| /* |
| * 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 "webrtc/api/audio_codecs/isac/audio_encoder_isac_fix.h" |
| |
| #include "webrtc/common_types.h" |
| #include "webrtc/modules/audio_coding/codecs/isac/fix/include/audio_encoder_isacfix.h" |
| #include "webrtc/rtc_base/ptr_util.h" |
| #include "webrtc/rtc_base/string_to_number.h" |
| |
| namespace webrtc { |
| |
| rtc::Optional<AudioEncoderIsacFix::Config> AudioEncoderIsacFix::SdpToConfig( |
| const SdpAudioFormat& format) { |
| if (STR_CASE_CMP(format.name.c_str(), "ISAC") == 0 && |
| format.clockrate_hz == 16000 && format.num_channels == 1) { |
| Config config; |
| const 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 >= 60) { |
| config.frame_size_ms = 60; |
| } |
| } |
| return rtc::Optional<Config>(config); |
| } else { |
| return rtc::Optional<Config>(); |
| } |
| } |
| |
| void AudioEncoderIsacFix::AppendSupportedEncoders( |
| std::vector<AudioCodecSpec>* specs) { |
| const SdpAudioFormat fmt = {"ISAC", 16000, 1}; |
| const AudioCodecInfo info = QueryAudioEncoder(*SdpToConfig(fmt)); |
| specs->push_back({fmt, info}); |
| } |
| |
| AudioCodecInfo AudioEncoderIsacFix::QueryAudioEncoder( |
| AudioEncoderIsacFix::Config config) { |
| RTC_DCHECK(config.IsOk()); |
| return {16000, 1, 32000, 10000, 32000}; |
| } |
| |
| std::unique_ptr<AudioEncoder> AudioEncoderIsacFix::MakeAudioEncoder( |
| AudioEncoderIsacFix::Config config, |
| int payload_type) { |
| RTC_DCHECK(config.IsOk()); |
| AudioEncoderIsacFixImpl::Config c; |
| c.frame_size_ms = config.frame_size_ms; |
| c.payload_type = payload_type; |
| return rtc::MakeUnique<AudioEncoderIsacFixImpl>(c); |
| } |
| |
| } // namespace webrtc |