream, buffer, kNumFrames, kNanosPerSecond));
90 AAudioStream_write(aaudioStream, buffer, kNumFrames, kNanosPerSecond));
96 delete[] buffer;
|
| test_recovery.cpp | 35 float *buffer = nullptr; local
45 // in a buffer if we hang or crash.
86 buffer = new float[samplesPerBurst];
101 buffer, framesPerBurst,
137 delete[] buffer;
|
| /frameworks/av/media/libaudioprocessing/ |
| RecordBufferConverter.cpp | 89 AudioBufferProvider::Buffer buffer; local
91 buffer.frameCount = i;
92 status_t status = provider->getNextBuffer(&buffer);
93 if (status != OK || buffer.frameCount == 0) {
97 // format convert to destination buffer
98 convertNoResampler(dst, buffer.raw, buffer.frameCount);
100 dst = (int8_t*)dst + buffer.frameCount * mDstFrameSize;
101 i -= buffer.frameCount
[all...] |
| /frameworks/av/media/libmediaextractor/ |
| MediaBuffer.cpp | 73 ALOGV("Allocated shared mem buffer of size %zu @ %p", size, mData);
78 MediaBuffer::MediaBuffer(const sp<ABuffer> &buffer)
81 mData(buffer->data()),
82 mSize(buffer->size()),
85 mBuffer(buffer),
182 MediaBuffer *buffer = new MediaBuffer(mData, mSize); local
183 buffer->set_range(mRangeOffset, mRangeLength);
184 buffer->mMetaData = new MetaDataBase(*mMetaData);
187 buffer->mOriginal = this;
189 return buffer;
[all...] |
| /frameworks/av/media/libnbaio/ |
| NBAIO.cpp | 76 double buffer[((frameSize * block) + sizeof(double) - 1) / sizeof(double)]; local
83 ssize_t ret = via(user, buffer, count);
87 ret = write(buffer, maxRet);
112 double buffer[((frameSize * block) + sizeof(double) - 1) / sizeof(double)]; local
119 ssize_t ret = read(buffer, count);
123 ret = via(user, buffer, maxRet);
|
| /frameworks/av/media/libstagefright/foundation/ |
| base64.cpp | 48 sp<ABuffer> buffer = new ABuffer(outLen); local
49 uint8_t *out = buffer->data();
50 if (out == NULL || buffer->size() < outLen) {
89 return buffer;
|
| /frameworks/av/media/libstagefright/httplive/ |
| LiveDataSource.cpp | 69 sp<ABuffer> buffer = *it; local
71 totalAvailable += buffer->size();
112 sp<ABuffer> buffer = *mBufferQueue.begin(); local
116 if (copy > buffer->size()) {
117 copy = buffer->size();
120 memcpy((uint8_t *)data + sizeDone, buffer->data(), copy);
124 buffer->setRange(buffer->offset() + copy, buffer->size() - copy);
126 if (buffer->size() == 0)
[all...] |
| /frameworks/av/media/libstagefright/rtsp/ |
| ARawAudioAssembler.cpp | 76 sp<ABuffer> buffer = *queue->begin(); local
80 mNextExpectedSeqNo = (uint32_t)buffer->int32Data();
81 } else if ((uint32_t)buffer->int32Data() != mNextExpectedSeqNo) {
87 // hexdump(buffer->data(), buffer->size());
89 if (buffer->size() < 1) {
99 msg->setBuffer("access-unit", buffer);
|
| SDPLoader.cpp | 119 sp<ABuffer> buffer = new ABuffer(sdpSize); local
122 ssize_t readSize = mHTTPDataSource->readAt(0, buffer->data(), sdpSize);
130 if (desc == NULL || !desc->setTo(buffer->data(), (size_t)readSize)) {
|
| UDPPusher.cpp | 82 sp<ABuffer> buffer = new ABuffer(length); local
83 if (fread(buffer->data(), 1, length, mFile) < length) {
89 mSocket, buffer->data(), buffer->size(), 0,
92 CHECK_EQ(n, (ssize_t)buffer->size());
118 sp<ABuffer> buffer = new ABuffer(8); local
119 uint8_t *data = buffer->data();
128 buffer->setRange(0, 8);
134 mSocket, buffer->data(), buffer->size(), 0
[all...] |
