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918f50c5d1
Instead of going through our wrappers in ptr_util.h.
This CL was generated by the following script:
git grep -l ptr_util | xargs perl -pi -e 's,#include "rtc_base/ptr_util.h",#include "absl/memory/memory.h",'
git grep -l MakeUnique | xargs perl -pi -e 's,\b(rtc::)?MakeUnique\b,absl::make_unique,g'
git grep -l WrapUnique | xargs perl -pi -e 's,\b(rtc::)?WrapUnique\b,absl::WrapUnique,g'
git checkout -- rtc_base/ptr_util{.h,_unittest.cc}
git cl format
Followed by manually adding dependencies on
//third_party/abseil-cpp/absl/memory until `gn check` stopped
complaining.
Bug: webrtc:9473
Change-Id: I89ccd363f070479b8c431eb2c3d404a46eaacc1c
Reviewed-on: https://webrtc-review.googlesource.com/86600
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23850}
68 lines
2.3 KiB
C++
68 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/g722/audio_encoder_g722.h"
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#include <memory>
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#include <vector>
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#include "absl/memory/memory.h"
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#include "common_types.h" // NOLINT(build/include)
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#include "modules/audio_coding/codecs/g722/audio_encoder_g722.h"
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#include "rtc_base/numerics/safe_conversions.h"
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#include "rtc_base/numerics/safe_minmax.h"
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#include "rtc_base/string_to_number.h"
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namespace webrtc {
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absl::optional<AudioEncoderG722Config> AudioEncoderG722::SdpToConfig(
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const SdpAudioFormat& format) {
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if (STR_CASE_CMP(format.name.c_str(), "g722") != 0 ||
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format.clockrate_hz != 8000) {
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return absl::nullopt;
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}
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AudioEncoderG722Config config;
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config.num_channels = rtc::checked_cast<int>(format.num_channels);
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auto ptime_iter = format.parameters.find("ptime");
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if (ptime_iter != format.parameters.end()) {
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auto ptime = rtc::StringToNumber<int>(ptime_iter->second);
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if (ptime && *ptime > 0) {
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const int whole_packets = *ptime / 10;
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config.frame_size_ms = rtc::SafeClamp<int>(whole_packets * 10, 10, 60);
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}
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}
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return config.IsOk() ? absl::optional<AudioEncoderG722Config>(config)
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: absl::nullopt;
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}
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void AudioEncoderG722::AppendSupportedEncoders(
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std::vector<AudioCodecSpec>* specs) {
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const SdpAudioFormat fmt = {"G722", 8000, 1};
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const AudioCodecInfo info = QueryAudioEncoder(*SdpToConfig(fmt));
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specs->push_back({fmt, info});
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}
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AudioCodecInfo AudioEncoderG722::QueryAudioEncoder(
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const AudioEncoderG722Config& config) {
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RTC_DCHECK(config.IsOk());
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return {16000, rtc::dchecked_cast<size_t>(config.num_channels),
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64000 * config.num_channels};
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}
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std::unique_ptr<AudioEncoder> AudioEncoderG722::MakeAudioEncoder(
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const AudioEncoderG722Config& config,
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int payload_type,
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absl::optional<AudioCodecPairId> /*codec_pair_id*/) {
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RTC_DCHECK(config.IsOk());
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return absl::make_unique<AudioEncoderG722Impl>(config, payload_type);
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}
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} // namespace webrtc
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