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
21namespace webrtc {
22
23class AudioFrame;
24
25namespace 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.
31void 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.
44void 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
54void 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