/external/webrtc/webrtc/modules/audio_coding/codecs/isac/ |
audio_decoder_isac_t.h | 29 size_t DecodePlc(size_t num_frames, int16_t* decoded) override;
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audio_decoder_isac_t_impl.h | 66 size_t AudioDecoderIsacT<T>::DecodePlc(size_t num_frames, int16_t* decoded) { 67 return T::DecodePlc(isac_state_, decoded, num_frames);
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/external/webrtc/webrtc/modules/audio_processing/ |
level_estimator_impl.cc | 39 rms_->Process(audio->channels_const()[i], audio->num_frames());
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splitting_filter.h | 48 SplittingFilter(size_t num_channels, size_t num_bands, size_t num_frames);
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audio_processing_impl.cc | 352 formats_.api_format.reverse_output_stream().num_frames() == 0 353 ? formats_.rev_proc_format.num_frames() 354 : formats_.api_format.reverse_output_stream().num_frames(); 357 formats_.api_format.reverse_input_stream().num_frames(), 359 formats_.rev_proc_format.num_frames(), 365 formats_.api_format.reverse_input_stream().num_frames(), 367 formats_.api_format.reverse_output_stream().num_frames()); 376 new AudioBuffer(formats_.api_format.input_stream().num_frames(), 378 capture_nonlocked_.fwd_proc_format.num_frames(), 380 formats_.api_format.output_stream().num_frames())); [all...] |
audio_buffer.cc | 35 size_t NumBandsFromSamplesPerChannel(size_t num_frames) { 37 if (num_frames == kSamplesPer32kHzChannel || 38 num_frames == kSamplesPer48kHzChannel) { 39 num_bands = rtc::CheckedDivExact(num_frames, kSamplesPer16kHzChannel); 107 assert(stream_config.num_frames() == input_num_frames_); 152 assert(stream_config.num_frames() == output_num_frames_); 352 size_t AudioBuffer::num_frames() const { function in class:webrtc::AudioBuffer
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/external/webrtc/webrtc/modules/audio_processing/beamformer/ |
nonlinear_beamformer_test.cc | 72 Deinterleave(&interleaved[0], in_buf.num_frames(), 77 Interleave(out_buf.channels(), out_buf.num_frames(),
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/external/webrtc/webrtc/modules/audio_processing/test/ |
debug_dump_test.cc | 34 if (!buffer_ref.get() || buffer_ref->num_frames() != config.num_frames() || 36 buffer_ref.reset(new ChannelBuffer<float>(config.num_frames(), 128 input_(new ChannelBuffer<float>(input_config_.num_frames(), 130 reverse_(new ChannelBuffer<float>(reverse_config_.num_frames(), 132 output_(new ChannelBuffer<float>(output_config_.num_frames(), 215 const size_t num_frames = config.num_frames(); local 218 std::vector<int16_t> signal(channels * num_frames); 220 audio->Read(num_frames * channels, &signal[0]) [all...] |
test_utils.cc | 48 Deinterleave(&interleaved_[0], buffer->num_frames(), buffer->num_channels(), 59 Interleave(buffer.channels(), buffer.num_frames(), buffer.num_channels(),
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/system/core/healthd/ |
healthd_mode_charger.cpp | 99 int num_frames; member in struct:animation 165 .num_frames = ARRAY_SIZE(batt_anim_frames), 311 if (batt_anim->num_frames != 0) { 326 if (batt_anim->capacity < 0 || batt_anim->num_frames == 0) 392 if (batt_prop && batt_prop->batteryLevel >= 0 && batt_anim->num_frames != 0) { 396 for (i = 1; i < batt_anim->num_frames; i++) { 419 if (batt_anim->num_frames == 0 || batt_anim->capacity < 0) { 438 while (batt_anim->cur_frame < batt_anim->num_frames && 441 if (batt_anim->cur_frame >= batt_anim->num_frames) { 708 charger->batt_anim->num_frames = 0 [all...] |
/external/webrtc/webrtc/modules/video_coding/codecs/test/ |
videoprocessor_integrationtest.cc | 65 int num_frames; member in struct:webrtc::RateProfile 232 void ResetRateControlMetrics(int num_frames) { 254 num_frames_to_hit_target_ = num_frames; 441 // Process each frame, up to |num_frames|. 442 int num_frames = rate_profile.num_frames; local 449 frame_number < num_frames) { 490 EXPECT_EQ(num_frames, frame_number); 491 EXPECT_EQ(num_frames + 1, static_cast<int>(stats_.stats_.size())); 590 rate_profile.num_frames = kNbrFramesShort [all...] |
/external/webrtc/webrtc/common_audio/ |
audio_converter_unittest.cc | 40 EXPECT_EQ(ref.num_frames(), test.num_frames()); 55 delay <= std::min(expected_delay + 1, ref.num_frames()); 61 for (size_t j = 0; j < ref.num_frames() - delay; ++j) { 69 const size_t length = ref.num_channels() * (ref.num_frames() - delay);
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lapped_transform.h | 97 size_t num_frames,
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/external/webrtc/webrtc/modules/audio_coding/test/ |
delay_test.cc | 181 int num_frames = 0; local 187 while (num_frames < (duration_sec * 100)) { 193 if ((num_frames & 0x3F) == 0x3F) { 217 if (num_frames > 10) 220 ++num_frames;
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/bionic/libc/malloc_debug/ |
FreeTrackData.cpp | 61 backtrace_log(&back_header->frames[0], back_header->num_frames); 112 back_header->num_frames = backtrace_get(&back_header->frames[0], backtrace_num_frames_); 140 backtrace_log(&back_iter->second->frames[0], back_iter->second->num_frames);
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/external/autotest/client/site_tests/graphics_SanAngeles/src/ |
app-linux.c | 227 int num_frames = 0; local 255 num_frames++; 261 fprintf(stdout, "frame_rate = %.1f\n", num_frames / total_time);
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/external/webrtc/webrtc/tools/agc/ |
activity_metric.cc | 111 if (features.num_frames > 0) { 119 for (size_t n = 0; n < features.num_frames; n++) { 128 for (size_t n = 0; n < features.num_frames; n++) { 142 return static_cast<int>(features.num_frames); 236 int num_frames = 0; local 277 num_frames++;
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/external/libunwind/include/ |
libunwind-dynamic.h | 193 #define _U_dyn_op_pop_frames(op, qp, when, num_frames) \ 194 (*(op) = _U_dyn_op (UNW_DYN_POP_FRAMES, (qp), (when), 0, (num_frames)))
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/external/webrtc/webrtc/modules/audio_coding/codecs/ |
audio_decoder.h | 65 // memory allocated in |decoded| should accommodate |num_frames| frames. 66 virtual size_t DecodePlc(size_t num_frames, int16_t* decoded);
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/external/webrtc/webrtc/modules/audio_processing/vad/ |
pitch_based_vad_unittest.cc | 50 audio_features.num_frames = 1;
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/system/core/libmemunreachable/include/memunreachable/ |
memunreachable.h | 45 size_t num_frames; member in struct:Leak::Backtrace
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/external/libyuv/files/util/ |
convert.cc | 37 int num_frames = 0; // Number of frames to convert. variable 100 num_frames = atoi(argv[++c]); // NOLINT 122 if (num_frames < 0) { 265 if (num_frames && number_of_frames >= num_frames)
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/external/webrtc/webrtc/modules/audio_device/android/ |
audio_device_unittest.cc | 94 virtual void Write(const void* source, size_t num_frames) = 0; 95 virtual void Read(void* destination, size_t num_frames) = 0; 125 void Write(const void* source, size_t num_frames) override {} 128 // |num_frames| (<=> 10ms) to the |destination| byte buffer. 129 void Read(void* destination, size_t num_frames) override { 132 num_frames * sizeof(int16_t)); 133 file_pos_ += num_frames; 175 // Allocate new memory, copy |num_frames| samples from |source| into memory 178 void Write(const void* source, size_t num_frames) override { 179 ASSERT_EQ(num_frames, frames_per_buffer_) [all...] |
/external/webrtc/webrtc/modules/audio_device/ios/ |
audio_device_unittest_ios.cc | 96 virtual void Write(const void* source, size_t num_frames) = 0; 97 virtual void Read(void* destination, size_t num_frames) = 0; 127 void Write(const void* source, size_t num_frames) override {} 130 // |num_frames| (<=> 10ms) to the |destination| byte buffer. 131 void Read(void* destination, size_t num_frames) override { 133 num_frames * sizeof(int16_t)); 134 file_pos_ += num_frames; 174 // Allocate new memory, copy |num_frames| samples from |source| into memory 177 void Write(const void* source, size_t num_frames) override { 178 ASSERT_EQ(num_frames, frames_per_buffer_) [all...] |
/external/opencv3/3rdparty/libwebp/webp/ |
demux.h | 128 int num_frames; // equivalent to WEBP_FF_FRAME_COUNT. member in struct:WebPIterator
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