mirror of
https://github.com/mollyim/webrtc.git
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50 lines
1.6 KiB
C++
50 lines
1.6 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/L16/audio_decoder_L16.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/pcm16b/audio_decoder_pcm16b.h"
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#include "modules/audio_coding/codecs/pcm16b/pcm16b_common.h"
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#include "rtc_base/numerics/safe_conversions.h"
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namespace webrtc {
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absl::optional<AudioDecoderL16::Config> AudioDecoderL16::SdpToConfig(
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const SdpAudioFormat& format) {
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Config config;
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config.sample_rate_hz = format.clockrate_hz;
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config.num_channels = rtc::checked_cast<int>(format.num_channels);
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if (absl::EqualsIgnoreCase(format.name, "L16") && config.IsOk()) {
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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 AudioDecoderL16::AppendSupportedDecoders(
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std::vector<AudioCodecSpec>* specs) {
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// RingRTC change to unused audio codecs
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// Pcm16BAppendSupportedCodecSpecs(specs);
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}
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std::unique_ptr<AudioDecoder> AudioDecoderL16::MakeAudioDecoder(
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const Config& config,
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absl::optional<AudioCodecPairId> /*codec_pair_id*/,
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const WebRtcKeyValueConfig* field_trials) {
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if (!config.IsOk()) {
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return nullptr;
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}
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return std::make_unique<AudioDecoderPcm16B>(config.sample_rate_hz,
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config.num_channels);
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}
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} // namespace webrtc
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