/* * Copyright (c) 2016 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_mixer/default_output_rate_calculator.h" #include #include #include "modules/audio_processing/include/audio_processing.h" #include "rtc_base/checks.h" namespace webrtc { int DefaultOutputRateCalculator::CalculateOutputRateFromRange( rtc::ArrayView preferred_sample_rates) { if (preferred_sample_rates.empty()) { return DefaultOutputRateCalculator::kDefaultFrequency; } using NativeRate = AudioProcessing::NativeRate; const int maximal_frequency = *std::max_element( preferred_sample_rates.cbegin(), preferred_sample_rates.cend()); RTC_DCHECK_LE(NativeRate::kSampleRate8kHz, maximal_frequency); RTC_DCHECK_GE(NativeRate::kSampleRate48kHz, maximal_frequency); static constexpr NativeRate native_rates[] = { NativeRate::kSampleRate8kHz, NativeRate::kSampleRate16kHz, NativeRate::kSampleRate32kHz, NativeRate::kSampleRate48kHz}; const auto* rounded_up_index = std::lower_bound( std::begin(native_rates), std::end(native_rates), maximal_frequency); RTC_DCHECK(rounded_up_index != std::end(native_rates)); return *rounded_up_index; } } // namespace webrtc