| /external/chromium_org/third_party/webrtc/modules/video_coding/codecs/test_framework/ |
| unit_test.cc | 258 unsigned int frameLength = 0;
263 while (frameLength == 0)
283 frameLength = WaitForDecodedFrame();
288 EXPECT_TRUE(frameLength == _lengthSourceFrame);
327 int frameLength = WaitForDecodedFrame();
329 return ret == WEBRTC_VIDEO_CODEC_OK ? frameLength : ret;
346 unsigned int frameLength = WaitForDecodedFrame();
347 assert(ret == WEBRTC_VIDEO_CODEC_OK && (frameLength == 0 || frameLength
349 EXPECT_TRUE(ret == WEBRTC_VIDEO_CODEC_OK && (frameLength == 0 || frameLengt
[all...] |
| /external/chromium_org/third_party/webrtc/modules/audio_coding/codecs/g711/test/ |
| testG711.cc | 48 int16_t framelength = 80; local
72 printf("./testG711.exe framelength law infile outfile \n\n");
73 printf("framelength: Framelength in samples.\n");
88 framelength = atoi(argv[1]);
125 endfile = readframe(shortdata, inp, framelength);
130 stream_len = WebRtcG711_EncodeA(NULL, shortdata, framelength, streamdata);
142 stream_len = WebRtcG711_EncodeU(NULL, shortdata, framelength, streamdata);
161 if (fwrite(decoded, sizeof(short), framelength, outp) !=
162 static_cast<size_t>(framelength)) {
[all...] |
| /external/chromium_org/third_party/webrtc/modules/audio_coding/codecs/g722/test/ |
| testG722.cc | 52 int16_t framelength = 160; local
74 printf("./testG722.exe framelength infile outbitfile outspeechfile \n\n");
76 printf("framelength : Framelength in samples.\n\n");
85 framelength = atoi(argv[1]);
120 endfile = readframe(shortdata, inp, framelength);
126 stream_len = WebRtcG722_Encode((G722EncInst *)G722enc_inst, shortdata, framelength, streamdata);
142 if (fwrite(decoded, sizeof(short), framelength,
143 outp) != static_cast<size_t>(framelength)) {
152 length_file = ((double)framecnt*(double)framelength/16000)
[all...] |
| /external/chromium_org/third_party/webrtc/test/testsupport/mock/ |
| mock_frame_reader.h | 26 MOCK_METHOD0(FrameLength, size_t());
|
| mock_frame_writer.h | 26 MOCK_METHOD0(FrameLength, size_t());
|
| /external/webrtc/test/testsupport/mock/ |
| mock_frame_reader.h | 26 MOCK_METHOD0(FrameLength, int());
|
| mock_frame_writer.h | 26 MOCK_METHOD0(FrameLength, int());
|
| /frameworks/av/media/libstagefright/rtsp/ |
| ARTPAssembler.cpp | 95 unsigned frameLength = nal->size() + 7;
106 dst[3] = ((channelConfig & 3) << 6) | (frameLength >> 11);
108 dst[4] = (frameLength >> 3) & 0xff;
109 dst[5] = (frameLength & 7) << 5;
|
| /external/chromium_org/third_party/webrtc/test/testsupport/ |
| frame_writer.h | 42 virtual size_t FrameLength() = 0;
59 virtual size_t FrameLength() OVERRIDE;
|
| frame_reader.h | 44 virtual size_t FrameLength() = 0;
61 virtual size_t FrameLength() OVERRIDE;
|
| frame_writer.cc | 51 size_t FrameWriterImpl::FrameLength() { return frame_length_in_bytes_; }
|
| frame_writer_unittest.cc | 43 ASSERT_EQ(kFrameLength, frame_writer.FrameLength());
|
| frame_reader.cc | 81 size_t FrameReaderImpl::FrameLength() { return frame_length_in_bytes_; }
|
| frame_reader_unittest.cc | 52 ASSERT_EQ(kFrameLength, frame_reader.FrameLength());
|
| /external/webrtc/src/modules/audio_coding/codecs/isac/fix/source/ |
| entropy_coding.h | 95 WebRtc_Word16 *framelength);
98 int WebRtcIsacfix_EncodeFrameLen(WebRtc_Word16 framelength,
|
| decode_bwe.c | 38 /* decode framelength */
|
| /external/webrtc/test/testsupport/ |
| frame_writer.h | 41 virtual int FrameLength() = 0;
58 int FrameLength() { return frame_length_in_bytes_; }
|
| frame_reader.h | 43 virtual int FrameLength() = 0;
60 int FrameLength() { return frame_length_in_bytes_; }
|
| frame_writer_unittest.cc | 43 ASSERT_EQ(kFrameLength, frame_writer.FrameLength());
|
| /external/aac/libAACenc/src/ |
| transform.cpp | 107 const INT frameLength,
134 tl = frameLength;
140 int offset = (windowShape == LOL_WINDOW) ? ((frameLength * 3)>>2) : 0;
141 fl = frameLength - offset;
142 fr = frameLength - offset;
146 fl = frameLength >> 3;
147 fr = frameLength;
150 fl = frameLength;
151 fr = frameLength >> 3;
154 fl = fr = frameLength >> 3
[all...] |
| /external/chromium_org/third_party/webrtc/modules/audio_coding/codecs/isac/fix/source/ |
| decode_bwe.c | 38 /* decode framelength */
|
| entropy_coding.h | 95 int16_t *framelength);
98 int WebRtcIsacfix_EncodeFrameLen(int16_t framelength,
|
| /frameworks/av/include/media/stagefright/ |
| AACWriter.h | 69 status_t writeAdtsHeader(uint32_t frameLength);
|
| /frameworks/wilhelm/tools/permute/ |
| permute.c | 79 static unsigned split(State *s, unsigned frameStart, unsigned frameLength, unsigned segmentBudget)
81 if (frameLength <= 0)
84 if ((frameLength <= s->mMinSegmentLengthFrames*2) || (segmentBudget <= 1)) {
88 seg->mFrameLength = frameLength;
93 unsigned slop = frameLength - s->mMinSegmentLengthFrames*2;
102 assert(leftLength + rightLength == frameLength);
|
| /external/chromium_org/third_party/webrtc/modules/audio_coding/codecs/isac/main/source/ |
| decode_bwe.c | 36 /* decode framelength and BW estimation */
|
