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Unified Diff: webrtc/modules/audio_device/include/audio_transport_capture.h

Issue 2753453002: Adding AudioDeviceDataObserver interface (Closed)
Patch Set: Fix gn typo Created 3 years, 9 months ago
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Index: webrtc/modules/audio_device/include/audio_transport_capture.h
diff --git a/webrtc/modules/audio_device/include/audio_transport_capture.h b/webrtc/modules/audio_device/include/audio_transport_capture.h
new file mode 100644
index 0000000000000000000000000000000000000000..a848ecb0c7d246a84118d01bbfb938afcbc3aee2
--- /dev/null
+++ b/webrtc/modules/audio_device/include/audio_transport_capture.h
@@ -0,0 +1,46 @@
+/*
+ * Copyright (c) 2017 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.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_DEVICE_INCLUDE_AUDIO_TRANSPORT_CAPTURE_H_
+#define WEBRTC_MODULES_AUDIO_DEVICE_INCLUDE_AUDIO_TRANSPORT_CAPTURE_H_
+
+#include "webrtc/base/refcountedobject.h"
+#include "webrtc/modules/audio_device/include/audio_device.h"
+#include "webrtc/modules/include/module.h"
+
+namespace webrtc {
+
+class AudioTransportCapture {
+ public:
+ virtual void CaptureRecordedData(
+ const void* audioSamples, const size_t nSamples,
henrika_webrtc 2017/03/23 09:02:42 Can't we use new notation in this interface? audio
lliuu 2017/03/24 00:12:58 Acknowledged.
+ const size_t nBytesPerSample, const size_t nChannels,
+ const uint32_t samplesPerSec) = 0;
+
+ virtual void CapturePlayoutData(
+ const void* audioSamples, const size_t nSamples,
+ const size_t nBytesPerSample, const size_t nChannels,
+ const uint32_t samplesPerSec) = 0;
+
+ AudioTransportCapture() = default;
+ virtual ~AudioTransportCapture() = default;
+};
+
+// Creates an ADM instance with AudioTransportCapture callback.
+rtc::scoped_refptr<AudioDeviceModule>
+ CreateAudioDeviceWithTransportCapture(
+ const int32_t id,
+ const AudioDeviceModule::AudioLayer audio_layer,
+ AudioTransportCapture* transport_cb);
+
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_AUDIO_DEVICE_INCLUDE_AUDIO_TRANSPORT_CAPTURE_H_
+

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