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