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bb547203bf
In order to eliminate the WebRTC Subtree mirror in Chromium, WebRTC is moving the content of the src/webrtc directory up to the src/ directory. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38 Reviewed-on: https://webrtc-review.googlesource.com/1560 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19845}
59 lines
2.1 KiB
C++
59 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 "webrtc/api/audio_codecs/L16/audio_encoder_L16.h"
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#include "webrtc/common_types.h"
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#include "webrtc/modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.h"
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#include "webrtc/modules/audio_coding/codecs/pcm16b/pcm16b_common.h"
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#include "webrtc/rtc_base/ptr_util.h"
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#include "webrtc/rtc_base/safe_conversions.h"
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namespace webrtc {
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rtc::Optional<AudioEncoderL16::Config> AudioEncoderL16::SdpToConfig(
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const SdpAudioFormat& format) {
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if (!rtc::IsValueInRangeForNumericType<int>(format.num_channels)) {
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return rtc::Optional<Config>();
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}
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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::dchecked_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 AudioEncoderL16::AppendSupportedEncoders(
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std::vector<AudioCodecSpec>* specs) {
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Pcm16BAppendSupportedCodecSpecs(specs);
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}
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AudioCodecInfo AudioEncoderL16::QueryAudioEncoder(
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const AudioEncoderL16::Config& config) {
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RTC_DCHECK(config.IsOk());
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return {config.sample_rate_hz,
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rtc::dchecked_cast<size_t>(config.num_channels),
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config.sample_rate_hz * config.num_channels * 16};
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}
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std::unique_ptr<AudioEncoder> AudioEncoderL16::MakeAudioEncoder(
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const AudioEncoderL16::Config& config,
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int payload_type) {
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RTC_DCHECK(config.IsOk());
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AudioEncoderPcm16B::Config c;
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c.sample_rate_hz = config.sample_rate_hz;
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c.num_channels = config.num_channels;
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c.frame_size_ms = config.frame_size_ms;
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c.payload_type = payload_type;
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return rtc::MakeUnique<AudioEncoderPcm16B>(c);
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}
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} // namespace webrtc
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