| /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
|
