| /* |
| * Copyright (c) 2017 The WebRTC project authors. All Rights Reserved. |
| * |
| * Use of this source code is governed by a BSD-style license |
| * that can be found in the LICENSE file in the root of the source |
| * tree. An additional intellectual property rights grant can be found |
| * in the file PATENTS. All contributing project authors may |
| * be found in the AUTHORS file in the root of the source tree. |
| */ |
| |
| #include "api/audio_codecs/L16/audio_encoder_L16.h" |
| |
| #include <memory> |
| |
| #include "absl/strings/match.h" |
| #include "modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.h" |
| #include "modules/audio_coding/codecs/pcm16b/pcm16b_common.h" |
| #include "rtc_base/numerics/safe_conversions.h" |
| #include "rtc_base/numerics/safe_minmax.h" |
| #include "rtc_base/string_to_number.h" |
| |
| namespace webrtc { |
| |
| absl::optional<AudioEncoderL16::Config> AudioEncoderL16::SdpToConfig( |
| const SdpAudioFormat& format) { |
| if (!rtc::IsValueInRangeForNumericType<int>(format.num_channels)) { |
| RTC_DCHECK_NOTREACHED(); |
| return absl::nullopt; |
| } |
| Config config; |
| config.sample_rate_hz = format.clockrate_hz; |
| config.num_channels = rtc::dchecked_cast<int>(format.num_channels); |
| auto ptime_iter = format.parameters.find("ptime"); |
| if (ptime_iter != format.parameters.end()) { |
| const auto ptime = rtc::StringToNumber<int>(ptime_iter->second); |
| if (ptime && *ptime > 0) { |
| config.frame_size_ms = rtc::SafeClamp(10 * (*ptime / 10), 10, 60); |
| } |
| } |
| if (absl::EqualsIgnoreCase(format.name, "L16") && config.IsOk()) { |
| return config; |
| } |
| return absl::nullopt; |
| } |
| |
| void AudioEncoderL16::AppendSupportedEncoders( |
| std::vector<AudioCodecSpec>* specs) { |
| Pcm16BAppendSupportedCodecSpecs(specs); |
| } |
| |
| AudioCodecInfo AudioEncoderL16::QueryAudioEncoder( |
| const AudioEncoderL16::Config& config) { |
| RTC_DCHECK(config.IsOk()); |
| return {config.sample_rate_hz, |
| rtc::dchecked_cast<size_t>(config.num_channels), |
| config.sample_rate_hz * config.num_channels * 16}; |
| } |
| |
| std::unique_ptr<AudioEncoder> AudioEncoderL16::MakeAudioEncoder( |
| const AudioEncoderL16::Config& config, |
| int payload_type, |
| absl::optional<AudioCodecPairId> /*codec_pair_id*/, |
| const FieldTrialsView* field_trials) { |
| AudioEncoderPcm16B::Config c; |
| c.sample_rate_hz = config.sample_rate_hz; |
| c.num_channels = config.num_channels; |
| c.frame_size_ms = config.frame_size_ms; |
| c.payload_type = payload_type; |
| if (!config.IsOk()) { |
| RTC_DCHECK_NOTREACHED(); |
| return nullptr; |
| } |
| return std::make_unique<AudioEncoderPcm16B>(c); |
| } |
| |
| } // namespace webrtc |