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
|
/*
* 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 <array>
#include <cstdint>
#include "api/audio/audio_frame.h"
#include "api/audio/audio_view.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_view(), 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(InterleavedView<const int16_t> src_data,
int sample_rate_hz,
PushResampler<int16_t>* resampler,
AudioFrame* dst_frame) {
// The `samples_per_channel_` members must have been set correctly based on
// the associated sample rate and the assumed 10ms buffer size.
// TODO(tommi): Remove the `sample_rate_hz` param.
RTC_DCHECK_EQ(SampleRateToDefaultChannelSize(sample_rate_hz),
src_data.samples_per_channel());
RTC_DCHECK_EQ(SampleRateToDefaultChannelSize(dst_frame->sample_rate_hz_),
dst_frame->samples_per_channel());
// Temporary buffer in case downmixing is required.
std::array<int16_t, AudioFrame::kMaxDataSizeSamples> downmixed_audio;
// Downmix before resampling.
if (src_data.num_channels() > dst_frame->num_channels_) {
RTC_DCHECK(src_data.num_channels() == 2 || src_data.num_channels() == 4)
<< "num_channels: " << src_data.num_channels();
RTC_DCHECK(dst_frame->num_channels_ == 1 || dst_frame->num_channels_ == 2)
<< "dst_frame->num_channels_: " << dst_frame->num_channels_;
InterleavedView<int16_t> downmixed(downmixed_audio.data(),
src_data.samples_per_channel(),
dst_frame->num_channels_);
AudioFrameOperations::DownmixChannels(src_data, downmixed);
src_data = downmixed;
}
// 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.
// Stash away the originally requested number of channels. Then provide
// `dst_frame` as a target buffer with the same number of channels as the
// source.
auto original_dst_number_of_channels = dst_frame->num_channels_;
resampler->Resample(src_data,
dst_frame->mutable_data(dst_frame->samples_per_channel_,
src_data.num_channels()));
// Upmix after resampling.
if (src_data.num_channels() == 1 && original_dst_number_of_channels == 2) {
// The audio in dst_frame really is mono at this point; MonoToStereo will
// set this back to stereo.
RTC_DCHECK_EQ(dst_frame->num_channels_, 1);
AudioFrameOperations::UpmixChannels(2, dst_frame);
}
}
} // namespace voe
} // namespace webrtc
|
