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Revert of Opus implementation of the AudioDecoderFactoryTemplate API (patchset #1 id:1 of https://codereview.webrtc.org/2942733003/ )

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Reason for revert:
breaking downstream projects

Original issue's description:
> Opus implementation of the AudioDecoderFactoryTemplate API
>
> BUG=webrtc:7837
>
> Review-Url: https://codereview.webrtc.org/2942733003
> Cr-Commit-Position: refs/heads/master@{#18646}
> Committed: d053fe4ab3

TBR=ossu@webrtc.org,solenberg@webrtc.org,kwiberg@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7837

Review-Url: https://codereview.webrtc.org/2944763002
Cr-Commit-Position: refs/heads/master@{#18648}
This commit is contained in:
charujain
2017-06-18 02:37:17 -07:00
committed by Commit Bot
parent af6293517f
commit eb2d2d31d1
11 changed files with 33 additions and 165 deletions
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62
webrtc/api/audio_codecs/opus/audio_decoder_opus.cc
View File

@ -1,62 +0,0 @@
/*
* 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/opus/audio_decoder_opus.h"
#include <memory>
#include <utility>
#include <vector>
#include "webrtc/base/ptr_util.h"
#include "webrtc/common_types.h"
#include "webrtc/modules/audio_coding/codecs/opus/audio_decoder_opus.h"
namespace webrtc {
rtc::Optional<AudioDecoderOpus::Config> AudioDecoderOpus::SdpToConfig(
const SdpAudioFormat& format) {
const rtc::Optional<int> num_channels = [&] {
auto stereo = format.parameters.find("stereo");
if (stereo != format.parameters.end()) {
if (stereo->second == "0") {
return rtc::Optional<int>(1);
} else if (stereo->second == "1") {
return rtc::Optional<int>(2);
} else {
return rtc::Optional<int>(); // Bad stereo parameter.
}
}
return rtc::Optional<int>(1); // Default to mono.
}();
if (STR_CASE_CMP(format.name.c_str(), "opus") == 0 &&
format.clockrate_hz == 48000 && format.num_channels == 2 &&
num_channels) {
return rtc::Optional<Config>(Config{*num_channels});
} else {
return rtc::Optional<Config>();
}
}
void AudioDecoderOpus::AppendSupportedDecoders(
std::vector<AudioCodecSpec>* specs) {
AudioCodecInfo opus_info{48000, 1, 64000, 6000, 510000};
opus_info.allow_comfort_noise = false;
opus_info.supports_network_adaption = true;
SdpAudioFormat opus_format(
{"opus", 48000, 2, {{"minptime", "10"}, {"useinbandfec", "1"}}});
specs->push_back({std::move(opus_format), std::move(opus_info)});
}
std::unique_ptr<AudioDecoder> AudioDecoderOpus::MakeAudioDecoder(
Config config) {
return rtc::MakeUnique<AudioDecoderOpusImpl>(config.num_channels);
}
} // namespace webrtc