From 26a029d407be480d791972afb5975cf62c9360a6 Mon Sep 17 00:00:00 2001 From: Daniel Baumann Date: Fri, 19 Apr 2024 02:47:55 +0200 Subject: Adding upstream version 124.0.1. Signed-off-by: Daniel Baumann --- .../modules/audio_coding/neteq/post_decode_vad.cc | 90 ++++++++++++++++++++++ 1 file changed, 90 insertions(+) create mode 100644 third_party/libwebrtc/modules/audio_coding/neteq/post_decode_vad.cc (limited to 'third_party/libwebrtc/modules/audio_coding/neteq/post_decode_vad.cc') diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/post_decode_vad.cc b/third_party/libwebrtc/modules/audio_coding/neteq/post_decode_vad.cc new file mode 100644 index 0000000000..9999d6764b --- /dev/null +++ b/third_party/libwebrtc/modules/audio_coding/neteq/post_decode_vad.cc @@ -0,0 +1,90 @@ +/* + * Copyright (c) 2013 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/post_decode_vad.h" + +namespace webrtc { + +PostDecodeVad::~PostDecodeVad() { + if (vad_instance_) + WebRtcVad_Free(vad_instance_); +} + +void PostDecodeVad::Enable() { + if (!vad_instance_) { + // Create the instance. + vad_instance_ = WebRtcVad_Create(); + if (vad_instance_ == nullptr) { + // Failed to create instance. + Disable(); + return; + } + } + Init(); + enabled_ = true; +} + +void PostDecodeVad::Disable() { + enabled_ = false; + running_ = false; +} + +void PostDecodeVad::Init() { + running_ = false; + if (vad_instance_) { + WebRtcVad_Init(vad_instance_); + WebRtcVad_set_mode(vad_instance_, kVadMode); + running_ = true; + } +} + +void PostDecodeVad::Update(int16_t* signal, + size_t length, + AudioDecoder::SpeechType speech_type, + bool sid_frame, + int fs_hz) { + if (!vad_instance_ || !enabled_) { + return; + } + + if (speech_type == AudioDecoder::kComfortNoise || sid_frame || + fs_hz > 16000) { + // TODO(hlundin): Remove restriction on fs_hz. + running_ = false; + active_speech_ = true; + sid_interval_counter_ = 0; + } else if (!running_) { + ++sid_interval_counter_; + } + + if (sid_interval_counter_ >= kVadAutoEnable) { + Init(); + } + + if (length > 0 && running_) { + size_t vad_sample_index = 0; + active_speech_ = false; + // Loop through frame sizes 30, 20, and 10 ms. + for (int vad_frame_size_ms = 30; vad_frame_size_ms >= 10; + vad_frame_size_ms -= 10) { + size_t vad_frame_size_samples = + static_cast(vad_frame_size_ms * fs_hz / 1000); + while (length - vad_sample_index >= vad_frame_size_samples) { + int vad_return = + WebRtcVad_Process(vad_instance_, fs_hz, &signal[vad_sample_index], + vad_frame_size_samples); + active_speech_ |= (vad_return == 1); + vad_sample_index += vad_frame_size_samples; + } + } + } +} + +} // namespace webrtc -- cgit v1.2.3