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) 2016 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 "modules/audio_coding/codecs/audio_format_conversion.h"
#include <string.h>
#include "absl/types/optional.h"
#include "api/array_view.h"
#include "rtc_base/checks.h"
#include "rtc_base/numerics/safe_conversions.h"
#include "rtc_base/sanitizer.h"
namespace webrtc {
namespace {
CodecInst MakeCodecInst(int payload_type,
const char* name,
int sample_rate,
size_t num_channels) {
// Create a CodecInst with some fields set. The remaining fields are zeroed,
// but we tell MSan to consider them uninitialized.
CodecInst ci = {0};
rtc::MsanMarkUninitialized(rtc::MakeArrayView(&ci, 1));
ci.pltype = payload_type;
strncpy(ci.plname, name, sizeof(ci.plname));
ci.plname[sizeof(ci.plname) - 1] = '\0';
ci.plfreq = sample_rate;
ci.channels = num_channels;
return ci;
}
} // namespace
SdpAudioFormat CodecInstToSdp(const CodecInst& ci) {
if (STR_CASE_CMP(ci.plname, "g722") == 0) {
RTC_CHECK_EQ(16000, ci.plfreq);
RTC_CHECK(ci.channels == 1 || ci.channels == 2);
return {"g722", 8000, ci.channels};
} else if (STR_CASE_CMP(ci.plname, "opus") == 0) {
RTC_CHECK_EQ(48000, ci.plfreq);
RTC_CHECK(ci.channels == 1 || ci.channels == 2);
return ci.channels == 1
? SdpAudioFormat("opus", 48000, 2)
: SdpAudioFormat("opus", 48000, 2, {{"stereo", "1"}});
} else {
return {ci.plname, ci.plfreq, ci.channels};
}
}
CodecInst SdpToCodecInst(int payload_type, const SdpAudioFormat& audio_format) {
if (STR_CASE_CMP(audio_format.name.c_str(), "g722") == 0) {
RTC_CHECK_EQ(8000, audio_format.clockrate_hz);
RTC_CHECK(audio_format.num_channels == 1 || audio_format.num_channels == 2);
return MakeCodecInst(payload_type, "g722", 16000,
audio_format.num_channels);
} else if (STR_CASE_CMP(audio_format.name.c_str(), "opus") == 0) {
RTC_CHECK_EQ(48000, audio_format.clockrate_hz);
RTC_CHECK_EQ(2, audio_format.num_channels);
const int num_channels = [&] {
auto stereo = audio_format.parameters.find("stereo");
if (stereo != audio_format.parameters.end()) {
if (stereo->second == "0") {
return 1;
} else if (stereo->second == "1") {
return 2;
} else {
RTC_CHECK(false); // Bad stereo parameter.
}
}
return 1; // Default to mono.
}();
return MakeCodecInst(payload_type, "opus", 48000, num_channels);
} else {
return MakeCodecInst(payload_type, audio_format.name.c_str(),
audio_format.clockrate_hz, audio_format.num_channels);
}
}
} // namespace webrtc
|
