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author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-19 00:47:55 +0000 |
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committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-19 00:47:55 +0000 |
commit | 26a029d407be480d791972afb5975cf62c9360a6 (patch) | |
tree | f435a8308119effd964b339f76abb83a57c29483 /third_party/libwebrtc/modules/audio_coding/neteq/accelerate.cc | |
parent | Initial commit. (diff) | |
download | firefox-26a029d407be480d791972afb5975cf62c9360a6.tar.xz firefox-26a029d407be480d791972afb5975cf62c9360a6.zip |
Adding upstream version 124.0.1.upstream/124.0.1
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to '')
-rw-r--r-- | third_party/libwebrtc/modules/audio_coding/neteq/accelerate.cc | 105 |
1 files changed, 105 insertions, 0 deletions
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/accelerate.cc b/third_party/libwebrtc/modules/audio_coding/neteq/accelerate.cc new file mode 100644 index 0000000000..06a38cc534 --- /dev/null +++ b/third_party/libwebrtc/modules/audio_coding/neteq/accelerate.cc @@ -0,0 +1,105 @@ +/* + * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#include "modules/audio_coding/neteq/accelerate.h" + +#include "api/array_view.h" +#include "modules/audio_coding/neteq/audio_multi_vector.h" + +namespace webrtc { + +Accelerate::ReturnCodes Accelerate::Process(const int16_t* input, + size_t input_length, + bool fast_accelerate, + AudioMultiVector* output, + size_t* length_change_samples) { + // Input length must be (almost) 30 ms. + static const size_t k15ms = 120; // 15 ms = 120 samples at 8 kHz sample rate. + if (num_channels_ == 0 || + input_length / num_channels_ < (2 * k15ms - 1) * fs_mult_) { + // Length of input data too short to do accelerate. Simply move all data + // from input to output. + output->PushBackInterleaved( + rtc::ArrayView<const int16_t>(input, input_length)); + return kError; + } + return TimeStretch::Process(input, input_length, fast_accelerate, output, + length_change_samples); +} + +void Accelerate::SetParametersForPassiveSpeech(size_t /*len*/, + int16_t* best_correlation, + size_t* /*peak_index*/) const { + // When the signal does not contain any active speech, the correlation does + // not matter. Simply set it to zero. + *best_correlation = 0; +} + +Accelerate::ReturnCodes Accelerate::CheckCriteriaAndStretch( + const int16_t* input, + size_t input_length, + size_t peak_index, + int16_t best_correlation, + bool active_speech, + bool fast_mode, + AudioMultiVector* output) const { + // Check for strong correlation or passive speech. + // Use 8192 (0.5 in Q14) in fast mode. + const int correlation_threshold = fast_mode ? 8192 : kCorrelationThreshold; + if ((best_correlation > correlation_threshold) || !active_speech) { + // Do accelerate operation by overlap add. + + // Pre-calculate common multiplication with `fs_mult_`. + // 120 corresponds to 15 ms. + size_t fs_mult_120 = fs_mult_ * 120; + + if (fast_mode) { + // Fit as many multiples of `peak_index` as possible in fs_mult_120. + // TODO(henrik.lundin) Consider finding multiple correlation peaks and + // pick the one with the longest correlation lag in this case. + peak_index = (fs_mult_120 / peak_index) * peak_index; + } + + RTC_DCHECK_GE(fs_mult_120, peak_index); // Should be handled in Process(). + // Copy first part; 0 to 15 ms. + output->PushBackInterleaved( + rtc::ArrayView<const int16_t>(input, fs_mult_120 * num_channels_)); + // Copy the `peak_index` starting at 15 ms to `temp_vector`. + AudioMultiVector temp_vector(num_channels_); + temp_vector.PushBackInterleaved(rtc::ArrayView<const int16_t>( + &input[fs_mult_120 * num_channels_], peak_index * num_channels_)); + // Cross-fade `temp_vector` onto the end of `output`. + output->CrossFade(temp_vector, peak_index); + // Copy the last unmodified part, 15 ms + pitch period until the end. + output->PushBackInterleaved(rtc::ArrayView<const int16_t>( + &input[(fs_mult_120 + peak_index) * num_channels_], + input_length - (fs_mult_120 + peak_index) * num_channels_)); + + if (active_speech) { + return kSuccess; + } else { + return kSuccessLowEnergy; + } + } else { + // Accelerate not allowed. Simply move all data from decoded to outData. + output->PushBackInterleaved( + rtc::ArrayView<const int16_t>(input, input_length)); + return kNoStretch; + } +} + +Accelerate* AccelerateFactory::Create( + int sample_rate_hz, + size_t num_channels, + const BackgroundNoise& background_noise) const { + return new Accelerate(sample_rate_hz, num_channels, background_noise); +} + +} // namespace webrtc |