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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     return absl::nullopt;
28   }
29   Config config;
30   config.sample_rate_hz = format.clockrate_hz;
31   config.num_channels = rtc::dchecked_cast<int>(format.num_channels);
32   auto ptime_iter = format.parameters.find("ptime");
33   if (ptime_iter != format.parameters.end()) {
34     const auto ptime = rtc::StringToNumber<int>(ptime_iter->second);
35     if (ptime && *ptime > 0) {
36       config.frame_size_ms = rtc::SafeClamp(10 * (*ptime / 10), 10, 60);
37     }
38   }
39   return absl::EqualsIgnoreCase(format.name, "L16") && config.IsOk()
40              ? absl::optional<Config>(config)
41              : absl::nullopt;
42 }
43 
AppendSupportedEncoders(std::vector<AudioCodecSpec> * specs)44 void AudioEncoderL16::AppendSupportedEncoders(
45     std::vector<AudioCodecSpec>* specs) {
46   Pcm16BAppendSupportedCodecSpecs(specs);
47 }
48 
QueryAudioEncoder(const AudioEncoderL16::Config & config)49 AudioCodecInfo AudioEncoderL16::QueryAudioEncoder(
50     const AudioEncoderL16::Config& config) {
51   RTC_DCHECK(config.IsOk());
52   return {config.sample_rate_hz,
53           rtc::dchecked_cast<size_t>(config.num_channels),
54           config.sample_rate_hz * config.num_channels * 16};
55 }
56 
MakeAudioEncoder(const AudioEncoderL16::Config & config,int payload_type,absl::optional<AudioCodecPairId>)57 std::unique_ptr<AudioEncoder> AudioEncoderL16::MakeAudioEncoder(
58     const AudioEncoderL16::Config& config,
59     int payload_type,
60     absl::optional<AudioCodecPairId> /*codec_pair_id*/) {
61   RTC_DCHECK(config.IsOk());
62   AudioEncoderPcm16B::Config c;
63   c.sample_rate_hz = config.sample_rate_hz;
64   c.num_channels = config.num_channels;
65   c.frame_size_ms = config.frame_size_ms;
66   c.payload_type = payload_type;
67   return std::make_unique<AudioEncoderPcm16B>(c);
68 }
69 
70 }  // namespace webrtc
71