webrtc_m130/modules/audio_mixer/sine_wave_generator.cc
Mirko Bonadei 92ea95e34a Fixing WebRTC after moving from src/webrtc to src/
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC
from src/webrtc to src/ (in order to preserve an healthy git history).
This CL takes care of fixing header guards, #include paths, etc...

NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org


Bug: chromium:611808
Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578
Reviewed-on: https://webrtc-review.googlesource.com/1561
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19846}
2017-09-15 05:02:56 +00:00

35 lines
1.1 KiB
C++

/*
* Copyright (c) 2017 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_mixer/sine_wave_generator.h"
#include <math.h>
#include "rtc_base/safe_conversions.h"
namespace webrtc {
namespace {
constexpr float kPi = 3.14159265f;
} // namespace
void SineWaveGenerator::GenerateNextFrame(AudioFrame* frame) {
RTC_DCHECK(frame);
int16_t* frame_data = frame->mutable_data();
for (size_t i = 0; i < frame->samples_per_channel_; ++i) {
for (size_t ch = 0; ch < frame->num_channels_; ++ch) {
frame_data[frame->num_channels_ * i + ch] =
rtc::saturated_cast<int16_t>(amplitude_ * sinf(phase_));
}
phase_ += wave_frequency_hz_ * 2 * kPi / frame->sample_rate_hz_;
}
}
} // namespace webrtc