blob: 57d88b638485f40d3b9c6d9bfd3569a6c0e4d387 [file] [log] [blame]
* 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 <algorithm>
#include <iterator>
#include "modules/audio_processing/include/audio_processing.h"
#include "rtc_base/checks.h"
namespace webrtc {
int DefaultOutputRateCalculator::CalculateOutputRate(
const std::vector<int>& 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.begin(), preferred_sample_rates.end());
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