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