/* * 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 "modules/audio_coding/acm2/acm_resampler.h" #include #include "api/audio/audio_frame.h" #include "rtc_base/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) { InterleavedView src( in_audio, SampleRateToDefaultChannelSize(in_freq_hz), num_audio_channels); InterleavedView dst(out_audio, SampleRateToDefaultChannelSize(out_freq_hz), num_audio_channels); RTC_DCHECK_GE(out_capacity_samples, dst.size()); if (in_freq_hz == out_freq_hz) { if (out_capacity_samples < src.data().size()) { RTC_DCHECK_NOTREACHED(); return -1; } CopySamples(dst, src); RTC_DCHECK_EQ(dst.samples_per_channel(), src.samples_per_channel()); return static_cast(dst.samples_per_channel()); } int out_length = resampler_.Resample(src, dst); if (out_length == -1) { RTC_LOG(LS_ERROR) << "Resample(" << in_audio << ", " << src.data().size() << ", " << out_audio << ", " << out_capacity_samples << ") failed."; return -1; } RTC_DCHECK_EQ(out_length, dst.size()); RTC_DCHECK_EQ(out_length / num_audio_channels, dst.samples_per_channel()); return static_cast(dst.samples_per_channel()); } } // namespace acm2 } // namespace webrtc