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fbeb76ab51
This reverts commit b46c4bf27b.
Reason for revert: breaks a downstream project
Original change's description:
> [ACM] iSAC audio codec removed
>
> Note: this CL has to leave behind one part of iSAC, which is its VAD
> currently used by AGC1 in APM. The target visibility has been
> restricted and the VAD will be removed together with AGC1 when the
> time comes.
>
> Tested: see https://chromium-review.googlesource.com/c/chromium/src/+/4013319
>
> Bug: webrtc:14450
> Change-Id: I69cc518b16280eae62a1f1977cdbfa24c08cf5f9
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/282421
> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
> Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#38652}
Bug: webrtc:14450
Change-Id: Ice138004e84e8c5f896684e8d01133d4b2a77bb7
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/283800
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Auto-Submit: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Owners-Override: Mirko Bonadei <mbonadei@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#38655}
69 lines
2.1 KiB
C++
69 lines
2.1 KiB
C++
/*
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* Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "api/audio_codecs/builtin_audio_decoder_factory.h"
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#include <memory>
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#include <vector>
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#include "api/audio_codecs/L16/audio_decoder_L16.h"
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#include "api/audio_codecs/audio_decoder_factory_template.h"
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#include "api/audio_codecs/g711/audio_decoder_g711.h"
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#include "api/audio_codecs/g722/audio_decoder_g722.h"
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#if WEBRTC_USE_BUILTIN_ILBC
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#include "api/audio_codecs/ilbc/audio_decoder_ilbc.h" // nogncheck
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#endif
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#include "api/audio_codecs/isac/audio_decoder_isac.h"
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#if WEBRTC_USE_BUILTIN_OPUS
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#include "api/audio_codecs/opus/audio_decoder_multi_channel_opus.h"
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#include "api/audio_codecs/opus/audio_decoder_opus.h" // nogncheck
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#endif
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namespace webrtc {
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namespace {
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// Modify an audio decoder to not advertise support for anything.
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template <typename T>
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struct NotAdvertised {
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using Config = typename T::Config;
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static absl::optional<Config> SdpToConfig(
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const SdpAudioFormat& audio_format) {
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return T::SdpToConfig(audio_format);
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}
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static void AppendSupportedDecoders(std::vector<AudioCodecSpec>* specs) {
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// Don't advertise support for anything.
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}
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static std::unique_ptr<AudioDecoder> MakeAudioDecoder(
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const Config& config,
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absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt) {
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return T::MakeAudioDecoder(config, codec_pair_id);
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}
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};
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} // namespace
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rtc::scoped_refptr<AudioDecoderFactory> CreateBuiltinAudioDecoderFactory() {
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return CreateAudioDecoderFactory<
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#if WEBRTC_USE_BUILTIN_OPUS
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AudioDecoderOpus, NotAdvertised<AudioDecoderMultiChannelOpus>,
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#endif
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AudioDecoderIsac, AudioDecoderG722,
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#if WEBRTC_USE_BUILTIN_ILBC
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AudioDecoderIlbc,
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#endif
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AudioDecoderG711, NotAdvertised<AudioDecoderL16>>();
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}
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} // namespace webrtc
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