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
74
75
76
77
78
79
80
81
82
83
84
85
86
87
88
89
90
91
|
/*
* 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 "audio/remix_resample.h"
#include "api/audio/audio_frame.h"
#include "audio/utility/audio_frame_operations.h"
#include "common_audio/resampler/include/push_resampler.h"
#include "rtc_base/checks.h"
namespace webrtc {
namespace voe {
void RemixAndResample(const AudioFrame& src_frame,
PushResampler<int16_t>* resampler,
AudioFrame* dst_frame) {
RemixAndResample(src_frame.data(), src_frame.samples_per_channel_,
src_frame.num_channels_, src_frame.sample_rate_hz_,
resampler, dst_frame);
dst_frame->timestamp_ = src_frame.timestamp_;
dst_frame->elapsed_time_ms_ = src_frame.elapsed_time_ms_;
dst_frame->ntp_time_ms_ = src_frame.ntp_time_ms_;
dst_frame->packet_infos_ = src_frame.packet_infos_;
}
void 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) {
const int16_t* audio_ptr = src_data;
size_t audio_ptr_num_channels = num_channels;
int16_t downmixed_audio[AudioFrame::kMaxDataSizeSamples];
// Downmix before resampling.
if (num_channels > dst_frame->num_channels_) {
RTC_DCHECK(num_channels == 2 || num_channels == 4)
<< "num_channels: " << num_channels;
RTC_DCHECK(dst_frame->num_channels_ == 1 || dst_frame->num_channels_ == 2)
<< "dst_frame->num_channels_: " << dst_frame->num_channels_;
AudioFrameOperations::DownmixChannels(
src_data, num_channels, samples_per_channel, dst_frame->num_channels_,
downmixed_audio);
audio_ptr = downmixed_audio;
audio_ptr_num_channels = dst_frame->num_channels_;
}
if (resampler->InitializeIfNeeded(sample_rate_hz, dst_frame->sample_rate_hz_,
audio_ptr_num_channels) == -1) {
RTC_FATAL() << "InitializeIfNeeded failed: sample_rate_hz = "
<< sample_rate_hz << ", dst_frame->sample_rate_hz_ = "
<< dst_frame->sample_rate_hz_
<< ", audio_ptr_num_channels = " << audio_ptr_num_channels;
}
// TODO(yujo): for muted input frames, don't resample. Either 1) allow
// resampler to return output length without doing the resample, so we know
// how much to zero here; or 2) make resampler accept a hint that the input is
// zeroed.
const size_t src_length = samples_per_channel * audio_ptr_num_channels;
int out_length =
resampler->Resample(audio_ptr, src_length, dst_frame->mutable_data(),
AudioFrame::kMaxDataSizeSamples);
if (out_length == -1) {
RTC_FATAL() << "Resample failed: audio_ptr = " << audio_ptr
<< ", src_length = " << src_length
<< ", dst_frame->mutable_data() = "
<< dst_frame->mutable_data();
}
dst_frame->samples_per_channel_ = out_length / audio_ptr_num_channels;
// Upmix after resampling.
if (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::UpmixChannels(2, dst_frame);
}
}
} // namespace voe
} // namespace webrtc
|
