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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/base/logging.h"
14 #include "webrtc/common_audio/resampler/include/push_resampler.h"
15 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
16 #include "webrtc/common_types.h"
17 #include "webrtc/modules/include/module_common_types.h"
18 #include "webrtc/modules/utility/include/audio_frame_operations.h"
19 #include "webrtc/voice_engine/voice_engine_defines.h"
20 
21 namespace webrtc {
22 namespace voe {
23 
RemixAndResample(const AudioFrame & src_frame,PushResampler<int16_t> * resampler,AudioFrame * dst_frame)24 void RemixAndResample(const AudioFrame& src_frame,
25                       PushResampler<int16_t>* resampler,
26                       AudioFrame* dst_frame) {
27   RemixAndResample(src_frame.data_, src_frame.samples_per_channel_,
28                    src_frame.num_channels_, src_frame.sample_rate_hz_,
29                    resampler, dst_frame);
30   dst_frame->timestamp_ = src_frame.timestamp_;
31   dst_frame->elapsed_time_ms_ = src_frame.elapsed_time_ms_;
32   dst_frame->ntp_time_ms_ = src_frame.ntp_time_ms_;
33 }
34 
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)35 void RemixAndResample(const int16_t* src_data,
36                       size_t samples_per_channel,
37                       size_t num_channels,
38                       int sample_rate_hz,
39                       PushResampler<int16_t>* resampler,
40                       AudioFrame* dst_frame) {
41   const int16_t* audio_ptr = src_data;
42   size_t audio_ptr_num_channels = num_channels;
43   int16_t mono_audio[AudioFrame::kMaxDataSizeSamples];
44 
45   // Downmix before resampling.
46   if (num_channels == 2 && dst_frame->num_channels_ == 1) {
47     AudioFrameOperations::StereoToMono(src_data, samples_per_channel,
48                                        mono_audio);
49     audio_ptr = mono_audio;
50     audio_ptr_num_channels = 1;
51   }
52 
53   if (resampler->InitializeIfNeeded(sample_rate_hz, dst_frame->sample_rate_hz_,
54                                     audio_ptr_num_channels) == -1) {
55     LOG(LS_ERROR) << "InitializeIfNeeded failed: sample_rate_hz = "
56                   << sample_rate_hz << ", dst_frame->sample_rate_hz_ = "
57                   << dst_frame->sample_rate_hz_
58                   << ", audio_ptr_num_channels = " << audio_ptr_num_channels;
59     assert(false);
60   }
61 
62   const size_t src_length = samples_per_channel * audio_ptr_num_channels;
63   int out_length = resampler->Resample(audio_ptr, src_length, dst_frame->data_,
64                                        AudioFrame::kMaxDataSizeSamples);
65   if (out_length == -1) {
66     LOG(LS_ERROR) << "Resample failed: audio_ptr = " << audio_ptr
67                   << ", src_length = " << src_length
68                   << ", dst_frame->data_ = " << dst_frame->data_;
69     assert(false);
70   }
71   dst_frame->samples_per_channel_ = out_length / audio_ptr_num_channels;
72 
73   // Upmix after resampling.
74   if (num_channels == 1 && dst_frame->num_channels_ == 2) {
75     // The audio in dst_frame really is mono at this point; MonoToStereo will
76     // set this back to stereo.
77     dst_frame->num_channels_ = 1;
78     AudioFrameOperations::MonoToStereo(dst_frame);
79   }
80 }
81 
MixWithSat(int16_t target[],size_t target_channel,const int16_t source[],size_t source_channel,size_t source_len)82 void MixWithSat(int16_t target[],
83                 size_t target_channel,
84                 const int16_t source[],
85                 size_t source_channel,
86                 size_t source_len) {
87   assert(target_channel == 1 || target_channel == 2);
88   assert(source_channel == 1 || source_channel == 2);
89 
90   if (target_channel == 2 && source_channel == 1) {
91     // Convert source from mono to stereo.
92     int32_t left = 0;
93     int32_t right = 0;
94     for (size_t i = 0; i < source_len; ++i) {
95       left = source[i] + target[i * 2];
96       right = source[i] + target[i * 2 + 1];
97       target[i * 2] = WebRtcSpl_SatW32ToW16(left);
98       target[i * 2 + 1] = WebRtcSpl_SatW32ToW16(right);
99     }
100   } else if (target_channel == 1 && source_channel == 2) {
101     // Convert source from stereo to mono.
102     int32_t temp = 0;
103     for (size_t i = 0; i < source_len / 2; ++i) {
104       temp = ((source[i * 2] + source[i * 2 + 1]) >> 1) + target[i];
105       target[i] = WebRtcSpl_SatW32ToW16(temp);
106     }
107   } else {
108     int32_t temp = 0;
109     for (size_t i = 0; i < source_len; ++i) {
110       temp = source[i] + target[i];
111       target[i] = WebRtcSpl_SatW32ToW16(temp);
112     }
113   }
114 }
115 
116 }  // namespace voe
117 }  // namespace webrtc
118