| /external/webrtc/webrtc/modules/audio_coding/neteq/ |
| D | audio_decoder_unittest.cc | 104 frame_size_(0), in AudioDecoderTest()
186 while (processed_samples + frame_size_ <= data_length_) { in EncodeDecodeTest()
188 input.resize(input.size() + frame_size_, 0); in EncodeDecodeTest()
190 ASSERT_GE(input.size() - processed_samples, frame_size_); in EncodeDecodeTest()
192 frame_size_, codec_input_rate_hz_, &input[processed_samples])); in EncodeDecodeTest()
194 &input[processed_samples], frame_size_, &encoded_[encoded_bytes_]); in EncodeDecodeTest()
196 decoded.resize((processed_samples + frame_size_) * channels_, 0); in EncodeDecodeTest()
200 frame_size_ * channels_ * sizeof(int16_t), in EncodeDecodeTest()
202 EXPECT_EQ(frame_size_ * channels_, dec_len); in EncodeDecodeTest()
204 processed_samples += frame_size_; in EncodeDecodeTest()
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| D | packet_buffer_unittest.cc | 36 int frame_size_; member in webrtc::PacketGenerator
49 frame_size_ = frame_size; in Reset()
65 ts_ += frame_size_; in NextPacket()
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| /external/webrtc/webrtc/modules/video_coding/codecs/vp8/ |
| D | screenshare_layers_unittest.cc | 36 ScreenshareLayerTest() : min_qp_(2), max_qp_(kDefaultQp), frame_size_(-1) {} in ScreenshareLayerTest()
45 ASSERT_NE(-1, frame_size_); in EncodeFrame()
46 layers_->FrameEncoded(frame_size_, timestamp, kDefaultQp); in EncodeFrame()
56 frame_size_ = ((vpx_cfg.rc_target_bitrate * 1000) / 8) / kFrameRate; in ConfigureBitrates()
92 layers_->FrameEncoded(frame_size_, timestamp, kDefaultQp); in SkipUntilTl()
103 int frame_size_; member in 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);
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| /external/v4l2_codec2/accel/ |
| D | vp8_decoder.cc | 17 frame_size_(0), in VP8Decoder()
35 frame_size_ = size; in SetStream()
43 frame_size_ = 0; in Reset()
54 if (!curr_frame_start_ || frame_size_ == 0) in Decode()
59 if (!parser_.ParseFrame(curr_frame_start_, frame_size_, in Decode()
181 frame_size_ = 0; in DecodeAndOutputCurrentFrame()
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| D | vp8_decoder.h | 101 size_t frame_size_; variable
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| /external/webrtc/webrtc/test/ |
| D | frame_generator.cc | 65 frame_size_(CalcBufferSize(kI420, in YuvFileGenerator()
68 frame_buffer_(new uint8_t[frame_size_]), in YuvFileGenerator()
98 fread(frame_buffer_.get(), 1, frame_size_, files_[file_index_]); in ReadNextFrame()
99 if (bytes_read < frame_size_) { in ReadNextFrame()
103 bytes_read = fread(frame_buffer_.get(), 1, frame_size_, in ReadNextFrame()
105 assert(bytes_read >= frame_size_); in ReadNextFrame()
123 const size_t frame_size_; member in webrtc::test::__anon53024f1c0111::YuvFileGenerator
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| /external/webrtc/talk/media/base/ |
| D | testutils.h | 142 uint32_t frame_size() const { return frame_size_; } in frame_size()
154 uint32_t frame_size_; variable
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| D | testutils.cc | 235 frame_size_(0), in VideoCapturerListener()
255 frame_size_ = frame->data_size; in OnFrameCaptured()
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| /external/v8/src/deoptimizer/ |
| D | deoptimizer.h | 707 DCHECK(static_cast<uint32_t>(frame_size_) == frame_size_); in GetFrameSize()
708 return static_cast<uint32_t>(frame_size_); in GetFrameSize()
790 return offsetof(FrameDescription, frame_size_); in frame_size_offset()
809 uintptr_t frame_size_; // Number of bytes. variable
827 DCHECK(offset < frame_size_); in GetFrameSlotPointer()
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| D | deoptimizer.cc | 2044 : frame_size_(frame_size), in FrameDescription()
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| /external/webrtc/webrtc/sound/ |
| D | alsasoundsystem.cc | 84 frame_size_(frame_size), in AlsaStream()
218 return frame_size_; in frame_size()
224 size_t frame_size_; member in rtc::AlsaStream
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