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The 6-parameter Initialize method is removed. The has_keyboard parameter in the StreamConfig constructor is removed together with the underlying member and helper functions. Bug: chromium:1271981, b/217349489 Change-Id: I7259a114a395f74f735a9c06510c0fc0f0f008e3 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250221 Reviewed-by: Sam Zackrisson <saza@google.com> Reviewed-by: Sam Zackrisson <saza@webrtc.org> Auto-Submit: Henrik Lundin <henrik.lundin@webrtc.org> Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35908}
66 lines
2.4 KiB
C++
66 lines
2.4 KiB
C++
/*
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* Copyright (c) 2020 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_processing/include/audio_frame_proxies.h"
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#include "api/audio/audio_frame.h"
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#include "modules/audio_processing/include/audio_processing.h"
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namespace webrtc {
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int ProcessAudioFrame(AudioProcessing* ap, AudioFrame* frame) {
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if (!frame || !ap) {
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return AudioProcessing::Error::kNullPointerError;
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}
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StreamConfig input_config(frame->sample_rate_hz_, frame->num_channels_);
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StreamConfig output_config(frame->sample_rate_hz_, frame->num_channels_);
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RTC_DCHECK_EQ(frame->samples_per_channel(), input_config.num_frames());
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int result = ap->ProcessStream(frame->data(), input_config, output_config,
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frame->mutable_data());
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AudioProcessingStats stats = ap->GetStatistics();
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if (stats.voice_detected) {
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frame->vad_activity_ = *stats.voice_detected
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? AudioFrame::VADActivity::kVadActive
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: AudioFrame::VADActivity::kVadPassive;
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}
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return result;
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}
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int ProcessReverseAudioFrame(AudioProcessing* ap, AudioFrame* frame) {
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if (!frame || !ap) {
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return AudioProcessing::Error::kNullPointerError;
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}
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// Must be a native rate.
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if (frame->sample_rate_hz_ != AudioProcessing::NativeRate::kSampleRate8kHz &&
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frame->sample_rate_hz_ != AudioProcessing::NativeRate::kSampleRate16kHz &&
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frame->sample_rate_hz_ != AudioProcessing::NativeRate::kSampleRate32kHz &&
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frame->sample_rate_hz_ != AudioProcessing::NativeRate::kSampleRate48kHz) {
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return AudioProcessing::Error::kBadSampleRateError;
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}
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if (frame->num_channels_ <= 0) {
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return AudioProcessing::Error::kBadNumberChannelsError;
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}
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StreamConfig input_config(frame->sample_rate_hz_, frame->num_channels_);
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StreamConfig output_config(frame->sample_rate_hz_, frame->num_channels_);
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int result = ap->ProcessReverseStream(frame->data(), input_config,
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output_config, frame->mutable_data());
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return result;
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}
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} // namespace webrtc
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