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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}
48 lines
1.7 KiB
C++
48 lines
1.7 KiB
C++
/*
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* Copyright (c) 2014 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_coding/neteq/tools/resample_input_audio_file.h"
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#include <memory>
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#include "rtc_base/checks.h"
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namespace webrtc {
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namespace test {
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bool ResampleInputAudioFile::Read(size_t samples,
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int output_rate_hz,
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int16_t* destination) {
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const size_t samples_to_read = samples * file_rate_hz_ / output_rate_hz;
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RTC_CHECK_EQ(samples_to_read * output_rate_hz, samples * file_rate_hz_)
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<< "Frame size and sample rates don't add up to an integer.";
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std::unique_ptr<int16_t[]> temp_destination(new int16_t[samples_to_read]);
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if (!InputAudioFile::Read(samples_to_read, temp_destination.get()))
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return false;
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resampler_.ResetIfNeeded(file_rate_hz_, output_rate_hz, 1);
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size_t output_length = 0;
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RTC_CHECK_EQ(resampler_.Push(temp_destination.get(), samples_to_read,
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destination, samples, output_length),
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0);
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RTC_CHECK_EQ(samples, output_length);
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return true;
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}
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bool ResampleInputAudioFile::Read(size_t samples, int16_t* destination) {
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RTC_CHECK_GT(output_rate_hz_, 0) << "Output rate not set.";
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return Read(samples, output_rate_hz_, destination);
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}
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void ResampleInputAudioFile::set_output_rate_hz(int rate_hz) {
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output_rate_hz_ = rate_hz;
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}
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} // namespace test
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} // namespace webrtc
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