c1eb560a5c
* The old resampler was found to have a wraparound bug. * Remove support for the old resampler from PushResampler. * Use PushResampler in AudioCodingModule. * The old resampler must still be removed from the file utility. BUG=webrtc:1867,webrtc:827 TESTED=unit tests, Chrome using apprtc and voe_cmd_test to verify wrap-around is corrected, voe_cmd_test running through all supported codec sample rates and channels to verify good quality audio R=henrika@webrtc.org, turaj@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1590004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4156 4adac7df-926f-26a2-2b94-8c16560cd09d
60 lines
2.0 KiB
C++
60 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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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 webrtc
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