webrtc/modules/video_coding/codecs/h264/h264.cc
Karl Wiberg 918f50c5d1 Use absl::make_unique and absl::WrapUnique directly
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}
2018-07-05 10:59:49 +00:00

117 lines
3.5 KiB
C++

/*
* Copyright (c) 2015 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 "modules/video_coding/codecs/h264/include/h264.h"
#include "api/video_codecs/sdp_video_format.h"
#include "media/base/h264_profile_level_id.h"
#if defined(WEBRTC_USE_H264)
#include "modules/video_coding/codecs/h264/h264_decoder_impl.h"
#include "modules/video_coding/codecs/h264/h264_encoder_impl.h"
#endif
#include "absl/memory/memory.h"
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
namespace webrtc {
namespace {
#if defined(WEBRTC_USE_H264)
bool g_rtc_use_h264 = true;
#endif
// If H.264 OpenH264/FFmpeg codec is supported.
bool IsH264CodecSupported() {
#if defined(WEBRTC_USE_H264)
return g_rtc_use_h264;
#else
return false;
#endif
}
SdpVideoFormat CreateH264Format(H264::Profile profile,
H264::Level level,
const std::string& packetization_mode) {
const absl::optional<std::string> profile_string =
H264::ProfileLevelIdToString(H264::ProfileLevelId(profile, level));
RTC_CHECK(profile_string);
return SdpVideoFormat(
cricket::kH264CodecName,
{{cricket::kH264FmtpProfileLevelId, *profile_string},
{cricket::kH264FmtpLevelAsymmetryAllowed, "1"},
{cricket::kH264FmtpPacketizationMode, packetization_mode}});
}
} // namespace
void DisableRtcUseH264() {
#if defined(WEBRTC_USE_H264)
g_rtc_use_h264 = false;
#endif
}
std::vector<SdpVideoFormat> SupportedH264Codecs() {
if (!IsH264CodecSupported())
return std::vector<SdpVideoFormat>();
// We only support encoding Constrained Baseline Profile (CBP), but the
// decoder supports more profiles. We can list all profiles here that are
// supported by the decoder and that are also supersets of CBP, i.e. the
// decoder for that profile is required to be able to decode CBP. This means
// we can encode and send CBP even though we negotiated a potentially
// higher profile. See the H264 spec for more information.
//
// We support both packetization modes 0 (mandatory) and 1 (optional,
// preferred).
return {
CreateH264Format(H264::kProfileBaseline, H264::kLevel3_1, "1"),
CreateH264Format(H264::kProfileBaseline, H264::kLevel3_1, "0"),
CreateH264Format(H264::kProfileConstrainedBaseline, H264::kLevel3_1, "1"),
CreateH264Format(H264::kProfileConstrainedBaseline, H264::kLevel3_1,
"0")};
}
std::unique_ptr<H264Encoder> H264Encoder::Create(
const cricket::VideoCodec& codec) {
RTC_DCHECK(H264Encoder::IsSupported());
#if defined(WEBRTC_USE_H264)
RTC_CHECK(g_rtc_use_h264);
RTC_LOG(LS_INFO) << "Creating H264EncoderImpl.";
return absl::make_unique<H264EncoderImpl>(codec);
#else
RTC_NOTREACHED();
return nullptr;
#endif
}
bool H264Encoder::IsSupported() {
return IsH264CodecSupported();
}
std::unique_ptr<H264Decoder> H264Decoder::Create() {
RTC_DCHECK(H264Decoder::IsSupported());
#if defined(WEBRTC_USE_H264)
RTC_CHECK(g_rtc_use_h264);
RTC_LOG(LS_INFO) << "Creating H264DecoderImpl.";
return absl::make_unique<H264DecoderImpl>();
#else
RTC_NOTREACHED();
return nullptr;
#endif
}
bool H264Decoder::IsSupported() {
return IsH264CodecSupported();
}
} // namespace webrtc