1 /*
2 * Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
3 *
4 * Use of this source code is governed by a BSD-style license
5 * that can be found in the LICENSE file in the root of the source
6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree.
9 */
10
11 #include "api/audio_codecs/g711/audio_encoder_g711.h"
12
13 #include <memory>
14 #include <vector>
15
16 #include "absl/strings/match.h"
17 #include "modules/audio_coding/codecs/g711/audio_encoder_pcm.h"
18 #include "rtc_base/numerics/safe_conversions.h"
19 #include "rtc_base/numerics/safe_minmax.h"
20 #include "rtc_base/string_to_number.h"
21
22 namespace webrtc {
23
SdpToConfig(const SdpAudioFormat & format)24 absl::optional<AudioEncoderG711::Config> AudioEncoderG711::SdpToConfig(
25 const SdpAudioFormat& format) {
26 const bool is_pcmu = absl::EqualsIgnoreCase(format.name, "PCMU");
27 const bool is_pcma = absl::EqualsIgnoreCase(format.name, "PCMA");
28 if (format.clockrate_hz == 8000 && format.num_channels >= 1 &&
29 (is_pcmu || is_pcma)) {
30 Config config;
31 config.type = is_pcmu ? Config::Type::kPcmU : Config::Type::kPcmA;
32 config.num_channels = rtc::dchecked_cast<int>(format.num_channels);
33 config.frame_size_ms = 20;
34 auto ptime_iter = format.parameters.find("ptime");
35 if (ptime_iter != format.parameters.end()) {
36 const auto ptime = rtc::StringToNumber<int>(ptime_iter->second);
37 if (ptime && *ptime > 0) {
38 config.frame_size_ms = rtc::SafeClamp(10 * (*ptime / 10), 10, 60);
39 }
40 }
41 if (!config.IsOk()) {
42 RTC_DCHECK_NOTREACHED();
43 return absl::nullopt;
44 }
45 return config;
46 } else {
47 return absl::nullopt;
48 }
49 }
50
AppendSupportedEncoders(std::vector<AudioCodecSpec> * specs)51 void AudioEncoderG711::AppendSupportedEncoders(
52 std::vector<AudioCodecSpec>* specs) {
53 for (const char* type : {"PCMU", "PCMA"}) {
54 specs->push_back({{type, 8000, 1}, {8000, 1, 64000}});
55 }
56 }
57
QueryAudioEncoder(const Config & config)58 AudioCodecInfo AudioEncoderG711::QueryAudioEncoder(const Config& config) {
59 RTC_DCHECK(config.IsOk());
60 return {8000, rtc::dchecked_cast<size_t>(config.num_channels),
61 64000 * config.num_channels};
62 }
63
MakeAudioEncoder(const Config & config,int payload_type,absl::optional<AudioCodecPairId>,const FieldTrialsView * field_trials)64 std::unique_ptr<AudioEncoder> AudioEncoderG711::MakeAudioEncoder(
65 const Config& config,
66 int payload_type,
67 absl::optional<AudioCodecPairId> /*codec_pair_id*/,
68 const FieldTrialsView* field_trials) {
69 if (!config.IsOk()) {
70 RTC_DCHECK_NOTREACHED();
71 return nullptr;
72 }
73 switch (config.type) {
74 case Config::Type::kPcmU: {
75 AudioEncoderPcmU::Config impl_config;
76 impl_config.num_channels = config.num_channels;
77 impl_config.frame_size_ms = config.frame_size_ms;
78 impl_config.payload_type = payload_type;
79 return std::make_unique<AudioEncoderPcmU>(impl_config);
80 }
81 case Config::Type::kPcmA: {
82 AudioEncoderPcmA::Config impl_config;
83 impl_config.num_channels = config.num_channels;
84 impl_config.frame_size_ms = config.frame_size_ms;
85 impl_config.payload_type = payload_type;
86 return std::make_unique<AudioEncoderPcmA>(impl_config);
87 }
88 default: {
89 RTC_DCHECK_NOTREACHED();
90 return nullptr;
91 }
92 }
93 }
94
95 } // namespace webrtc
96