|  | /* | 
|  | *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. | 
|  | * | 
|  | *  Use of this source code is governed by a BSD-style license | 
|  | *  that can be found in the LICENSE file in the root of the source | 
|  | *  tree. An additional intellectual property rights grant can be found | 
|  | *  in the file PATENTS.  All contributing project authors may | 
|  | *  be found in the AUTHORS file in the root of the source tree. | 
|  | */ | 
|  |  | 
|  | /* | 
|  | *  Contains functions often used by different parts of VoiceEngine. | 
|  | */ | 
|  |  | 
|  | #ifndef WEBRTC_VOICE_ENGINE_UTILITY_H_ | 
|  | #define WEBRTC_VOICE_ENGINE_UTILITY_H_ | 
|  |  | 
|  | #include "webrtc/common_audio/resampler/include/push_resampler.h" | 
|  | #include "webrtc/typedefs.h" | 
|  |  | 
|  | namespace webrtc { | 
|  |  | 
|  | class AudioFrame; | 
|  |  | 
|  | namespace voe { | 
|  |  | 
|  | // Upmix or downmix and resample the audio to |dst_frame|. Expects |dst_frame| | 
|  | // to have its sample rate and channels members set to the desired values. | 
|  | // Updates the |samples_per_channel_| member accordingly. | 
|  | // | 
|  | // This version has an AudioFrame |src_frame| as input and sets the output | 
|  | // |timestamp_|, |elapsed_time_ms_| and |ntp_time_ms_| members equals to the | 
|  | // input ones. | 
|  | void RemixAndResample(const AudioFrame& src_frame, | 
|  | PushResampler<int16_t>* resampler, | 
|  | AudioFrame* dst_frame); | 
|  |  | 
|  | // This version has a pointer to the samples |src_data| as input and receives | 
|  | // |samples_per_channel|, |num_channels| and |sample_rate_hz| of the data as | 
|  | // parameters. | 
|  | void RemixAndResample(const int16_t* src_data, | 
|  | size_t samples_per_channel, | 
|  | int num_channels, | 
|  | int sample_rate_hz, | 
|  | PushResampler<int16_t>* resampler, | 
|  | AudioFrame* dst_frame); | 
|  |  | 
|  | void MixWithSat(int16_t target[], | 
|  | int target_channel, | 
|  | const int16_t source[], | 
|  | int source_channel, | 
|  | size_t source_len); | 
|  |  | 
|  | }  // namespace voe | 
|  | }  // namespace webrtc | 
|  |  | 
|  | #endif  // WEBRTC_VOICE_ENGINE_UTILITY_H_ |