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