The AudioMixer is now split in a mixer and audio source interface part, which has moved to webrtc/api, and a default implementation part, which lies in webrtc/modules. This change makes it possible to create other mixer implementations and is a first step to facilitate passing down a mixer from outside of WebRTC. It will also create less build dependencies when the new mixer has replaced the old one. NOTRY=True TBR=henrik.lundin@webrtc.org BUG=webrtc:6346 Review-Url: https://codereview.webrtc.org/2411313003 Cr-Commit-Position: refs/heads/master@{#14705}
31 lines
1.1 KiB
C++
31 lines
1.1 KiB
C++
/*
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* Copyright (c) 2011 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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#ifndef WEBRTC_MODULES_AUDIO_MIXER_AUDIO_FRAME_MANIPULATOR_H_
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#define WEBRTC_MODULES_AUDIO_MIXER_AUDIO_FRAME_MANIPULATOR_H_
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#include "webrtc/modules/include/module_common_types.h"
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namespace webrtc {
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// Updates the audioFrame's energy (based on its samples).
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uint32_t AudioMixerCalculateEnergy(const AudioFrame& audio_frame);
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// Ramps up or down the provided audio frame. Ramp(0, 1, frame) will
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// linearly increase the samples in the frame from 0 to full volume.
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void Ramp(float start_gain, float target_gain, AudioFrame* audio_frame);
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// Downmixes or upmixes a frame between stereo and mono.
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void RemixFrame(size_t target_number_of_channels, AudioFrame* frame);
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} // namespace webrtc
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#endif // WEBRTC_MODULES_AUDIO_MIXER_AUDIO_FRAME_MANIPULATOR_H_
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