7959e16cc2
nack{.cc, .h, _unittest.cc} are basically copies from main/source/ folder, with cpplint warning cleaned up.
BUG=
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/2190009
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4736 4adac7df-926f-26a2-2b94-8c16560cd09d
67 lines
2.2 KiB
C++
67 lines
2.2 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_coding/main/source/acm_resampler.h"
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#include <string.h>
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#include "webrtc/common_audio/resampler/include/resampler.h"
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#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
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#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
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#include "webrtc/system_wrappers/interface/trace.h"
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namespace webrtc {
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ACMResampler::ACMResampler()
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: resampler_crit_sect_(CriticalSectionWrapper::CreateCriticalSection()) {
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}
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ACMResampler::~ACMResampler() {
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delete resampler_crit_sect_;
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}
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int ACMResampler::Resample10Msec(const int16_t* in_audio,
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int in_freq_hz,
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int out_freq_hz,
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int num_audio_channels,
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int16_t* out_audio) {
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CriticalSectionScoped cs(resampler_crit_sect_);
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if (in_freq_hz == out_freq_hz) {
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size_t length = static_cast<size_t>(in_freq_hz * num_audio_channels / 100);
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memcpy(out_audio, in_audio, length * sizeof(int16_t));
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return static_cast<int16_t>(in_freq_hz / 100);
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}
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// |maxLen| is maximum number of samples for 10ms at 48kHz.
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int max_len = 480 * num_audio_channels;
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int length_in = (in_freq_hz / 100) * num_audio_channels;
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int out_len;
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ResamplerType type = (num_audio_channels == 1) ? kResamplerSynchronous :
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kResamplerSynchronousStereo;
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if (resampler_.ResetIfNeeded(in_freq_hz, out_freq_hz, type) < 0) {
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WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, 0,
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"Error in reset of resampler");
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return -1;
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}
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if (resampler_.Push(in_audio, length_in, out_audio, max_len, out_len) < 0) {
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WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, 0,
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"Error in resampler: resampler.Push");
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return -1;
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}
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return out_len / num_audio_channels;
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}
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} // namespace webrtc
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