aleloi 6321b49a0d Move functionality out from AudioFrame and into AudioFrameOperations.
This CL is in preparation to move the AudioFrame into webrtc/api. The
AudioFrame is a POD type used for representing 10ms of audio. It
appears as a parameter and return value of interfaces being migrated
to webrtc/api, in particular AudioMixer.

Here, methods operator+=, operator>>=, Mute are
moved into a new target webrtc/audio/utility/audio_frame_operations,
and dependencies are changed to use
the new versions. The old AudioFrame methods are marked deprecated.

The audio frame utilities in webrtc/modules/utility:audio_frame_operations
are also moved to the new location.

TBR=kjellander@webrtc.org
BUG=webrtc:6548
NOPRESUBMIT=True

Review-Url: https://codereview.webrtc.org/2424173003
Cr-Commit-Position: refs/heads/master@{#15413}
2016-12-05 09:46:20 +00:00

93 lines
2.5 KiB
C++

/*
* 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 "webrtc/modules/audio_processing/vad/standalone_vad.h"
#include "webrtc/audio/utility/audio_frame_operations.h"
#include "webrtc/base/checks.h"
#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/typedefs.h"
namespace webrtc {
static const int kDefaultStandaloneVadMode = 3;
StandaloneVad::StandaloneVad(VadInst* vad)
: vad_(vad), buffer_(), index_(0), mode_(kDefaultStandaloneVadMode) {
}
StandaloneVad::~StandaloneVad() {
WebRtcVad_Free(vad_);
}
StandaloneVad* StandaloneVad::Create() {
VadInst* vad = WebRtcVad_Create();
if (!vad)
return nullptr;
int err = WebRtcVad_Init(vad);
err |= WebRtcVad_set_mode(vad, kDefaultStandaloneVadMode);
if (err != 0) {
WebRtcVad_Free(vad);
return nullptr;
}
return new StandaloneVad(vad);
}
int StandaloneVad::AddAudio(const int16_t* data, size_t length) {
if (length != kLength10Ms)
return -1;
if (index_ + length > kLength10Ms * kMaxNum10msFrames)
// Reset the buffer if it's full.
// TODO(ajm): Instead, consider just processing every 10 ms frame. Then we
// can forgo the buffering.
index_ = 0;
memcpy(&buffer_[index_], data, sizeof(int16_t) * length);
index_ += length;
return 0;
}
int StandaloneVad::GetActivity(double* p, size_t length_p) {
if (index_ == 0)
return -1;
const size_t num_frames = index_ / kLength10Ms;
if (num_frames > length_p)
return -1;
RTC_DCHECK_EQ(0, WebRtcVad_ValidRateAndFrameLength(kSampleRateHz, index_));
int activity = WebRtcVad_Process(vad_, kSampleRateHz, buffer_, index_);
if (activity < 0)
return -1;
else if (activity == 0)
p[0] = 0.01; // Arbitrary but small and non-zero.
else
p[0] = 0.5; // 0.5 is neutral values when combinned by other probabilities.
for (size_t n = 1; n < num_frames; n++)
p[n] = p[0];
// Reset the buffer to start from the beginning.
index_ = 0;
return activity;
}
int StandaloneVad::set_mode(int mode) {
if (mode < 0 || mode > 3)
return -1;
if (WebRtcVad_set_mode(vad_, mode) != 0)
return -1;
mode_ = mode;
return 0;
}
} // namespace webrtc