| /external/webrtc/webrtc/modules/video_coding/codecs/vp8/ |
| screenshare_layers_unittest.cc | 36 ScreenshareLayerTest() : min_qp_(2), max_qp_(kDefaultQp), frame_size_(-1) {}
45 ASSERT_NE(-1, frame_size_);
46 layers_->FrameEncoded(frame_size_, timestamp, kDefaultQp);
56 frame_size_ = ((vpx_cfg.rc_target_bitrate * 1000) / 8) / kFrameRate;
92 layers_->FrameEncoded(frame_size_, timestamp, kDefaultQp);
103 int frame_size_; member in class:webrtc::ScreenshareLayerTest
122 layers_->FrameEncoded(frame_size_, timestamp, kDefaultQp);
130 layers_->FrameEncoded(frame_size_, timestamp, kDefaultQp);
219 layers_->FrameEncoded(frame_size_, timestamp, kDefaultQp);
221 layers_->FrameEncoded(frame_size_, timestamp, kDefaultQp - 8)
[all...] |
| /external/webrtc/talk/media/base/ |
| testutils.h | 142 uint32_t frame_size() const { return frame_size_; }
154 uint32_t frame_size_; member in class:cricket::VideoCapturerListener
|
| /external/webrtc/webrtc/modules/audio_coding/neteq/ |
| audio_decoder_unittest.cc | 104 frame_size_(0),
186 while (processed_samples + frame_size_ <= data_length_) {
187 // Extend input vector with |frame_size_|.
188 input.resize(input.size() + frame_size_, 0);
190 ASSERT_GE(input.size() - processed_samples, frame_size_);
192 frame_size_, codec_input_rate_hz_, &input[processed_samples]));
194 &input[processed_samples], frame_size_, &encoded_[encoded_bytes_]);
195 // Make sure that frame_size_ * channels_ samples are allocated and free.
196 decoded.resize((processed_samples + frame_size_) * channels_, 0);
200 frame_size_ * channels_ * sizeof(int16_t)
277 size_t frame_size_; member in namespace:webrtc
[all...] |
| packet_buffer_unittest.cc | 36 int frame_size_; member in class:webrtc::PacketGenerator
49 frame_size_ = frame_size;
65 ts_ += frame_size_;
|
| /external/webrtc/webrtc/test/ |
| frame_generator.cc | 65 frame_size_(CalcBufferSize(kI420,
68 frame_buffer_(new uint8_t[frame_size_]),
98 fread(frame_buffer_.get(), 1, frame_size_, files_[file_index_]);
99 if (bytes_read < frame_size_) {
103 bytes_read = fread(frame_buffer_.get(), 1, frame_size_,
105 assert(bytes_read >= frame_size_);
123 const size_t frame_size_; member in class:webrtc::test::__anon27327::YuvFileGenerator
|
| /external/webrtc/webrtc/sound/ |
| alsasoundsystem.cc | 84 frame_size_(frame_size),
218 return frame_size_;
224 size_t frame_size_; member in class:rtc::AlsaStream
|
| /art/compiler/optimizing/ |
| code_generator.h | 215 uint32_t GetFrameSize() const { return frame_size_; }
216 void SetFrameSize(uint32_t size) { frame_size_ = size; }
499 : frame_size_(0),
583 uint32_t frame_size_; member in class:art::CodeGenerator
|
