AnotherFoxGuy/webrtc-audio-processing
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Update to current webrtc library

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This is from the upstream library commit id
3326535126e435f1ba647885ce43a8f0f3d317eb, corresponding to Chromium
88.0.4290.1.
This commit is contained in:
Arun Raghavan
2020-10-12 18:08:02 -04:00
parent b1b02581d3
commit bcec8b0b21
859 changed files with 76187 additions and 49580 deletions
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70
webrtc/modules/audio_processing/include/audio_frame_proxies.cc Normal file
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/*
* Copyright (c) 2020 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_processing/include/audio_frame_proxies.h"
#include "api/audio/audio_frame.h"
#include "modules/audio_processing/include/audio_processing.h"
namespace webrtc {
int ProcessAudioFrame(AudioProcessing* ap, AudioFrame* frame) {
if (!frame || !ap) {
return AudioProcessing::Error::kNullPointerError;
}
StreamConfig input_config(frame->sample_rate_hz_, frame->num_channels_,
/*has_keyboard=*/false);
StreamConfig output_config(frame->sample_rate_hz_, frame->num_channels_,
/*has_keyboard=*/false);
RTC_DCHECK_EQ(frame->samples_per_channel(), input_config.num_frames());
int result = ap->ProcessStream(frame->data(), input_config, output_config,
frame->mutable_data());
AudioProcessingStats stats = ap->GetStatistics();
if (stats.voice_detected) {
frame->vad_activity_ = *stats.voice_detected
? AudioFrame::VADActivity::kVadActive
: AudioFrame::VADActivity::kVadPassive;
}
return result;
}
int ProcessReverseAudioFrame(AudioProcessing* ap, AudioFrame* frame) {
if (!frame || !ap) {
return AudioProcessing::Error::kNullPointerError;
}
// Must be a native rate.
if (frame->sample_rate_hz_ != AudioProcessing::NativeRate::kSampleRate8kHz &&
frame->sample_rate_hz_ != AudioProcessing::NativeRate::kSampleRate16kHz &&
frame->sample_rate_hz_ != AudioProcessing::NativeRate::kSampleRate32kHz &&
frame->sample_rate_hz_ != AudioProcessing::NativeRate::kSampleRate48kHz) {
return AudioProcessing::Error::kBadSampleRateError;
}
if (frame->num_channels_ <= 0) {
return AudioProcessing::Error::kBadNumberChannelsError;
}
StreamConfig input_config(frame->sample_rate_hz_, frame->num_channels_,
/*has_keyboard=*/false);
StreamConfig output_config(frame->sample_rate_hz_, frame->num_channels_,
/*has_keyboard=*/false);
int result = ap->ProcessReverseStream(frame->data(), input_config,
output_config, frame->mutable_data());
return result;
}
} // namespace webrtc