Searched refs:S16ToFloat (Results 1 – 10 of 10) sorted by relevance
/external/webrtc/common_audio/ |
D | audio_util.cc | 20 void S16ToFloat(const int16_t* src, size_t size, float* dest) { in S16ToFloat() function 22 dest[i] = S16ToFloat(src[i]); in S16ToFloat()
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D | audio_util_unittest.cc | 34 TEST(AudioUtilTest, S16ToFloat) { in TEST() argument 41 S16ToFloat(kInput, kSize, output); in TEST()
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D | wav_file.cc | 232 converted_samples[j] = S16ToFloat(samples[i + j]); in WriteSamples()
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/external/webrtc/common_audio/include/ |
D | audio_util.h | 35 static inline float S16ToFloat(int16_t v) { in S16ToFloat() function 67 void S16ToFloat(const int16_t* src, size_t size, float* dest);
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/external/webrtc/modules/audio_coding/codecs/opus/ |
D | opus_bandwidth_unittest.cc | 94 S16ToFloat(decoded.data(), decoded.size(), decoded_float.data()); in EncodedPowerRatio()
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/external/webrtc/modules/audio_processing/transient/ |
D | transient_suppression_test.cc | 111 S16ToFloat(ibuf.get(), audio_buffer_size, reference_buffer); in ReadBuffers()
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/external/webrtc/modules/audio_processing/test/ |
D | audio_processing_simulator.h | 52 S16ToFloat(data.data(), tmp.size(), tmp.data()); in CopyTo()
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D | debug_dump_test.cc | 250 buffer[channel][i] = S16ToFloat(signal[i * channels + channel]); in ReadAndDeinterleave()
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/external/webrtc/common_audio/resampler/ |
D | push_sinc_resampler_unittest.cc | 182 S16ToFloat(destination_int.get(), output_block_size, in ResampleTest()
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/external/webrtc/modules/audio_processing/ |
D | audio_processing_unittest.cc | 88 S16ToFloat(cb_int.channels()[i], cb->num_frames(), cb->channels()[i]); in ConvertToFloat() 341 S16ToFloat(int_data, frame_size, float_data); in ReadChunk()
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