/* * 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/acm2/acm_resampler.h" #include #include #include "webrtc/common_audio/resampler/include/resampler.h" #include "webrtc/system_wrappers/include/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(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(out_length / num_audio_channels); } } // namespace acm2 } // namespace webrtc