ff761fba82
This changes the following module directories:
* webrtc/modules/audio_conference_mixer/interface
* webrtc/modules/interface
* webrtc/modules/media_file/interface
* webrtc/modules/rtp_rtcp/interface
* webrtc/modules/utility/interface
To avoid breaking downstream, I followed this recipe:
1. Copy the interface dir to a new sibling directory: include
2. Update the header guards in the include directory to match the style guide.
3. Update the header guards in the interface directory to match the ones in include. This is required to avoid getting redefinitions in the not-yet-updated downstream code.
4. Add a pragma warning in the header files in the interface dir. Example:
#pragma message("WARNING: webrtc/modules/interface is DEPRECATED; "
"use webrtc/modules/include")
5. Search for all source references to webrtc/modules/interface and update them to webrtc/modules/include (*.c*,*.h,*.mm,*.S)
6. Update all GYP+GN files. This required manual inspection since many subdirectories of webrtc/modules referenced the interface dir using ../interface etc(*.gyp*,*.gn*)
BUG=5095
TESTED=Passing compile-trybots with --clobber flag:
git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel -m tryserver.webrtc
R=stefan@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1417683006 .
Cr-Commit-Position: refs/heads/master@{#10500}
94 lines
2.5 KiB
C++
94 lines
2.5 KiB
C++
/*
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* Copyright (c) 2012 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_processing/vad/standalone_vad.h"
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#include <assert.h>
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#include "webrtc/modules/include/module_common_types.h"
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#include "webrtc/modules/utility/include/audio_frame_operations.h"
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#include "webrtc/typedefs.h"
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namespace webrtc {
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static const int kDefaultStandaloneVadMode = 3;
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StandaloneVad::StandaloneVad(VadInst* vad)
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: vad_(vad), buffer_(), index_(0), mode_(kDefaultStandaloneVadMode) {
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}
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StandaloneVad::~StandaloneVad() {
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WebRtcVad_Free(vad_);
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}
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StandaloneVad* StandaloneVad::Create() {
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VadInst* vad = WebRtcVad_Create();
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if (!vad)
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return nullptr;
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int err = WebRtcVad_Init(vad);
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err |= WebRtcVad_set_mode(vad, kDefaultStandaloneVadMode);
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if (err != 0) {
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WebRtcVad_Free(vad);
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return nullptr;
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}
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return new StandaloneVad(vad);
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}
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int StandaloneVad::AddAudio(const int16_t* data, size_t length) {
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if (length != kLength10Ms)
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return -1;
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if (index_ + length > kLength10Ms * kMaxNum10msFrames)
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// Reset the buffer if it's full.
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// TODO(ajm): Instead, consider just processing every 10 ms frame. Then we
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// can forgo the buffering.
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index_ = 0;
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memcpy(&buffer_[index_], data, sizeof(int16_t) * length);
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index_ += length;
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return 0;
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}
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int StandaloneVad::GetActivity(double* p, size_t length_p) {
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if (index_ == 0)
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return -1;
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const size_t num_frames = index_ / kLength10Ms;
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if (num_frames > length_p)
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return -1;
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assert(WebRtcVad_ValidRateAndFrameLength(kSampleRateHz, index_) == 0);
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int activity = WebRtcVad_Process(vad_, kSampleRateHz, buffer_, index_);
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if (activity < 0)
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return -1;
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else if (activity == 0)
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p[0] = 0.01; // Arbitrary but small and non-zero.
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else
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p[0] = 0.5; // 0.5 is neutral values when combinned by other probabilities.
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for (size_t n = 1; n < num_frames; n++)
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p[n] = p[0];
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// Reset the buffer to start from the beginning.
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index_ = 0;
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return activity;
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}
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int StandaloneVad::set_mode(int mode) {
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if (mode < 0 || mode > 3)
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return -1;
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if (WebRtcVad_set_mode(vad_, mode) != 0)
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return -1;
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mode_ = mode;
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return 0;
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}
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} // namespace webrtc
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