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/g722/audio_encoder_g722.h"
12
13 #include <memory>
14 #include <vector>
15
16 #include "absl/strings/match.h"
17 #include "modules/audio_coding/codecs/g722/audio_encoder_g722.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<AudioEncoderG722Config> AudioEncoderG722::SdpToConfig(
25 const SdpAudioFormat& format) {
26 if (!absl::EqualsIgnoreCase(format.name, "g722") ||
27 format.clockrate_hz != 8000) {
28 return absl::nullopt;
29 }
30
31 AudioEncoderG722Config config;
32 config.num_channels = rtc::checked_cast<int>(format.num_channels);
33 auto ptime_iter = format.parameters.find("ptime");
34 if (ptime_iter != format.parameters.end()) {
35 auto ptime = rtc::StringToNumber<int>(ptime_iter->second);
36 if (ptime && *ptime > 0) {
37 const int whole_packets = *ptime / 10;
38 config.frame_size_ms = rtc::SafeClamp<int>(whole_packets * 10, 10, 60);
39 }
40 }
41 return config.IsOk() ? absl::optional<AudioEncoderG722Config>(config)
42 : absl::nullopt;
43 }
44
AppendSupportedEncoders(std::vector<AudioCodecSpec> * specs)45 void AudioEncoderG722::AppendSupportedEncoders(
46 std::vector<AudioCodecSpec>* specs) {
47 const SdpAudioFormat fmt = {"G722", 8000, 1};
48 const AudioCodecInfo info = QueryAudioEncoder(*SdpToConfig(fmt));
49 specs->push_back({fmt, info});
50 }
51
QueryAudioEncoder(const AudioEncoderG722Config & config)52 AudioCodecInfo AudioEncoderG722::QueryAudioEncoder(
53 const AudioEncoderG722Config& config) {
54 RTC_DCHECK(config.IsOk());
55 return {16000, rtc::dchecked_cast<size_t>(config.num_channels),
56 64000 * config.num_channels};
57 }
58
MakeAudioEncoder(const AudioEncoderG722Config & config,int payload_type,absl::optional<AudioCodecPairId>)59 std::unique_ptr<AudioEncoder> AudioEncoderG722::MakeAudioEncoder(
60 const AudioEncoderG722Config& config,
61 int payload_type,
62 absl::optional<AudioCodecPairId> /*codec_pair_id*/) {
63 RTC_DCHECK(config.IsOk());
64 return std::make_unique<AudioEncoderG722Impl>(config, payload_type);
65 }
66
67 } // namespace webrtc
68