| /* |
| * 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/typedefs.h" |
| |
| namespace webrtc { |
| |
| class AudioFrame; |
| class PushResampler; |
| |
| namespace voe { |
| |
| // Upmix or downmix and resample the audio in |src_frame| to |dst_frame|. |
| // Expects |dst_frame| to have its |num_channels_| and |sample_rate_hz_| set to |
| // the desired values. Updates |samples_per_channel_| accordingly. |
| // |
| // On failure, returns -1 and copies |src_frame| to |dst_frame|. |
| void RemixAndResample(const AudioFrame& src_frame, |
| PushResampler* resampler, |
| AudioFrame* dst_frame); |
| |
| // Downmix and downsample the audio in |src_data| to |dst_af| as necessary, |
| // specified by |codec_num_channels| and |codec_rate_hz|. |mono_buffer| is |
| // temporary space and must be of sufficient size to hold the downmixed source |
| // audio (recommend using a size of kMaxMonoDataSizeSamples). |
| void DownConvertToCodecFormat(const int16_t* src_data, |
| int samples_per_channel, |
| int num_channels, |
| int sample_rate_hz, |
| int codec_num_channels, |
| int codec_rate_hz, |
| int16_t* mono_buffer, |
| PushResampler* resampler, |
| AudioFrame* dst_af); |
| |
| void MixWithSat(int16_t target[], |
| int target_channel, |
| const int16_t source[], |
| int source_channel, |
| int source_len); |
| |
| } // namespace voe |
| } // namespace webrtc |
| |
| #endif // WEBRTC_VOICE_ENGINE_UTILITY_H_ |