92ea95e34a
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC from src/webrtc to src/ (in order to preserve an healthy git history). This CL takes care of fixing header guards, #include paths, etc... NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578 Reviewed-on: https://webrtc-review.googlesource.com/1561 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19846}
44 lines
1.5 KiB
C++
44 lines
1.5 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 "common_types.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/ptr_util.h"
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#include "rtc_base/safe_conversions.h"
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namespace webrtc {
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rtc::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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return STR_CASE_CMP(format.name.c_str(), "L16") == 0 && config.IsOk()
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? rtc::Optional<Config>(config)
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: rtc::Optional<Config>();
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}
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void AudioDecoderL16::AppendSupportedDecoders(
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std::vector<AudioCodecSpec>* specs) {
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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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return config.IsOk() ? rtc::MakeUnique<AudioDecoderPcm16B>(
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config.sample_rate_hz, config.num_channels)
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: nullptr;
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}
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} // namespace webrtc
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