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| 1 /* | 1 /* |
| 2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. |
| 3 * | 3 * |
| 4 * Use of this source code is governed by a BSD-style license | 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 | 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 | 6 * tree. An additional intellectual property rights grant can be found |
| 7 * in the file PATENTS. All contributing project authors may | 7 * in the file PATENTS. All contributing project authors may |
| 8 * be found in the AUTHORS file in the root of the source tree. | 8 * be found in the AUTHORS file in the root of the source tree. |
| 9 */ | 9 */ |
| 10 | 10 |
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| 36 // specified by |codec_num_channels| and |codec_rate_hz|. |mono_buffer| is | 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 | 37 // temporary space and must be of sufficient size to hold the downmixed source |
| 38 // audio (recommend using a size of kMaxMonoDataSizeSamples). | 38 // audio (recommend using a size of kMaxMonoDataSizeSamples). |
| 39 // | 39 // |
| 40 // |dst_af| will have its data and format members (sample rate, channels and | 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. | 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 | 42 // TODO(ajm): For now, this still calls Reset() on |dst_af|. Remove this, as |
| 43 // it shouldn't be needed. | 43 // it shouldn't be needed. |
| 44 void DownConvertToCodecFormat(const int16_t* src_data, | 44 void DownConvertToCodecFormat(const int16_t* src_data, |
| 45 size_t samples_per_channel, | 45 size_t samples_per_channel, |
| 46 int num_channels, | 46 size_t num_channels, |
| 47 int sample_rate_hz, | 47 int sample_rate_hz, |
| 48 int codec_num_channels, | 48 size_t codec_num_channels, |
| 49 int codec_rate_hz, | 49 int codec_rate_hz, |
| 50 int16_t* mono_buffer, | 50 int16_t* mono_buffer, |
| 51 PushResampler<int16_t>* resampler, | 51 PushResampler<int16_t>* resampler, |
| 52 AudioFrame* dst_af); | 52 AudioFrame* dst_af); |
| 53 | 53 |
| 54 void MixWithSat(int16_t target[], | 54 void MixWithSat(int16_t target[], |
| 55 int target_channel, | 55 size_t target_channel, |
| 56 const int16_t source[], | 56 const int16_t source[], |
| 57 int source_channel, | 57 size_t source_channel, |
| 58 size_t source_len); | 58 size_t source_len); |
| 59 | 59 |
| 60 } // namespace voe | 60 } // namespace voe |
| 61 } // namespace webrtc | 61 } // namespace webrtc |
| 62 | 62 |
| 63 #endif // WEBRTC_VOICE_ENGINE_UTILITY_H_ | 63 #endif // WEBRTC_VOICE_ENGINE_UTILITY_H_ |
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