1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
|
/*
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "output_mixer_internal.h"
#include "audio_frame_operations.h"
#include "common_audio/resampler/include/resampler.h"
#include "module_common_types.h"
#include "trace.h"
namespace webrtc {
namespace voe {
int RemixAndResample(const AudioFrame& src_frame,
Resampler* resampler,
AudioFrame* dst_frame) {
const int16_t* audio_ptr = src_frame.data_;
int audio_ptr_num_channels = src_frame.num_channels_;
int16_t mono_audio[AudioFrame::kMaxDataSizeSamples];
// Downmix before resampling.
if (src_frame.num_channels_ == 2 && dst_frame->num_channels_ == 1) {
AudioFrameOperations::StereoToMono(src_frame.data_,
src_frame.samples_per_channel_,
mono_audio);
audio_ptr = mono_audio;
audio_ptr_num_channels = 1;
}
const ResamplerType resampler_type = audio_ptr_num_channels == 1 ?
kResamplerSynchronous : kResamplerSynchronousStereo;
if (resampler->ResetIfNeeded(src_frame.sample_rate_hz_,
dst_frame->sample_rate_hz_,
resampler_type) == -1) {
dst_frame->CopyFrom(src_frame);
WEBRTC_TRACE(kTraceError, kTraceVoice, -1,
"%s ResetIfNeeded failed", __FUNCTION__);
return -1;
}
int out_length = 0;
if (resampler->Push(audio_ptr,
src_frame.samples_per_channel_* audio_ptr_num_channels,
dst_frame->data_,
AudioFrame::kMaxDataSizeSamples,
out_length) == 0) {
dst_frame->samples_per_channel_ = out_length / audio_ptr_num_channels;
} else {
dst_frame->CopyFrom(src_frame);
WEBRTC_TRACE(kTraceError, kTraceVoice, -1,
"%s resampling failed", __FUNCTION__);
return -1;
}
// Upmix after resampling.
if (src_frame.num_channels_ == 1 && dst_frame->num_channels_ == 2) {
// The audio in dst_frame really is mono at this point; MonoToStereo will
// set this back to stereo.
dst_frame->num_channels_ = 1;
AudioFrameOperations::MonoToStereo(dst_frame);
}
return 0;
}
} // namespace voe
} // namespace webrtc
|