| /* |
| * Copyright (c) 2016 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/modules/audio_coding/codecs/audio_format_conversion.h" |
| |
| #include "webrtc/base/checks.h" |
| #include "webrtc/base/safe_conversions.h" |
| |
| namespace webrtc { |
| |
| SdpAudioFormat CodecInstToSdp(const CodecInst& ci) { |
| if (STR_CASE_CMP(ci.plname, "g722") == 0 && ci.plfreq == 16000) { |
| RTC_CHECK(ci.channels == 1 || ci.channels == 2); |
| return {"g722", 8000, static_cast<int>(ci.channels)}; |
| } else if (STR_CASE_CMP(ci.plname, "opus") == 0 && ci.plfreq == 48000) { |
| RTC_CHECK(ci.channels == 1 || ci.channels == 2); |
| return {"opus", 48000, 2, {{"stereo", ci.channels == 1 ? "0" : "1"}}}; |
| } else { |
| return {ci.plname, ci.plfreq, rtc::checked_cast<int>(ci.channels)}; |
| } |
| } |
| |
| } // namespace webrtc |
