mirror of
https://github.com/mollyim/webrtc.git
synced 2025-05-12 21:30:45 +01:00
1f3679884c
Start introducing ArrayView to AudioFrame and code that flows down from there. In this first step: * Add `data_view()` that returns a read-only ArrayView for the audio buffer. When AudioFrame is not initialized however, data_view() will return a nullptr whereas the current data() method never returns nullptr. * Add `mutable_data()` that requires two arguments for properly setting the samples per channel and number of channels that's required for accurately reserving the returned mutable ArrayView. A notable behavior change is that if the requested number of channels is larger than supported or the calculated buffer size is too large, the function will trigger a check. * Add TODOs for following work. Bug: chromium:335805780 Change-Id: I2937de800422589ebe6a3840b3caadf3d9ff8b00 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/347982 Reviewed-by: Per Åhgren <peah@webrtc.org> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42202}
62 lines
2 KiB
C++
62 lines
2 KiB
C++
/*
|
|
* Copyright (c) 2012 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_coding/acm2/acm_resampler.h"
|
|
|
|
#include <string.h>
|
|
|
|
#include "rtc_base/logging.h"
|
|
|
|
namespace webrtc {
|
|
namespace acm2 {
|
|
|
|
ACMResampler::ACMResampler() {}
|
|
|
|
ACMResampler::~ACMResampler() {}
|
|
|
|
int ACMResampler::Resample10Msec(const int16_t* in_audio,
|
|
int in_freq_hz,
|
|
int out_freq_hz,
|
|
size_t num_audio_channels,
|
|
size_t out_capacity_samples,
|
|
int16_t* out_audio) {
|
|
size_t in_length = in_freq_hz * num_audio_channels / 100;
|
|
if (in_freq_hz == out_freq_hz) {
|
|
if (out_capacity_samples < in_length) {
|
|
RTC_DCHECK_NOTREACHED();
|
|
return -1;
|
|
}
|
|
memcpy(out_audio, in_audio, in_length * sizeof(int16_t));
|
|
return static_cast<int>(in_length / num_audio_channels);
|
|
}
|
|
|
|
if (resampler_.InitializeIfNeeded(in_freq_hz, out_freq_hz,
|
|
num_audio_channels) != 0) {
|
|
RTC_LOG(LS_ERROR) << "InitializeIfNeeded(" << in_freq_hz << ", "
|
|
<< out_freq_hz << ", " << num_audio_channels
|
|
<< ") failed.";
|
|
return -1;
|
|
}
|
|
|
|
int out_length = resampler_.Resample(
|
|
rtc::ArrayView<const int16_t>(in_audio, in_length),
|
|
rtc::ArrayView<int16_t>(out_audio, out_capacity_samples));
|
|
if (out_length == -1) {
|
|
RTC_LOG(LS_ERROR) << "Resample(" << in_audio << ", " << in_length << ", "
|
|
<< out_audio << ", " << out_capacity_samples
|
|
<< ") failed.";
|
|
return -1;
|
|
}
|
|
|
|
return static_cast<int>(out_length / num_audio_channels);
|
|
}
|
|
|
|
} // namespace acm2
|
|
} // namespace webrtc
|