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/L16/audio_encoder_L16.h"
12
13 #include <memory>
14
15 #include "absl/strings/match.h"
16 #include "modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.h"
17 #include "modules/audio_coding/codecs/pcm16b/pcm16b_common.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<AudioEncoderL16::Config> AudioEncoderL16::SdpToConfig(
25 const SdpAudioFormat& format) {
26 if (!rtc::IsValueInRangeForNumericType<int>(format.num_channels)) {
27 RTC_DCHECK_NOTREACHED();
28 return absl::nullopt;
29 }
30 Config config;
31 config.sample_rate_hz = format.clockrate_hz;
32 config.num_channels = rtc::dchecked_cast<int>(format.num_channels);
33 auto ptime_iter = format.parameters.find("ptime");
34 if (ptime_iter != format.parameters.end()) {
35 const auto ptime = rtc::StringToNumber<int>(ptime_iter->second);
36 if (ptime && *ptime > 0) {
37 config.frame_size_ms = rtc::SafeClamp(10 * (*ptime / 10), 10, 60);
38 }
39 }
40 if (absl::EqualsIgnoreCase(format.name, "L16") && config.IsOk()) {
41 return config;
42 }
43 return absl::nullopt;
44 }
45
AppendSupportedEncoders(std::vector<AudioCodecSpec> * specs)46 void AudioEncoderL16::AppendSupportedEncoders(
47 std::vector<AudioCodecSpec>* specs) {
48 Pcm16BAppendSupportedCodecSpecs(specs);
49 }
50
QueryAudioEncoder(const AudioEncoderL16::Config & config)51 AudioCodecInfo AudioEncoderL16::QueryAudioEncoder(
52 const AudioEncoderL16::Config& config) {
53 RTC_DCHECK(config.IsOk());
54 return {config.sample_rate_hz,
55 rtc::dchecked_cast<size_t>(config.num_channels),
56 config.sample_rate_hz * config.num_channels * 16};
57 }
58
MakeAudioEncoder(const AudioEncoderL16::Config & config,int payload_type,absl::optional<AudioCodecPairId>,const FieldTrialsView * field_trials)59 std::unique_ptr<AudioEncoder> AudioEncoderL16::MakeAudioEncoder(
60 const AudioEncoderL16::Config& config,
61 int payload_type,
62 absl::optional<AudioCodecPairId> /*codec_pair_id*/,
63 const FieldTrialsView* field_trials) {
64 AudioEncoderPcm16B::Config c;
65 c.sample_rate_hz = config.sample_rate_hz;
66 c.num_channels = config.num_channels;
67 c.frame_size_ms = config.frame_size_ms;
68 c.payload_type = payload_type;
69 if (!config.IsOk()) {
70 RTC_DCHECK_NOTREACHED();
71 return nullptr;
72 }
73 return std::make_unique<AudioEncoderPcm16B>(c);
74 }
75
76 } // namespace webrtc
77