mirror of
https://github.com/mollyim/webrtc.git
synced 2025-05-14 22:30:40 +01:00
317a1f09ed
WebRTC is now using C++14 so there is no need to use the Abseil version of std::make_unique. This CL has been created with the following steps: git grep -l absl::make_unique | sort | uniq > /tmp/make_unique.txt git grep -l absl::WrapUnique | sort | uniq > /tmp/wrap_unique.txt git grep -l "#include <memory>" | sort | uniq > /tmp/memory.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/make_unique.txt /tmp/wrap_unique.txt | sort | \ uniq > /tmp/only_make_unique.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/only_make_unique.txt /tmp/memory.txt | \ xargs grep -l "absl/memory" > /tmp/add-memory.txt git grep -l "\babsl::make_unique\b" | \ xargs sed -i "s/\babsl::make_unique\b/std::make_unique/g" git checkout PRESUBMIT.py abseil-in-webrtc.md cat /tmp/add-memory.txt | \ xargs sed -i \ 's/#include "absl\/memory\/memory.h"/#include <memory>/g' git cl format # Manual fix order of the new inserted #include <memory> cat /tmp/only_make_unique | xargs grep -l "#include <memory>" | \ xargs sed -i '/#include "absl\/memory\/memory.h"/d' git ls-files | grep BUILD.gn | \ xargs sed -i '/\/\/third_party\/abseil-cpp\/absl\/memory/d' python tools_webrtc/gn_check_autofix.py \ -m tryserver.webrtc -b linux_rel # Repead the gn_check_autofix step for other platforms git ls-files | grep BUILD.gn | \ xargs sed -i 's/absl\/memory:memory/absl\/memory/g' git cl format Bug: webrtc:10945 Change-Id: I3fe28ea80f4dd3ba3cf28effd151d5e1f19aff89 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153221 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29209}
67 lines
2.3 KiB
C++
67 lines
2.3 KiB
C++
/*
|
|
* Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
|
|
*
|
|
* Use of this source code is governed by a BSD-style license
|
|
* that can be found in the LICENSE file in the root of the source
|
|
* tree. An additional intellectual property rights grant can be found
|
|
* in the file PATENTS. All contributing project authors may
|
|
* be found in the AUTHORS file in the root of the source tree.
|
|
*/
|
|
|
|
#include "api/audio_codecs/g722/audio_encoder_g722.h"
|
|
|
|
#include <memory>
|
|
#include <vector>
|
|
|
|
#include "absl/strings/match.h"
|
|
#include "modules/audio_coding/codecs/g722/audio_encoder_g722.h"
|
|
#include "rtc_base/numerics/safe_conversions.h"
|
|
#include "rtc_base/numerics/safe_minmax.h"
|
|
#include "rtc_base/string_to_number.h"
|
|
|
|
namespace webrtc {
|
|
|
|
absl::optional<AudioEncoderG722Config> AudioEncoderG722::SdpToConfig(
|
|
const SdpAudioFormat& format) {
|
|
if (!absl::EqualsIgnoreCase(format.name, "g722") ||
|
|
format.clockrate_hz != 8000) {
|
|
return absl::nullopt;
|
|
}
|
|
|
|
AudioEncoderG722Config config;
|
|
config.num_channels = rtc::checked_cast<int>(format.num_channels);
|
|
auto ptime_iter = format.parameters.find("ptime");
|
|
if (ptime_iter != format.parameters.end()) {
|
|
auto ptime = rtc::StringToNumber<int>(ptime_iter->second);
|
|
if (ptime && *ptime > 0) {
|
|
const int whole_packets = *ptime / 10;
|
|
config.frame_size_ms = rtc::SafeClamp<int>(whole_packets * 10, 10, 60);
|
|
}
|
|
}
|
|
return config.IsOk() ? absl::optional<AudioEncoderG722Config>(config)
|
|
: absl::nullopt;
|
|
}
|
|
|
|
void AudioEncoderG722::AppendSupportedEncoders(
|
|
std::vector<AudioCodecSpec>* specs) {
|
|
const SdpAudioFormat fmt = {"G722", 8000, 1};
|
|
const AudioCodecInfo info = QueryAudioEncoder(*SdpToConfig(fmt));
|
|
specs->push_back({fmt, info});
|
|
}
|
|
|
|
AudioCodecInfo AudioEncoderG722::QueryAudioEncoder(
|
|
const AudioEncoderG722Config& config) {
|
|
RTC_DCHECK(config.IsOk());
|
|
return {16000, rtc::dchecked_cast<size_t>(config.num_channels),
|
|
64000 * config.num_channels};
|
|
}
|
|
|
|
std::unique_ptr<AudioEncoder> AudioEncoderG722::MakeAudioEncoder(
|
|
const AudioEncoderG722Config& config,
|
|
int payload_type,
|
|
absl::optional<AudioCodecPairId> /*codec_pair_id*/) {
|
|
RTC_DCHECK(config.IsOk());
|
|
return std::make_unique<AudioEncoderG722Impl>(config, payload_type);
|
|
}
|
|
|
|
} // namespace webrtc
|