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/*
* 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"
Use std::make_unique instead of absl::make_unique. 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}
2019-09-17 17:06:18 +02:00
#include <memory>
#include <string>
#include "absl/container/inlined_vector.h"
#include "absl/types/optional.h"
#include "api/video_codecs/sdp_video_format.h"
#include "media/base/media_constants.h"
#include "rtc_base/trace_event.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 "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
}
constexpr ScalabilityMode kSupportedScalabilityModes[] = {
ScalabilityMode::kL1T1, ScalabilityMode::kL1T2, ScalabilityMode::kL1T3};
Reland "Handle scalability mode in QueryCodecSupport" This reverts commit 74281bed5350af9c15f83e0b1aec5c5921dbf76f. Reason for revert: Fixed unit test by removing VP9 profile 2 from encoder factory unit test since this is platform dependent. Original change's description: > Revert "Handle scalability mode in QueryCodecSupport" > > This reverts commit 715a14811883a642e3acca21fb6017f8a128c0a5. > > Reason for revert: Speculative revert. Breaks upstream project http://b/200009579 > > Original change's description: > > Handle scalability mode in QueryCodecSupport > > > > All valid scalability modes should be supported by the builtin > > software decoder/encoder. > > > > Bug: chromium:1187565 > > Change-Id: If66105d210d5055019f35dae2f80a18ad4a70cdd > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222642 > > Commit-Queue: Johannes Kron <kron@webrtc.org> > > Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> > > Reviewed-by: Erik Språng <sprang@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#34998} > > TBR=danilchap@webrtc.org,sprang@webrtc.org,kron@webrtc.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com > > Change-Id: Ibf40d523c50791d73e2afdc3917892b859d2bcb6 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: chromium:1187565 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/232020 > Reviewed-by: Andrey Logvin <landrey@webrtc.org> > Commit-Queue: Andrey Logvin <landrey@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#35001} Bug: chromium:1187565 Change-Id: I598a2a530b8fea22997bbb5910eb3b864d1e28a2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/232021 Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Johannes Kron <kron@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35003}
2021-09-15 10:56:04 +00:00
} // namespace
SdpVideoFormat CreateH264Format(H264Profile profile,
H264Level level,
const std::string& packetization_mode,
bool add_scalability_modes) {
const absl::optional<std::string> profile_string =
H264ProfileLevelIdToString(H264ProfileLevelId(profile, level));
RTC_CHECK(profile_string);
absl::InlinedVector<ScalabilityMode, kScalabilityModeCount> scalability_modes;
if (add_scalability_modes) {
for (const auto scalability_mode : kSupportedScalabilityModes) {
scalability_modes.push_back(scalability_mode);
}
}
Reland "Start supporting H264 packetization mode 0." This is a reland of 3409cfa378e75c0c08d900e0848147929249a62b Needed to change RtpVideoStreamReceiver to stop deregistering a payload type if two payload types refer to the same codec (which now happens, with the packetization mode 0/1 payload types). It's not clear why this was being done in the first place. Original change's description: > Start supporting H264 packetization mode 0. > > The work was already done to support it, but it wasn't being negotiated > in SDP. > > This means we'll now see 8 H264 payload types instead of 4; one for each > combination of BP/CBP profiles, packetization modes 0/1, and RTX/non-RTX. > This could be problematic in the future, since we're starting to run > out of dynamic payload types (using 25 of 32). > > Bug: chromium:600254 > Change-Id: Ief2340db77c796f12980445b547b87e939170fae > Reviewed-on: https://webrtc-review.googlesource.com/77264 > Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org> > Reviewed-by: Erik Språng <sprang@webrtc.org> > Reviewed-by: Magnus Jedvert <magjed@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#23372} Bug: chromium:600254 Change-Id: Ice1acc05acd1543d9b46e918de2bba0694d86259 Reviewed-on: https://webrtc-review.googlesource.com/78399 Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23494}
2018-05-30 14:56:50 -07:00
return SdpVideoFormat(
cricket::kH264CodecName,
{{cricket::kH264FmtpProfileLevelId, *profile_string},
{cricket::kH264FmtpLevelAsymmetryAllowed, "1"},
{cricket::kH264FmtpPacketizationMode, packetization_mode}},
scalability_modes);
}
void DisableRtcUseH264() {
#if defined(WEBRTC_USE_H264)
g_rtc_use_h264 = false;
#endif
}
std::vector<SdpVideoFormat> SupportedH264Codecs(bool add_scalability_modes) {
TRACE_EVENT0("webrtc", __func__);
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.
Reland "Start supporting H264 packetization mode 0." This is a reland of 3409cfa378e75c0c08d900e0848147929249a62b Needed to change RtpVideoStreamReceiver to stop deregistering a payload type if two payload types refer to the same codec (which now happens, with the packetization mode 0/1 payload types). It's not clear why this was being done in the first place. Original change's description: > Start supporting H264 packetization mode 0. > > The work was already done to support it, but it wasn't being negotiated > in SDP. > > This means we'll now see 8 H264 payload types instead of 4; one for each > combination of BP/CBP profiles, packetization modes 0/1, and RTX/non-RTX. > This could be problematic in the future, since we're starting to run > out of dynamic payload types (using 25 of 32). > > Bug: chromium:600254 > Change-Id: Ief2340db77c796f12980445b547b87e939170fae > Reviewed-on: https://webrtc-review.googlesource.com/77264 > Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org> > Reviewed-by: Erik Språng <sprang@webrtc.org> > Reviewed-by: Magnus Jedvert <magjed@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#23372} Bug: chromium:600254 Change-Id: Ice1acc05acd1543d9b46e918de2bba0694d86259 Reviewed-on: https://webrtc-review.googlesource.com/78399 Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23494}
2018-05-30 14:56:50 -07:00
//
// We support both packetization modes 0 (mandatory) and 1 (optional,
// preferred).
return {CreateH264Format(H264Profile::kProfileBaseline, H264Level::kLevel3_1,
"1", add_scalability_modes),
CreateH264Format(H264Profile::kProfileBaseline, H264Level::kLevel3_1,
"0", add_scalability_modes),
CreateH264Format(H264Profile::kProfileConstrainedBaseline,
H264Level::kLevel3_1, "1", add_scalability_modes),
CreateH264Format(H264Profile::kProfileConstrainedBaseline,
H264Level::kLevel3_1, "0", add_scalability_modes),
CreateH264Format(H264Profile::kProfileMain, H264Level::kLevel3_1, "1",
add_scalability_modes),
CreateH264Format(H264Profile::kProfileMain, H264Level::kLevel3_1, "0",
add_scalability_modes)};
}
std::vector<SdpVideoFormat> SupportedH264DecoderCodecs() {
TRACE_EVENT0("webrtc", __func__);
if (!IsH264CodecSupported())
return std::vector<SdpVideoFormat>();
std::vector<SdpVideoFormat> supportedCodecs = SupportedH264Codecs();
// OpenH264 doesn't yet support High Predictive 4:4:4 encoding but it does
// support decoding.
supportedCodecs.push_back(CreateH264Format(
H264Profile::kProfilePredictiveHigh444, H264Level::kLevel3_1, "1"));
supportedCodecs.push_back(CreateH264Format(
H264Profile::kProfilePredictiveHigh444, H264Level::kLevel3_1, "0"));
return supportedCodecs;
}
Reland "Update internal SW codecs to return unique_ptrs" This reverts commit 34c8e6bce8af0c31f2b0b31d691a6a931fa3cb7b. Reason for revert: Fix Android compilation Original change's description: > Revert "Update internal SW codecs to return unique_ptrs" > > This reverts commit 4fe6adc06a8524ac25f85260bfe392eb31dae6b4. > > Reason for revert: Breaks android compile. > > Original change's description: > > Update internal SW codecs to return unique_ptrs > > > > TBR=stefan@webrtc.org > > > > Bug: webrtc:7925 > > Change-Id: I84239b071a2608d928f09b06809090eec5eafb14 > > Reviewed-on: https://webrtc-review.googlesource.com/21165 > > Commit-Queue: Magnus Jedvert <magjed@webrtc.org> > > Reviewed-by: Erik Språng <sprang@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#20650} > > TBR=magjed@webrtc.org,sprang@webrtc.org,stefan@webrtc.org > > Change-Id: If33c3a0ee0dfce63d105558a2897a472f0633306 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:7925 > Reviewed-on: https://webrtc-review.googlesource.com/22540 > Reviewed-by: Magnus Jedvert <magjed@webrtc.org> > Commit-Queue: Magnus Jedvert <magjed@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#20652} TBR=magjed@webrtc.org,sprang@webrtc.org,stefan@webrtc.org Change-Id: Ic8551af4687e927c9b605060155abdd5bc6208b2 Bug: webrtc:7925 Reviewed-on: https://webrtc-review.googlesource.com/22541 Commit-Queue: Magnus Jedvert <magjed@webrtc.org> Reviewed-by: Magnus Jedvert <magjed@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20655}
2017-11-13 14:10:02 +01:00
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.";
Use std::make_unique instead of absl::make_unique. 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}
2019-09-17 17:06:18 +02:00
return std::make_unique<H264EncoderImpl>(codec);
#else
RTC_DCHECK_NOTREACHED();
return nullptr;
#endif
}
bool H264Encoder::IsSupported() {
return IsH264CodecSupported();
}
bool H264Encoder::SupportsScalabilityMode(ScalabilityMode scalability_mode) {
Reland "Handle scalability mode in QueryCodecSupport" This reverts commit 74281bed5350af9c15f83e0b1aec5c5921dbf76f. Reason for revert: Fixed unit test by removing VP9 profile 2 from encoder factory unit test since this is platform dependent. Original change's description: > Revert "Handle scalability mode in QueryCodecSupport" > > This reverts commit 715a14811883a642e3acca21fb6017f8a128c0a5. > > Reason for revert: Speculative revert. Breaks upstream project http://b/200009579 > > Original change's description: > > Handle scalability mode in QueryCodecSupport > > > > All valid scalability modes should be supported by the builtin > > software decoder/encoder. > > > > Bug: chromium:1187565 > > Change-Id: If66105d210d5055019f35dae2f80a18ad4a70cdd > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222642 > > Commit-Queue: Johannes Kron <kron@webrtc.org> > > Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> > > Reviewed-by: Erik Språng <sprang@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#34998} > > TBR=danilchap@webrtc.org,sprang@webrtc.org,kron@webrtc.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com > > Change-Id: Ibf40d523c50791d73e2afdc3917892b859d2bcb6 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: chromium:1187565 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/232020 > Reviewed-by: Andrey Logvin <landrey@webrtc.org> > Commit-Queue: Andrey Logvin <landrey@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#35001} Bug: chromium:1187565 Change-Id: I598a2a530b8fea22997bbb5910eb3b864d1e28a2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/232021 Reviewed-by: Erik Språng <sprang@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Johannes Kron <kron@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35003}
2021-09-15 10:56:04 +00:00
for (const auto& entry : kSupportedScalabilityModes) {
if (entry == scalability_mode) {
return true;
}
}
return false;
}
Reland "Update internal SW codecs to return unique_ptrs" This reverts commit 34c8e6bce8af0c31f2b0b31d691a6a931fa3cb7b. Reason for revert: Fix Android compilation Original change's description: > Revert "Update internal SW codecs to return unique_ptrs" > > This reverts commit 4fe6adc06a8524ac25f85260bfe392eb31dae6b4. > > Reason for revert: Breaks android compile. > > Original change's description: > > Update internal SW codecs to return unique_ptrs > > > > TBR=stefan@webrtc.org > > > > Bug: webrtc:7925 > > Change-Id: I84239b071a2608d928f09b06809090eec5eafb14 > > Reviewed-on: https://webrtc-review.googlesource.com/21165 > > Commit-Queue: Magnus Jedvert <magjed@webrtc.org> > > Reviewed-by: Erik Språng <sprang@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#20650} > > TBR=magjed@webrtc.org,sprang@webrtc.org,stefan@webrtc.org > > Change-Id: If33c3a0ee0dfce63d105558a2897a472f0633306 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:7925 > Reviewed-on: https://webrtc-review.googlesource.com/22540 > Reviewed-by: Magnus Jedvert <magjed@webrtc.org> > Commit-Queue: Magnus Jedvert <magjed@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#20652} TBR=magjed@webrtc.org,sprang@webrtc.org,stefan@webrtc.org Change-Id: Ic8551af4687e927c9b605060155abdd5bc6208b2 Bug: webrtc:7925 Reviewed-on: https://webrtc-review.googlesource.com/22541 Commit-Queue: Magnus Jedvert <magjed@webrtc.org> Reviewed-by: Magnus Jedvert <magjed@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20655}
2017-11-13 14:10:02 +01:00
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.";
Use std::make_unique instead of absl::make_unique. 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}
2019-09-17 17:06:18 +02:00
return std::make_unique<H264DecoderImpl>();
#else
RTC_DCHECK_NOTREACHED();
return nullptr;
#endif
}
bool H264Decoder::IsSupported() {
return IsH264CodecSupported();
}
} // namespace webrtc