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 #include "webrtc/modules/include/module_common_types.h" 12 #include "webrtc/modules/utility/include/audio_frame_operations.h" 13 14 namespace webrtc { 15 16 void AudioFrameOperations::MonoToStereo(const int16_t* src_audio, 17 size_t samples_per_channel, 18 int16_t* dst_audio) { 19 for (size_t i = 0; i < samples_per_channel; i++) { 20 dst_audio[2 * i] = src_audio[i]; 21 dst_audio[2 * i + 1] = src_audio[i]; 22 } 23 } 24 25 int AudioFrameOperations::MonoToStereo(AudioFrame* frame) { 26 if (frame->num_channels_ != 1) { 27 return -1; 28 } 29 if ((frame->samples_per_channel_ * 2) >= AudioFrame::kMaxDataSizeSamples) { 30 // Not enough memory to expand from mono to stereo. 31 return -1; 32 } 33 34 int16_t data_copy[AudioFrame::kMaxDataSizeSamples]; 35 memcpy(data_copy, frame->data_, 36 sizeof(int16_t) * frame->samples_per_channel_); 37 MonoToStereo(data_copy, frame->samples_per_channel_, frame->data_); 38 frame->num_channels_ = 2; 39 40 return 0; 41 } 42 43 void AudioFrameOperations::StereoToMono(const int16_t* src_audio, 44 size_t samples_per_channel, 45 int16_t* dst_audio) { 46 for (size_t i = 0; i < samples_per_channel; i++) { 47 dst_audio[i] = (src_audio[2 * i] + src_audio[2 * i + 1]) >> 1; 48 } 49 } 50 51 int AudioFrameOperations::StereoToMono(AudioFrame* frame) { 52 if (frame->num_channels_ != 2) { 53 return -1; 54 } 55 56 StereoToMono(frame->data_, frame->samples_per_channel_, frame->data_); 57 frame->num_channels_ = 1; 58 59 return 0; 60 } 61 62 void AudioFrameOperations::SwapStereoChannels(AudioFrame* frame) { 63 if (frame->num_channels_ != 2) return; 64 65 for (size_t i = 0; i < frame->samples_per_channel_ * 2; i += 2) { 66 int16_t temp_data = frame->data_[i]; 67 frame->data_[i] = frame->data_[i + 1]; 68 frame->data_[i + 1] = temp_data; 69 } 70 } 71 72 void AudioFrameOperations::Mute(AudioFrame& frame) { 73 memset(frame.data_, 0, sizeof(int16_t) * 74 frame.samples_per_channel_ * frame.num_channels_); 75 } 76 77 int AudioFrameOperations::Scale(float left, float right, AudioFrame& frame) { 78 if (frame.num_channels_ != 2) { 79 return -1; 80 } 81 82 for (size_t i = 0; i < frame.samples_per_channel_; i++) { 83 frame.data_[2 * i] = 84 static_cast<int16_t>(left * frame.data_[2 * i]); 85 frame.data_[2 * i + 1] = 86 static_cast<int16_t>(right * frame.data_[2 * i + 1]); 87 } 88 return 0; 89 } 90 91 int AudioFrameOperations::ScaleWithSat(float scale, AudioFrame& frame) { 92 int32_t temp_data = 0; 93 94 // Ensure that the output result is saturated [-32768, +32767]. 95 for (size_t i = 0; i < frame.samples_per_channel_ * frame.num_channels_; 96 i++) { 97 temp_data = static_cast<int32_t>(scale * frame.data_[i]); 98 if (temp_data < -32768) { 99 frame.data_[i] = -32768; 100 } else if (temp_data > 32767) { 101 frame.data_[i] = 32767; 102 } else { 103 frame.data_[i] = static_cast<int16_t>(temp_data); 104 } 105 } 106 return 0; 107 } 108 109 } // namespace webrtc 110
