mirror of
https://github.com/mollyim/webrtc.git
synced 2025-05-20 00:57:49 +01:00
28deb90728
This is a reland of 3409cfa378
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}
117 lines
3.5 KiB
C++
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 "rtc_base/checks.h"
|
|
#include "rtc_base/logging.h"
|
|
#include "rtc_base/ptr_util.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 rtc::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 rtc::MakeUnique<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 rtc::MakeUnique<H264DecoderImpl>();
|
|
#else
|
|
RTC_NOTREACHED();
|
|
return nullptr;
|
|
#endif
|
|
}
|
|
|
|
bool H264Decoder::IsSupported() {
|
|
return IsH264CodecSupported();
|
|
}
|
|
|
|
} // namespace webrtc
|