48af652ea5
ACM1 code is wrapped in namespace acm1. Inculde paths and define guards of ACM2 source codes are corrected. gypi file of ACM2 is changed so that ACM1 will later on depends on ACM2. BUG= R=andrew@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2206004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4743 4adac7df-926f-26a2-2b94-8c16560cd09d
67 lines
2.2 KiB
C++
67 lines
2.2 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 "webrtc/modules/audio_coding/main/acm2/acm_resampler.h"
|
|
|
|
#include <string.h>
|
|
|
|
#include "webrtc/common_audio/resampler/include/resampler.h"
|
|
#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
|
|
#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
|
|
#include "webrtc/system_wrappers/interface/trace.h"
|
|
|
|
namespace webrtc {
|
|
|
|
ACMResampler::ACMResampler()
|
|
: resampler_crit_sect_(CriticalSectionWrapper::CreateCriticalSection()) {
|
|
}
|
|
|
|
ACMResampler::~ACMResampler() {
|
|
delete resampler_crit_sect_;
|
|
}
|
|
|
|
int ACMResampler::Resample10Msec(const int16_t* in_audio,
|
|
int in_freq_hz,
|
|
int out_freq_hz,
|
|
int num_audio_channels,
|
|
int16_t* out_audio) {
|
|
CriticalSectionScoped cs(resampler_crit_sect_);
|
|
|
|
if (in_freq_hz == out_freq_hz) {
|
|
size_t length = static_cast<size_t>(in_freq_hz * num_audio_channels / 100);
|
|
memcpy(out_audio, in_audio, length * sizeof(int16_t));
|
|
return static_cast<int16_t>(in_freq_hz / 100);
|
|
}
|
|
|
|
// |maxLen| is maximum number of samples for 10ms at 48kHz.
|
|
int max_len = 480 * num_audio_channels;
|
|
int length_in = (in_freq_hz / 100) * num_audio_channels;
|
|
int out_len;
|
|
|
|
ResamplerType type = (num_audio_channels == 1) ? kResamplerSynchronous :
|
|
kResamplerSynchronousStereo;
|
|
|
|
if (resampler_.ResetIfNeeded(in_freq_hz, out_freq_hz, type) < 0) {
|
|
WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, 0,
|
|
"Error in reset of resampler");
|
|
return -1;
|
|
}
|
|
|
|
if (resampler_.Push(in_audio, length_in, out_audio, max_len, out_len) < 0) {
|
|
WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, 0,
|
|
"Error in resampler: resampler.Push");
|
|
return -1;
|
|
}
|
|
|
|
return out_len / num_audio_channels;
|
|
}
|
|
|
|
} // namespace webrtc
|