f5a33f145b
- Templatize PushResampler to support int16 and float. - Add a helper method to PushSincResampler to compute the algorithmic delay. This is a prerequisite of: http://review.webrtc.org/9919004/ BUG=2894 R=turaj@webrtc.org Review URL: https://webrtc-codereview.appspot.com/12169004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5943 4adac7df-926f-26a2-2b94-8c16560cd09d
60 lines
2.2 KiB
C++
60 lines
2.2 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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/*
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* Contains functions often used by different parts of VoiceEngine.
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*/
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#ifndef WEBRTC_VOICE_ENGINE_UTILITY_H_
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#define WEBRTC_VOICE_ENGINE_UTILITY_H_
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#include "webrtc/common_audio/resampler/include/push_resampler.h"
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#include "webrtc/typedefs.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 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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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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// Downmix and downsample the audio in |src_data| to |dst_af| as necessary,
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// specified by |codec_num_channels| and |codec_rate_hz|. |mono_buffer| is
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// temporary space and must be of sufficient size to hold the downmixed source
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// audio (recommend using a size of kMaxMonoDataSizeSamples).
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void DownConvertToCodecFormat(const int16_t* src_data,
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int samples_per_channel,
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int num_channels,
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int sample_rate_hz,
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int codec_num_channels,
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int codec_rate_hz,
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int16_t* mono_buffer,
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PushResampler<int16_t>* resampler,
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AudioFrame* dst_af);
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void MixWithSat(int16_t target[],
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int target_channel,
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const int16_t source[],
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int source_channel,
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int source_len);
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} // namespace voe
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} // namespace webrtc
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#endif // WEBRTC_VOICE_ENGINE_UTILITY_H_
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