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)24absl::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)44void AudioEncoderL16::AppendSupportedEncoders( 45 std::vector<AudioCodecSpec>* specs) { 46 Pcm16BAppendSupportedCodecSpecs(specs); 47 } 48 QueryAudioEncoder(const AudioEncoderL16::Config & config)49AudioCodecInfo 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>)57std::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