aefed55c25
First batch of applying iwyu to the repo. Done with: > ./tools_webrtc/iwyu/apply-iwyu api > git add api/[a-s]* > python3 gn_autodeps.py ~/local/webrtc/src out/Default Last step is a custom script I wrote to automatically apply new required dependencies for target in gn, which saved tons of time manually going over the files and fixing. If this is something that interest others, I can submit it as well. Bug: webrtc:42226242 Change-Id: Id109e77f50835827495bc4512880c4ec9ae175f6 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/343680 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Dor Hen <dorhen@meta.com> Cr-Commit-Position: refs/heads/main@{#42512}
75 lines
2.3 KiB
C++
75 lines
2.3 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/g711/audio_decoder_g711.h"
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#include <initializer_list>
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#include <memory>
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#include <vector>
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#include "absl/strings/match.h"
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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 "modules/audio_coding/codecs/g711/audio_decoder_pcm.h"
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#include "rtc_base/checks.h"
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#include "rtc_base/numerics/safe_conversions.h"
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namespace webrtc {
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absl::optional<AudioDecoderG711::Config> AudioDecoderG711::SdpToConfig(
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const SdpAudioFormat& format) {
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const bool is_pcmu = absl::EqualsIgnoreCase(format.name, "PCMU");
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const bool is_pcma = absl::EqualsIgnoreCase(format.name, "PCMA");
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if (format.clockrate_hz == 8000 && format.num_channels >= 1 &&
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(is_pcmu || is_pcma)) {
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Config config;
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config.type = is_pcmu ? Config::Type::kPcmU : Config::Type::kPcmA;
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config.num_channels = rtc::dchecked_cast<int>(format.num_channels);
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if (!config.IsOk()) {
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RTC_DCHECK_NOTREACHED();
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return absl::nullopt;
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}
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return config;
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} else {
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return absl::nullopt;
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}
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}
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void AudioDecoderG711::AppendSupportedDecoders(
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std::vector<AudioCodecSpec>* specs) {
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for (const char* type : {"PCMU", "PCMA"}) {
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specs->push_back({{type, 8000, 1}, {8000, 1, 64000}});
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}
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}
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std::unique_ptr<AudioDecoder> AudioDecoderG711::MakeAudioDecoder(
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const Config& config,
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absl::optional<AudioCodecPairId> /*codec_pair_id*/,
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const FieldTrialsView* field_trials) {
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if (!config.IsOk()) {
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RTC_DCHECK_NOTREACHED();
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return nullptr;
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}
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switch (config.type) {
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case Config::Type::kPcmU:
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return std::make_unique<AudioDecoderPcmU>(config.num_channels);
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case Config::Type::kPcmA:
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return std::make_unique<AudioDecoderPcmA>(config.num_channels);
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default:
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RTC_DCHECK_NOTREACHED();
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return nullptr;
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}
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}
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} // namespace webrtc
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