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Stop using old AudioCodingModule::RegisterReceiveCodec overloads

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BUG=webrtc:5801

Review-Url: https://codereview.webrtc.org/2388153004
Cr-Commit-Position: refs/heads/master@{#14753}
This commit is contained in:
kwiberg
2016-10-24 13:47:09 -07:00
committed by Commit bot
parent 88b7074745
commit da2bf4e150
24 changed files with 202 additions and 78 deletions
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30
webrtc/modules/audio_coding/codecs/audio_format_conversion.cc Normal file
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/*
* 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

23
webrtc/modules/audio_coding/codecs/audio_format_conversion.h Normal file
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/*
* 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.
*/
#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_AUDIO_FORMAT_CONVERSION_H_
#define WEBRTC_MODULES_AUDIO_CODING_CODECS_AUDIO_FORMAT_CONVERSION_H_
#include "webrtc/common_types.h"
#include "webrtc/modules/audio_coding/codecs/audio_format.h"
namespace webrtc {
SdpAudioFormat CodecInstToSdp(const CodecInst& codec_inst);
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_AUDIO_FORMAT_CONVERSION_H_