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a8b7c7f4c6
- Move files from voice_engine/ to audio/. - Rename voice_engine/utility.* to remix_resample.* since there are no other utilities in those files. - Move test/mock_voe_channel_proxy.h to audio/. - Removed voe_channel_id from Audio[Receive|Send]Stream::Config. - Remove VoiceEngine* from AudioState::Config. - Fix a few cpplint complaints which showed when moving files. NOPRESUBMIT=true Bug: webrtc:4690 Change-Id: Id266c822d956625c358fa5e193e6f4837164aef8 Reviewed-on: https://webrtc-review.googlesource.com/39268 Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21657}
46 lines
1.6 KiB
C++
46 lines
1.6 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 AUDIO_REMIX_RESAMPLE_H_
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#define AUDIO_REMIX_RESAMPLE_H_
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#include "common_audio/resampler/include/push_resampler.h"
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namespace webrtc {
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class AudioFrame;
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namespace voe {
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// Upmix or downmix and resample the audio to |dst_frame|. Expects |dst_frame|
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// to have its sample rate and channels members set to the desired values.
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// Updates the |samples_per_channel_| member accordingly.
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//
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// This version has an AudioFrame |src_frame| as input and sets the output
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// |timestamp_|, |elapsed_time_ms_| and |ntp_time_ms_| members equals to the
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// input ones.
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void RemixAndResample(const AudioFrame& src_frame,
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PushResampler<int16_t>* resampler,
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AudioFrame* dst_frame);
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// This version has a pointer to the samples |src_data| as input and receives
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// |samples_per_channel|, |num_channels| and |sample_rate_hz| of the data as
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// parameters.
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void RemixAndResample(const int16_t* src_data,
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size_t samples_per_channel,
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size_t num_channels,
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int sample_rate_hz,
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PushResampler<int16_t>* 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 // AUDIO_REMIX_RESAMPLE_H_
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