25702cb162
* Better param names * Avoid using negative values for (bogus) placeholder channel counts (mostly in tests). Since channels will be changing to size_t, negative values will be illegal; it's sufficient to use 0 in these cases. * Use arraysize() * Use size_t for counting frames, samples, blocks, buffers, and bytes -- most of these are already size_t in most places, this just fixes some stragglers * reinterpret_cast<int64_t>(void*) is not necessarily safe; use uintptr_t instead * Remove unnecessary code, e.g. dead code, needlessly long/repetitive code, or function overrides that exactly match the base definition * Fix indenting * Use uint32_t for timestamps (matching how it's already a uint32_t in most places) * Spelling * RTC_CHECK_EQ(expected, actual) * Rewrap * Use .empty() * Be more pedantic about matching int/int32_t/ * Remove pointless consts on input parameters to functions * Add missing sanity checks All this was found in the course of constructing https://codereview.webrtc.org/1316523002/ , and is being landed separately first. BUG=none TEST=none Review URL: https://codereview.webrtc.org/1534193008 Cr-Commit-Position: refs/heads/master@{#11191}
64 lines
2.0 KiB
C++
64 lines
2.0 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 "webrtc/modules/audio_coding/acm2/acm_resampler.h"
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#include <assert.h>
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#include <string.h>
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#include "webrtc/common_audio/resampler/include/resampler.h"
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#include "webrtc/system_wrappers/include/logging.h"
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namespace webrtc {
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namespace acm2 {
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ACMResampler::ACMResampler() {
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}
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ACMResampler::~ACMResampler() {
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}
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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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int num_audio_channels,
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size_t out_capacity_samples,
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int16_t* out_audio) {
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size_t in_length = static_cast<size_t>(in_freq_hz * num_audio_channels / 100);
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if (in_freq_hz == out_freq_hz) {
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if (out_capacity_samples < in_length) {
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assert(false);
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return -1;
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}
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memcpy(out_audio, in_audio, in_length * sizeof(int16_t));
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return static_cast<int>(in_length / num_audio_channels);
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}
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if (resampler_.InitializeIfNeeded(in_freq_hz, out_freq_hz,
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num_audio_channels) != 0) {
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LOG(LS_ERROR) << "InitializeIfNeeded(" << in_freq_hz << ", " << out_freq_hz
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<< ", " << num_audio_channels << ") failed.";
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return -1;
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}
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int out_length =
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resampler_.Resample(in_audio, in_length, out_audio, out_capacity_samples);
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if (out_length == -1) {
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LOG(LS_ERROR) << "Resample(" << in_audio << ", " << in_length << ", "
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<< out_audio << ", " << out_capacity_samples << ") failed.";
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return -1;
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}
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return out_length / num_audio_channels;
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}
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} // namespace acm2
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} // namespace webrtc
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