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d6ef33e59b
This switches from accepting a sample rate and convert to channel size over to accepting the channel size. Instead of InitializeIfNeeded: * Offer a way to explicitly initialize PushResampler via the ctor (needed for VoiceActivityDetectorWrapper) * Implicitly check for the right configuration from within Resample(). (All calls to Resample() were preceded by a call to Initialize) As part of this, refactor VoiceActivityDetectorWrapper (VADW): * VADW is now initialized in the constructor and more const. * Remove VADW::Initialize() and instead reconstruct VADW if needed. Add constants for max sample rate and num channels to audio_util.h In many cases the numbers for these values are embedded in the code which has led to some inconsistency. Bug: chromium:335805780 Change-Id: Iead0d52eb1b261a8d64e93f51401147c8fba32f0 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/353360 Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42587}
60 lines
2.1 KiB
C++
60 lines
2.1 KiB
C++
/*
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* Copyright (c) 2012 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/acm2/acm_resampler.h"
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#include <string.h>
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#include "api/audio/audio_frame.h"
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#include "rtc_base/logging.h"
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namespace webrtc {
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namespace acm2 {
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ACMResampler::ACMResampler() {}
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ACMResampler::~ACMResampler() {}
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int ACMResampler::Resample10Msec(const int16_t* in_audio,
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int in_freq_hz,
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int out_freq_hz,
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size_t num_audio_channels,
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size_t out_capacity_samples,
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int16_t* out_audio) {
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InterleavedView<const int16_t> src(
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in_audio, SampleRateToDefaultChannelSize(in_freq_hz), num_audio_channels);
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InterleavedView<int16_t> dst(out_audio,
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SampleRateToDefaultChannelSize(out_freq_hz),
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num_audio_channels);
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RTC_DCHECK_GE(out_capacity_samples, dst.size());
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if (in_freq_hz == out_freq_hz) {
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if (out_capacity_samples < src.data().size()) {
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RTC_DCHECK_NOTREACHED();
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return -1;
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}
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CopySamples(dst, src);
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RTC_DCHECK_EQ(dst.samples_per_channel(), src.samples_per_channel());
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return static_cast<int>(dst.samples_per_channel());
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}
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int out_length = resampler_.Resample(src, dst);
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if (out_length == -1) {
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RTC_LOG(LS_ERROR) << "Resample(" << in_audio << ", " << src.data().size()
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<< ", " << out_audio << ", " << out_capacity_samples
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<< ") failed.";
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return -1;
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}
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RTC_DCHECK_EQ(out_length, dst.size());
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RTC_DCHECK_EQ(out_length / num_audio_channels, dst.samples_per_channel());
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return static_cast<int>(dst.samples_per_channel());
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}
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} // namespace acm2
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} // namespace webrtc
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