Changed the WebRtcVad_Create() function to the more conventional format of returning the handle directly instead of an error code to take care of. In addition NULL was changed to nullptr in the files where it applied. Affected components: * AGC * VAD * NetEQ BUG=441, 3347 TESTED=locally on Linux and trybots R=kwiberg@webrtc.org Review URL: https://webrtc-codereview.appspot.com/51919004 Cr-Commit-Position: refs/heads/master@{#9291}
89 lines
2.3 KiB
C++
89 lines
2.3 KiB
C++
/*
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* Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "webrtc/modules/audio_coding/neteq/post_decode_vad.h"
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namespace webrtc {
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PostDecodeVad::~PostDecodeVad() {
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if (vad_instance_)
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WebRtcVad_Free(vad_instance_);
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}
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void PostDecodeVad::Enable() {
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if (!vad_instance_) {
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// Create the instance.
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vad_instance_ = WebRtcVad_Create();
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if (vad_instance_ == nullptr) {
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// Failed to create instance.
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Disable();
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return;
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}
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}
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Init();
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enabled_ = true;
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}
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void PostDecodeVad::Disable() {
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enabled_ = false;
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running_ = false;
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}
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void PostDecodeVad::Init() {
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running_ = false;
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if (vad_instance_) {
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WebRtcVad_Init(vad_instance_);
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WebRtcVad_set_mode(vad_instance_, kVadMode);
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running_ = true;
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}
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}
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void PostDecodeVad::Update(int16_t* signal, int length,
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AudioDecoder::SpeechType speech_type,
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bool sid_frame,
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int fs_hz) {
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if (!vad_instance_ || !enabled_) {
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return;
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}
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if (speech_type == AudioDecoder::kComfortNoise || sid_frame ||
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fs_hz > 16000) {
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// TODO(hlundin): Remove restriction on fs_hz.
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running_ = false;
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active_speech_ = true;
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sid_interval_counter_ = 0;
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} else if (!running_) {
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++sid_interval_counter_;
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}
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if (sid_interval_counter_ >= kVadAutoEnable) {
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Init();
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}
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if (length > 0 && running_) {
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int vad_sample_index = 0;
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active_speech_ = false;
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// Loop through frame sizes 30, 20, and 10 ms.
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for (int vad_frame_size_ms = 30; vad_frame_size_ms >= 10;
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vad_frame_size_ms -= 10) {
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int vad_frame_size_samples = vad_frame_size_ms * fs_hz / 1000;
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while (length - vad_sample_index >= vad_frame_size_samples) {
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int vad_return = WebRtcVad_Process(
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vad_instance_, fs_hz, &signal[vad_sample_index],
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vad_frame_size_samples);
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active_speech_ |= (vad_return == 1);
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vad_sample_index += vad_frame_size_samples;
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}
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}
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}
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}
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} // namespace webrtc
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