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/voice_engine/utility.h" 12 13#include "webrtc/common_audio/resampler/include/push_resampler.h" 14#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h" 15#include "webrtc/common_types.h" 16#include "webrtc/modules/interface/module_common_types.h" 17#include "webrtc/modules/utility/interface/audio_frame_operations.h" 18#include "webrtc/system_wrappers/interface/logging.h" 19#include "webrtc/voice_engine/voice_engine_defines.h" 20 21namespace webrtc { 22namespace voe { 23 24// TODO(ajm): There is significant overlap between RemixAndResample and 25// ConvertToCodecFormat, but if we're to consolidate we should probably make a 26// real converter class. 27void RemixAndResample(const AudioFrame& src_frame, 28 PushResampler<int16_t>* resampler, 29 AudioFrame* dst_frame) { 30 const int16_t* audio_ptr = src_frame.data_; 31 int audio_ptr_num_channels = src_frame.num_channels_; 32 int16_t mono_audio[AudioFrame::kMaxDataSizeSamples]; 33 34 // Downmix before resampling. 35 if (src_frame.num_channels_ == 2 && dst_frame->num_channels_ == 1) { 36 AudioFrameOperations::StereoToMono(src_frame.data_, 37 src_frame.samples_per_channel_, 38 mono_audio); 39 audio_ptr = mono_audio; 40 audio_ptr_num_channels = 1; 41 } 42 43 if (resampler->InitializeIfNeeded(src_frame.sample_rate_hz_, 44 dst_frame->sample_rate_hz_, 45 audio_ptr_num_channels) == -1) { 46 LOG_FERR3(LS_ERROR, InitializeIfNeeded, src_frame.sample_rate_hz_, 47 dst_frame->sample_rate_hz_, audio_ptr_num_channels); 48 assert(false); 49 } 50 51 const int src_length = src_frame.samples_per_channel_ * 52 audio_ptr_num_channels; 53 int out_length = resampler->Resample(audio_ptr, src_length, dst_frame->data_, 54 AudioFrame::kMaxDataSizeSamples); 55 if (out_length == -1) { 56 LOG_FERR3(LS_ERROR, Resample, audio_ptr, src_length, dst_frame->data_); 57 assert(false); 58 } 59 dst_frame->samples_per_channel_ = out_length / audio_ptr_num_channels; 60 61 // Upmix after resampling. 62 if (src_frame.num_channels_ == 1 && dst_frame->num_channels_ == 2) { 63 // The audio in dst_frame really is mono at this point; MonoToStereo will 64 // set this back to stereo. 65 dst_frame->num_channels_ = 1; 66 AudioFrameOperations::MonoToStereo(dst_frame); 67 } 68 69 dst_frame->timestamp_ = src_frame.timestamp_; 70 dst_frame->elapsed_time_ms_ = src_frame.elapsed_time_ms_; 71 dst_frame->ntp_time_ms_ = src_frame.ntp_time_ms_; 72} 73 74void DownConvertToCodecFormat(const int16_t* src_data, 75 int samples_per_channel, 76 int num_channels, 77 int sample_rate_hz, 78 int codec_num_channels, 79 int codec_rate_hz, 80 int16_t* mono_buffer, 81 PushResampler<int16_t>* resampler, 82 AudioFrame* dst_af) { 83 assert(samples_per_channel <= kMaxMonoDataSizeSamples); 84 assert(num_channels == 1 || num_channels == 2); 85 assert(codec_num_channels == 1 || codec_num_channels == 2); 86 dst_af->Reset(); 87 88 // Never upsample the capture signal here. This should be done at the 89 // end of the send chain. 90 int destination_rate = std::min(codec_rate_hz, sample_rate_hz); 91 92 // If no stereo codecs are in use, we downmix a stereo stream from the 93 // device early in the chain, before resampling. 94 if (num_channels == 2 && codec_num_channels == 1) { 95 AudioFrameOperations::StereoToMono(src_data, samples_per_channel, 96 mono_buffer); 97 src_data = mono_buffer; 98 num_channels = 1; 99 } 100 101 if (resampler->InitializeIfNeeded( 102 sample_rate_hz, destination_rate, num_channels) != 0) { 103 LOG_FERR3(LS_ERROR, 104 InitializeIfNeeded, 105 sample_rate_hz, 106 destination_rate, 107 num_channels); 108 assert(false); 109 } 110 111 const int in_length = samples_per_channel * num_channels; 112 int out_length = resampler->Resample( 113 src_data, in_length, dst_af->data_, AudioFrame::kMaxDataSizeSamples); 114 if (out_length == -1) { 115 LOG_FERR3(LS_ERROR, Resample, src_data, in_length, dst_af->data_); 116 assert(false); 117 } 118 119 dst_af->samples_per_channel_ = out_length / num_channels; 120 dst_af->sample_rate_hz_ = destination_rate; 121 dst_af->num_channels_ = num_channels; 122} 123 124void MixWithSat(int16_t target[], 125 int target_channel, 126 const int16_t source[], 127 int source_channel, 128 int source_len) { 129 assert(target_channel == 1 || target_channel == 2); 130 assert(source_channel == 1 || source_channel == 2); 131 132 if (target_channel == 2 && source_channel == 1) { 133 // Convert source from mono to stereo. 134 int32_t left = 0; 135 int32_t right = 0; 136 for (int i = 0; i < source_len; ++i) { 137 left = source[i] + target[i * 2]; 138 right = source[i] + target[i * 2 + 1]; 139 target[i * 2] = WebRtcSpl_SatW32ToW16(left); 140 target[i * 2 + 1] = WebRtcSpl_SatW32ToW16(right); 141 } 142 } else if (target_channel == 1 && source_channel == 2) { 143 // Convert source from stereo to mono. 144 int32_t temp = 0; 145 for (int i = 0; i < source_len / 2; ++i) { 146 temp = ((source[i * 2] + source[i * 2 + 1]) >> 1) + target[i]; 147 target[i] = WebRtcSpl_SatW32ToW16(temp); 148 } 149 } else { 150 int32_t temp = 0; 151 for (int i = 0; i < source_len; ++i) { 152 temp = source[i] + target[i]; 153 target[i] = WebRtcSpl_SatW32ToW16(temp); 154 } 155 } 156} 157 158} // namespace voe 159} // namespace webrtc 160