[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}
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WebRTC LUCI CQ
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/*
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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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#ifndef API_AUDIO_CODECS_ISAC_AUDIO_DECODER_ISAC_FLOAT_H_
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#define API_AUDIO_CODECS_ISAC_AUDIO_DECODER_ISAC_FLOAT_H_
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#include <memory>
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#include <vector>
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#include "absl/types/optional.h"
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#include "api/audio_codecs/audio_codec_pair_id.h"
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#include "api/audio_codecs/audio_decoder.h"
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#include "api/audio_codecs/audio_format.h"
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#include "api/field_trials_view.h"
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#include "rtc_base/system/rtc_export.h"
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namespace webrtc {
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// iSAC decoder API (floating-point implementation) for use as a template
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// parameter to CreateAudioDecoderFactory<...>().
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struct RTC_EXPORT AudioDecoderIsacFloat {
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struct Config {
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bool IsOk() const {
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return sample_rate_hz == 16000 || sample_rate_hz == 32000;
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}
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int sample_rate_hz = 16000;
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};
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static absl::optional<Config> SdpToConfig(const SdpAudioFormat& audio_format);
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static void AppendSupportedDecoders(std::vector<AudioCodecSpec>* specs);
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static std::unique_ptr<AudioDecoder> MakeAudioDecoder(
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Config config,
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absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt,
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const FieldTrialsView* field_trials = nullptr);
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};
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} // namespace webrtc
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#endif // API_AUDIO_CODECS_ISAC_AUDIO_DECODER_ISAC_FLOAT_H_
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