50b2efef6e
The old resampler is used whenever it supports the requested rates. Otherwise the sinc resampler is enabled. Integrated with output_mixer in order to test the change through output_mixer_unittest. The sinc resampler will not yet be used, since we don't feed VoE with any rates that trigger it. BUG=webrtc:1395 R=bjornv@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1355004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3915 4adac7df-926f-26a2-2b94-8c16560cd09d
34 lines
1.1 KiB
C++
34 lines
1.1 KiB
C++
/*
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* Copyright (c) 2012 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_VOICE_ENGINE_OUTPUT_MIXER_INTERNAL_H_
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#define WEBRTC_VOICE_ENGINE_OUTPUT_MIXER_INTERNAL_H_
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namespace webrtc {
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class AudioFrame;
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class PushResampler;
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namespace voe {
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// Upmix or downmix and resample the audio in |src_frame| to |dst_frame|.
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// Expects |dst_frame| to have its |num_channels_| and |sample_rate_hz_| set to
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// the desired values. Updates |samples_per_channel_| accordingly.
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//
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// On failure, returns -1 and copies |src_frame| to |dst_frame|.
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int RemixAndResample(const AudioFrame& src_frame,
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PushResampler* resampler,
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AudioFrame* dst_frame);
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} // namespace voe
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} // namespace webrtc
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#endif // VOICE_ENGINE_OUTPUT_MIXER_INTERNAL_H_
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