| /external/webrtc/webrtc/modules/rtp_rtcp/source/ |
| rtp_format_video_generic.cc | 24 payload_size_(0),
37 payload_size_ = payload_size;
41 (payload_size_ + max_payload_len_ - 1) / max_payload_len_;
42 payload_length_ = (payload_size_ + num_packets - 1) / num_packets;
51 if (payload_size_ < payload_length_) {
52 payload_length_ = payload_size_;
55 payload_size_ -= payload_length_;
72 *last_packet = payload_size_ <= 0;
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| rtp_format_vp8_test_helper.h | 42 size_t payload_size() const { return payload_size_; }
63 size_t payload_size_; member in class:webrtc::test::RtpFormatVp8TestHelper
|
| rtp_format_vp8_test_helper.cc | 27 payload_size_(0),
43 payload_size_ = 0;
46 payload_size_ += partition_sizes[p];
48 buffer_size_ = payload_size_ + 6; // Add space for payload descriptor.
49 payload_data_ = new uint8_t[payload_size_];
57 assert(j < payload_size_);
231 // If data_ptr_ has advanced payload_size_ bytes from the start (payload_data_)
234 EXPECT_EQ(last, data_ptr_ == payload_data_ + payload_size_);
|
| rtp_format_video_generic.h | 56 size_t payload_size_; member in class:webrtc::RtpPacketizerGeneric
|
| rtp_format_h264.h | 83 size_t payload_size_; member in class:webrtc::RtpPacketizerH264
|
| rtp_format_vp9.h | 91 size_t payload_size_; // The size in bytes of the payload data. member in class:webrtc::RtpPacketizerVp9
|
| rtp_format_vp8.h | 202 size_t payload_size_; member in class:webrtc::RtpPacketizerVp8
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| rtp_format_vp8.cc | 154 payload_size_(0),
168 payload_size_(0),
188 payload_size_ = payload_size;
288 while (total_bytes_processed < payload_size_) {
338 assert(total_bytes_processed == payload_size_);
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| rtp_format_vp9.cc | 485 payload_size_(0) {
518 payload_size_ = payload_size;
528 while (bytes_processed < payload_size_) {
529 size_t rem_bytes = payload_size_ - bytes_processed;
545 assert(bytes_processed == payload_size_);
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| rtp_format_h264.cc | 155 payload_size_(0),
169 payload_size_ = payload_size;
|
| rtp_format_vp9_unittest.cc | 132 size_t payload_size_; member in class:webrtc::RtpPacketizerVp9Test
140 payload_size_ = payload_size;
143 packetizer_->SetPayloadData(payload_.get(), payload_size_, NULL);
146 packet_.reset(new uint8_t[payload_size_ + kMaxPayloadDescriptorLength]);
156 EXPECT_EQ(last, payload_pos_ == payload_size_);
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| /external/webrtc/webrtc/modules/rtp_rtcp/test/testAPI/ |
| test_api.h | 63 size_t payload_size() const { return payload_size_; }
68 size_t payload_size_; member in class:webrtc::TestRtpReceiver
|
| test_api.cc | 76 payload_size_ = payload_size;
|
| /external/webrtc/webrtc/modules/audio_coding/test/ |
| TestAllCodecs.h | 49 size_t payload_size_; member in class:webrtc::TestPack
|
| TestStereo.h | 57 int payload_size_; member in class:webrtc::TestPackStereo
|
| TestAllCodecs.cc | 49 payload_size_(0) {
88 payload_size_ = payload_size;
96 return payload_size_;
104 payload_size_ = 0;
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| TestStereo.cc | 34 payload_size_(0),
78 payload_size_ = static_cast<int>(payload_size);
80 payload_size_ = -1;
91 return static_cast<uint16_t>(payload_size_);
99 payload_size_ = 0;
|
| /external/webrtc/webrtc/modules/remote_bitrate_estimator/test/ |
| packet.h | 40 virtual size_t payload_size() const { return payload_size_; }
60 size_t payload_size_; // Size of the (non-existent, simulated) payload. member in class:webrtc::testing::bwe::Packet
|
| bwe_test_framework.cc | 100 payload_size_(0),
109 payload_size_(payload_size),
|
| /external/libchrome/base/ |
| pickle.h | 120 size_t payload_size() const { return payload_size_; }
144 size_t payload_size_ = 0; member in class:base::PickleSizer
|
| pickle.cc | 231 payload_size_ += bits::Align(length, sizeof(uint32_t));
|
| /system/update_engine/update_manager/ |
| real_updater_provider.cc | 65 &payload_size_);
75 inline int64_t payload_size() { return payload_size_; }
83 int64_t payload_size_; member in class:chromeos_update_manager::GetStatusHelper
|
| /external/libvpx/libvpx/third_party/libwebm/mkvmuxer/ |
| mkvmuxer.h | 1357 uint64_t payload_size_; member in class:mkvmuxer::Cluster
[all...] |
| mkvmuxer.cc | [all...] |
