| /* | 
 |  *  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, | 
 |                       size_t num_channels, | 
 |                       int sample_rate_hz, | 
 |                       PushResampler<int16_t>* resampler, | 
 |                       AudioFrame* dst_frame); | 
 |  | 
 | void MixWithSat(int16_t target[], | 
 |                 size_t target_channel, | 
 |                 const int16_t source[], | 
 |                 size_t source_channel, | 
 |                 size_t source_len); | 
 |  | 
 | }  // namespace voe | 
 | }  // namespace webrtc | 
 |  | 
 | #endif  // WEBRTC_VOICE_ENGINE_UTILITY_H_ |