diff options
Diffstat (limited to 'third_party/libwebrtc/modules/audio_processing/include/audio_frame_proxies.cc')
-rw-r--r-- | third_party/libwebrtc/modules/audio_processing/include/audio_frame_proxies.cc | 66 |
1 files changed, 66 insertions, 0 deletions
diff --git a/third_party/libwebrtc/modules/audio_processing/include/audio_frame_proxies.cc b/third_party/libwebrtc/modules/audio_processing/include/audio_frame_proxies.cc new file mode 100644 index 0000000000..7cc4fb75e4 --- /dev/null +++ b/third_party/libwebrtc/modules/audio_processing/include/audio_frame_proxies.cc @@ -0,0 +1,66 @@ +/* + * 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_); + StreamConfig output_config(frame->sample_rate_hz_, frame->num_channels_); + 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_); + StreamConfig output_config(frame->sample_rate_hz_, frame->num_channels_); + + int result = ap->ProcessReverseStream(frame->data(), input_config, + output_config, frame->mutable_data()); + return result; +} + +} // namespace webrtc |