mirror of
https://github.com/mollyim/webrtc.git
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44 lines
1.4 KiB
C++
44 lines
1.4 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/isac/audio_decoder_isac_fix.h"
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#include <memory>
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#include "absl/strings/match.h"
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#include "modules/audio_coding/codecs/isac/fix/include/audio_decoder_isacfix.h"
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namespace webrtc {
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absl::optional<AudioDecoderIsacFix::Config> AudioDecoderIsacFix::SdpToConfig(
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const SdpAudioFormat& format) {
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if (absl::EqualsIgnoreCase(format.name, "ISAC") &&
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format.clockrate_hz == 16000 && format.num_channels == 1) {
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return Config();
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}
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return absl::nullopt;
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}
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void AudioDecoderIsacFix::AppendSupportedDecoders(
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std::vector<AudioCodecSpec>* specs) {
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// RingRTC change to unused audio codecs
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// specs->push_back({{"ISAC", 16000, 1}, {16000, 1, 32000, 10000, 32000}});
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}
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std::unique_ptr<AudioDecoder> AudioDecoderIsacFix::MakeAudioDecoder(
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Config config,
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absl::optional<AudioCodecPairId> /*codec_pair_id*/,
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const WebRtcKeyValueConfig* field_trials) {
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AudioDecoderIsacFixImpl::Config c;
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c.sample_rate_hz = 16000;
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return std::make_unique<AudioDecoderIsacFixImpl>(c);
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}
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} // namespace webrtc
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