mirror of
https://github.com/mollyim/webrtc.git
synced 2025-05-13 22:00:47 +01:00
92ea95e34a
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}
63 lines
2 KiB
C++
63 lines
2 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 "modules/audio_coding/acm2/acm_resampler.h"
|
|
|
|
#include <assert.h>
|
|
#include <string.h>
|
|
|
|
#include "common_audio/resampler/include/resampler.h"
|
|
#include "rtc_base/logging.h"
|
|
|
|
namespace webrtc {
|
|
namespace acm2 {
|
|
|
|
ACMResampler::ACMResampler() {
|
|
}
|
|
|
|
ACMResampler::~ACMResampler() {
|
|
}
|
|
|
|
int ACMResampler::Resample10Msec(const int16_t* in_audio,
|
|
int in_freq_hz,
|
|
int out_freq_hz,
|
|
size_t num_audio_channels,
|
|
size_t out_capacity_samples,
|
|
int16_t* out_audio) {
|
|
size_t in_length = in_freq_hz * num_audio_channels / 100;
|
|
if (in_freq_hz == out_freq_hz) {
|
|
if (out_capacity_samples < in_length) {
|
|
assert(false);
|
|
return -1;
|
|
}
|
|
memcpy(out_audio, in_audio, in_length * sizeof(int16_t));
|
|
return static_cast<int>(in_length / num_audio_channels);
|
|
}
|
|
|
|
if (resampler_.InitializeIfNeeded(in_freq_hz, out_freq_hz,
|
|
num_audio_channels) != 0) {
|
|
LOG(LS_ERROR) << "InitializeIfNeeded(" << in_freq_hz << ", " << out_freq_hz
|
|
<< ", " << num_audio_channels << ") failed.";
|
|
return -1;
|
|
}
|
|
|
|
int out_length =
|
|
resampler_.Resample(in_audio, in_length, out_audio, out_capacity_samples);
|
|
if (out_length == -1) {
|
|
LOG(LS_ERROR) << "Resample(" << in_audio << ", " << in_length << ", "
|
|
<< out_audio << ", " << out_capacity_samples << ") failed.";
|
|
return -1;
|
|
}
|
|
|
|
return static_cast<int>(out_length / num_audio_channels);
|
|
}
|
|
|
|
} // namespace acm2
|
|
} // namespace webrtc
|