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1 /*
2  *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
3  *
4  *  Use of this source code is governed by a BSD-style license
5  *  that can be found in the LICENSE file in the root of the source
6  *  tree. An additional intellectual property rights grant can be found
7  *  in the file PATENTS.  All contributing project authors may
8  *  be found in the AUTHORS file in the root of the source tree.
9  */
10 
11 /*
12  *  Contains functions often used by different parts of VoiceEngine.
13  */
14 
15 #ifndef WEBRTC_VOICE_ENGINE_UTILITY_H_
16 #define WEBRTC_VOICE_ENGINE_UTILITY_H_
17 
18 #include "webrtc/common_audio/resampler/include/push_resampler.h"
19 #include "webrtc/typedefs.h"
20 
21 namespace webrtc {
22 
23 class AudioFrame;
24 
25 namespace voe {
26 
27 // Upmix or downmix and resample the audio to |dst_frame|. Expects |dst_frame|
28 // to have its sample rate and channels members set to the desired values.
29 // Updates the |samples_per_channel_| member accordingly.
30 //
31 // This version has an AudioFrame |src_frame| as input and sets the output
32 // |timestamp_|, |elapsed_time_ms_| and |ntp_time_ms_| members equals to the
33 // input ones.
34 void RemixAndResample(const AudioFrame& src_frame,
35                       PushResampler<int16_t>* resampler,
36                       AudioFrame* dst_frame);
37 
38 // This version has a pointer to the samples |src_data| as input and receives
39 // |samples_per_channel|, |num_channels| and |sample_rate_hz| of the data as
40 // parameters.
41 void RemixAndResample(const int16_t* src_data,
42                       size_t samples_per_channel,
43                       size_t num_channels,
44                       int sample_rate_hz,
45                       PushResampler<int16_t>* resampler,
46                       AudioFrame* dst_frame);
47 
48 void MixWithSat(int16_t target[],
49                 size_t target_channel,
50                 const int16_t source[],
51                 size_t source_channel,
52                 size_t source_len);
53 
54 }  // namespace voe
55 }  // namespace webrtc
56 
57 #endif  // WEBRTC_VOICE_ENGINE_UTILITY_H_
58