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92ea95e34a
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC from src/webrtc to src/ (in order to preserve an healthy git history). This CL takes care of fixing header guards, #include paths, etc... NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578 Reviewed-on: https://webrtc-review.googlesource.com/1561 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19846}
39 lines
1.5 KiB
C++
39 lines
1.5 KiB
C++
/*
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* Copyright (c) 2016 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 "modules/audio_mixer/default_output_rate_calculator.h"
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#include <algorithm>
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#include "modules/audio_processing/include/audio_processing.h"
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namespace webrtc {
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int DefaultOutputRateCalculator::CalculateOutputRate(
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const std::vector<int>& preferred_sample_rates) {
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if (preferred_sample_rates.empty()) {
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return DefaultOutputRateCalculator::kDefaultFrequency;
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}
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using NativeRate = AudioProcessing::NativeRate;
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const int maximal_frequency = *std::max_element(
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preferred_sample_rates.begin(), preferred_sample_rates.end());
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RTC_DCHECK_LE(NativeRate::kSampleRate8kHz, maximal_frequency);
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RTC_DCHECK_GE(NativeRate::kSampleRate48kHz, maximal_frequency);
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static constexpr NativeRate native_rates[] = {
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NativeRate::kSampleRate8kHz, NativeRate::kSampleRate16kHz,
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NativeRate::kSampleRate32kHz, NativeRate::kSampleRate48kHz};
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const auto* rounded_up_index = std::lower_bound(
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std::begin(native_rates), std::end(native_rates), maximal_frequency);
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RTC_DCHECK(rounded_up_index != std::end(native_rates));
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return *rounded_up_index;
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}
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} // namespace webrtc
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