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
64 lines
2.0 KiB
C++
64 lines
2.0 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/push_resampler.h"
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#include "webrtc/system_wrappers/interface/logging.h"
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namespace webrtc {
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namespace acm1 {
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ACMResampler::ACMResampler() {
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}
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ACMResampler::~ACMResampler() {
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}
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int16_t ACMResampler::Resample10Msec(const int16_t* in_audio,
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int32_t in_freq_hz,
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int16_t* out_audio,
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int32_t out_freq_hz,
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uint8_t num_audio_channels) {
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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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// |max_length| is the maximum number of samples for 10ms at 48kHz.
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// TODO(turajs): is this actually the capacity of the |out_audio| buffer?
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int max_length = 480 * num_audio_channels;
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int in_length = in_freq_hz / 100 * num_audio_channels;
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if (resampler_.InitializeIfNeeded(in_freq_hz, out_freq_hz,
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num_audio_channels) != 0) {
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LOG_FERR3(LS_ERROR, InitializeIfNeeded, in_freq_hz, out_freq_hz,
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num_audio_channels);
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return -1;
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}
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int out_length = resampler_.Resample(in_audio, in_length, out_audio,
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max_length);
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if (out_length == -1) {
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LOG_FERR4(LS_ERROR, Resample, in_audio, in_length, out_audio, max_length);
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return -1;
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}
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return out_length / num_audio_channels;
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}
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} // namespace acm1
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} // namespace webrtc
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