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
92
93
94
95
96
97
98
|
/*
* 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 "webrtc/modules/audio_coding/include/audio_coding_module.h"
#include "webrtc/base/checks.h"
#include "webrtc/common_types.h"
#include "webrtc/modules/audio_coding/acm2/audio_coding_module_impl.h"
#include "webrtc/modules/audio_coding/acm2/rent_a_codec.h"
#include "webrtc/system_wrappers/include/clock.h"
#include "webrtc/system_wrappers/include/trace.h"
namespace webrtc {
// Create module
AudioCodingModule* AudioCodingModule::Create(int id) {
Config config;
config.id = id;
config.clock = Clock::GetRealTimeClock();
return Create(config);
}
AudioCodingModule* AudioCodingModule::Create(int id, Clock* clock) {
Config config;
config.id = id;
config.clock = clock;
return Create(config);
}
AudioCodingModule* AudioCodingModule::Create(const Config& config) {
return new acm2::AudioCodingModuleImpl(config);
}
int AudioCodingModule::NumberOfCodecs() {
return static_cast<int>(acm2::RentACodec::NumberOfCodecs());
}
int AudioCodingModule::Codec(int list_id, CodecInst* codec) {
auto codec_id = acm2::RentACodec::CodecIdFromIndex(list_id);
if (!codec_id)
return -1;
auto ci = acm2::RentACodec::CodecInstById(*codec_id);
if (!ci)
return -1;
*codec = *ci;
return 0;
}
int AudioCodingModule::Codec(const char* payload_name,
CodecInst* codec,
int sampling_freq_hz,
size_t channels) {
rtc::Optional<CodecInst> ci = acm2::RentACodec::CodecInstByParams(
payload_name, sampling_freq_hz, channels);
if (ci) {
*codec = *ci;
return 0;
} else {
// We couldn't find a matching codec, so set the parameters to unacceptable
// values and return.
codec->plname[0] = '\0';
codec->pltype = -1;
codec->pacsize = 0;
codec->rate = 0;
codec->plfreq = 0;
return -1;
}
}
int AudioCodingModule::Codec(const char* payload_name,
int sampling_freq_hz,
size_t channels) {
rtc::Optional<acm2::RentACodec::CodecId> ci =
acm2::RentACodec::CodecIdByParams(payload_name, sampling_freq_hz,
channels);
if (!ci)
return -1;
rtc::Optional<int> i = acm2::RentACodec::CodecIndexFromId(*ci);
return i ? *i : -1;
}
// Checks the validity of the parameters of the given codec
bool AudioCodingModule::IsCodecValid(const CodecInst& codec) {
bool valid = acm2::RentACodec::IsCodecValid(codec);
if (!valid)
WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, -1,
"Invalid codec setting");
return valid;
}
} // namespace webrtc
|
