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Diffstat (limited to 'webrtc/modules/audio_processing/include/audio_frame_proxies.cc')
-rw-r--r--webrtc/modules/audio_processing/include/audio_frame_proxies.cc70
1 files changed, 70 insertions, 0 deletions
diff --git a/webrtc/modules/audio_processing/include/audio_frame_proxies.cc b/webrtc/modules/audio_processing/include/audio_frame_proxies.cc
new file mode 100644
index 0000000..b960e72
--- /dev/null
+++ b/webrtc/modules/audio_processing/include/audio_frame_proxies.cc
@@ -0,0 +1,70 @@
+/*
+ * 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