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 in |src_frame| to |dst_frame|.
28 // Expects |dst_frame| to have its sample rate and channels members set to the
29 // desired values. Updates the samples per channel member accordingly. No other
30 // members will be changed.
31 void RemixAndResample(const AudioFrame& src_frame,
32                       PushResampler<int16_t>* resampler,
33                       AudioFrame* dst_frame);
34 
35 // Downmix and downsample the audio in |src_data| to |dst_af| as necessary,
36 // specified by |codec_num_channels| and |codec_rate_hz|. |mono_buffer| is
37 // temporary space and must be of sufficient size to hold the downmixed source
38 // audio (recommend using a size of kMaxMonoDataSizeSamples).
39 //
40 // |dst_af| will have its data and format members (sample rate, channels and
41 // samples per channel) set appropriately. No other members will be changed.
42 // TODO(ajm): For now, this still calls Reset() on |dst_af|. Remove this, as
43 // it shouldn't be needed.
44 void DownConvertToCodecFormat(const int16_t* src_data,
45                               int samples_per_channel,
46                               int num_channels,
47                               int sample_rate_hz,
48                               int codec_num_channels,
49                               int codec_rate_hz,
50                               int16_t* mono_buffer,
51                               PushResampler<int16_t>* resampler,
52                               AudioFrame* dst_af);
53 
54 void MixWithSat(int16_t target[],
55                 int target_channel,
56                 const int16_t source[],
57                 int source_channel,
58                 int source_len);
59 
60 }  // namespace voe
61 }  // namespace webrtc
62 
63 #endif  // WEBRTC_VOICE_ENGINE_UTILITY_H_
64