zqz/platform-external-webrtc
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity
Files
72467209a89dc58e4cc750141f21e6ea9f1de49d
platform-external-webrtc/media/engine/webrtcmediaengine.cc
Jiawei Ou c2ebe21ba9 Reland "Use the factory instead of using the builtin code path in VideoCodecInitializer" 
Compared the original CL: https://webrtc-review.googlesource.com/c/src/+/94782

This new CL added backward compatible functions to WebRtcMediaEngineFactory so that internal projects will not be broken.

Because of that, now we can revert all the changes to SDK and PeerConnection and do it in following CLs. This makes this CL cleaner.

One temporary disadvantage of this is the media engine now need to take a dependency onto builtin video bitrate factory, but practically it just moved code around and should not result in a large binary size change. We can remove this dependency later if needed.

Bug: webrtc:9513
Change-Id: I38708762ff365e4ca05974b99fac71edc739a756
Reviewed-on: https://webrtc-review.googlesource.com/c/109040
Commit-Queue: Jiawei Ou <ouj@fb.com>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25574}
2018-11-08 19:10:47 +00:00

291 lines
11 KiB
C++
Raw Blame History

/*
* Copyright (c) 2014 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 "media/engine/webrtcmediaengine.h"
#include <algorithm>
#include <memory>
#include <tuple>
#include <utility>
#include "api/video/builtin_video_bitrate_allocator_factory.h"
#include "api/video_codecs/video_decoder_factory.h"
#include "api/video_codecs/video_encoder_factory.h"
#include "media/engine/webrtcvoiceengine.h"
#ifdef HAVE_WEBRTC_VIDEO
#include "media/engine/webrtcvideoengine.h"
#else
#include "media/engine/nullwebrtcvideoengine.h"
#endif
namespace cricket {
#if defined(USE_BUILTIN_SW_CODECS)
namespace {
MediaEngineInterface* CreateWebRtcMediaEngine(
webrtc::AudioDeviceModule* adm,
const rtc::scoped_refptr<webrtc::AudioEncoderFactory>&
audio_encoder_factory,
const rtc::scoped_refptr<webrtc::AudioDecoderFactory>&
audio_decoder_factory,
WebRtcVideoEncoderFactory* video_encoder_factory,
WebRtcVideoDecoderFactory* video_decoder_factory,
std::unique_ptr<webrtc::VideoBitrateAllocatorFactory>
video_bitrate_allocator_factory,
rtc::scoped_refptr<webrtc::AudioMixer> audio_mixer,
rtc::scoped_refptr<webrtc::AudioProcessing> audio_processing) {
#ifdef HAVE_WEBRTC_VIDEO
typedef WebRtcVideoEngine VideoEngine;
std::tuple<std::unique_ptr<WebRtcVideoEncoderFactory>,
std::unique_ptr<WebRtcVideoDecoderFactory>,
std::unique_ptr<webrtc::VideoBitrateAllocatorFactory>>
video_args(
(std::unique_ptr<WebRtcVideoEncoderFactory>(video_encoder_factory)),
(std::unique_ptr<WebRtcVideoDecoderFactory>(video_decoder_factory)),
(std::move(video_bitrate_allocator_factory)));
#else
typedef NullWebRtcVideoEngine VideoEngine;
std::tuple<> video_args;
#endif
return new CompositeMediaEngine<WebRtcVoiceEngine, VideoEngine>(
std::forward_as_tuple(adm, audio_encoder_factory, audio_decoder_factory,
audio_mixer, audio_processing),
std::move(video_args));
}
} // namespace
MediaEngineInterface* WebRtcMediaEngineFactory::Create(
webrtc::AudioDeviceModule* adm,
const rtc::scoped_refptr<webrtc::AudioEncoderFactory>&
audio_encoder_factory,
const rtc::scoped_refptr<webrtc::AudioDecoderFactory>&
audio_decoder_factory,
WebRtcVideoEncoderFactory* video_encoder_factory,
WebRtcVideoDecoderFactory* video_decoder_factory) {
return WebRtcMediaEngineFactory::Create(
adm, audio_encoder_factory, audio_decoder_factory, video_encoder_factory,
video_decoder_factory,
webrtc::CreateBuiltinVideoBitrateAllocatorFactory());
}
MediaEngineInterface* WebRtcMediaEngineFactory::Create(
webrtc::AudioDeviceModule* adm,
const rtc::scoped_refptr<webrtc::AudioEncoderFactory>&
audio_encoder_factory,
const rtc::scoped_refptr<webrtc::AudioDecoderFactory>&
audio_decoder_factory,
WebRtcVideoEncoderFactory* video_encoder_factory,
WebRtcVideoDecoderFactory* video_decoder_factory,
std::unique_ptr<webrtc::VideoBitrateAllocatorFactory>
video_bitrate_allocator_factory) {
return CreateWebRtcMediaEngine(
adm, audio_encoder_factory, audio_decoder_factory, video_encoder_factory,
video_decoder_factory, std::move(video_bitrate_allocator_factory),
nullptr, webrtc::AudioProcessingBuilder().Create());
}
MediaEngineInterface* WebRtcMediaEngineFactory::Create(
webrtc::AudioDeviceModule* adm,
const rtc::scoped_refptr<webrtc::AudioEncoderFactory>&
audio_encoder_factory,
const rtc::scoped_refptr<webrtc::AudioDecoderFactory>&
audio_decoder_factory,
WebRtcVideoEncoderFactory* video_encoder_factory,
WebRtcVideoDecoderFactory* video_decoder_factory,
rtc::scoped_refptr<webrtc::AudioMixer> audio_mixer,
rtc::scoped_refptr<webrtc::AudioProcessing> audio_processing) {
return WebRtcMediaEngineFactory::Create(
adm, audio_encoder_factory, audio_decoder_factory, video_encoder_factory,
video_decoder_factory,
webrtc::CreateBuiltinVideoBitrateAllocatorFactory(), audio_mixer,
audio_processing);
}
MediaEngineInterface* WebRtcMediaEngineFactory::Create(
webrtc::AudioDeviceModule* adm,
const rtc::scoped_refptr<webrtc::AudioEncoderFactory>&
audio_encoder_factory,
const rtc::scoped_refptr<webrtc::AudioDecoderFactory>&
audio_decoder_factory,
WebRtcVideoEncoderFactory* video_encoder_factory,
WebRtcVideoDecoderFactory* video_decoder_factory,
std::unique_ptr<webrtc::VideoBitrateAllocatorFactory>
video_bitrate_allocator_factory,
rtc::scoped_refptr<webrtc::AudioMixer> audio_mixer,
rtc::scoped_refptr<webrtc::AudioProcessing> audio_processing) {
return CreateWebRtcMediaEngine(
adm, audio_encoder_factory, audio_decoder_factory, video_encoder_factory,
video_decoder_factory, std::move(video_bitrate_allocator_factory),
audio_mixer, audio_processing);
}
#endif
std::unique_ptr<MediaEngineInterface> WebRtcMediaEngineFactory::Create(
rtc::scoped_refptr<webrtc::AudioDeviceModule> adm,
rtc::scoped_refptr<webrtc::AudioEncoderFactory> audio_encoder_factory,
rtc::scoped_refptr<webrtc::AudioDecoderFactory> audio_decoder_factory,
std::unique_ptr<webrtc::VideoEncoderFactory> video_encoder_factory,
std::unique_ptr<webrtc::VideoDecoderFactory> video_decoder_factory,
rtc::scoped_refptr<webrtc::AudioMixer> audio_mixer,
rtc::scoped_refptr<webrtc::AudioProcessing> audio_processing) {
return WebRtcMediaEngineFactory::Create(
adm, audio_encoder_factory, audio_decoder_factory,
std::move(video_encoder_factory), std::move(video_decoder_factory),
webrtc::CreateBuiltinVideoBitrateAllocatorFactory(), audio_mixer,
audio_processing);
}
std::unique_ptr<MediaEngineInterface> WebRtcMediaEngineFactory::Create(
rtc::scoped_refptr<webrtc::AudioDeviceModule> adm,
rtc::scoped_refptr<webrtc::AudioEncoderFactory> audio_encoder_factory,
rtc::scoped_refptr<webrtc::AudioDecoderFactory> audio_decoder_factory,
std::unique_ptr<webrtc::VideoEncoderFactory> video_encoder_factory,
std::unique_ptr<webrtc::VideoDecoderFactory> video_decoder_factory,
std::unique_ptr<webrtc::VideoBitrateAllocatorFactory>
video_bitrate_allocator_factory,
rtc::scoped_refptr<webrtc::AudioMixer> audio_mixer,
rtc::scoped_refptr<webrtc::AudioProcessing> audio_processing) {
#ifdef HAVE_WEBRTC_VIDEO
typedef WebRtcVideoEngine VideoEngine;
std::tuple<std::unique_ptr<webrtc::VideoEncoderFactory>,
std::unique_ptr<webrtc::VideoDecoderFactory>,
std::unique_ptr<webrtc::VideoBitrateAllocatorFactory>>
video_args(std::move(video_encoder_factory),
std::move(video_decoder_factory),
std::move(video_bitrate_allocator_factory));
#else
typedef NullWebRtcVideoEngine VideoEngine;
std::tuple<> video_args;
#endif
return std::unique_ptr<MediaEngineInterface>(
new CompositeMediaEngine<WebRtcVoiceEngine, VideoEngine>(
std::forward_as_tuple(adm, audio_encoder_factory,
audio_decoder_factory, audio_mixer,
audio_processing),
std::move(video_args)));
}
namespace {
// Remove mutually exclusive extensions with lower priority.
void DiscardRedundantExtensions(
std::vector<webrtc::RtpExtension>* extensions,
rtc::ArrayView<const char* const> extensions_decreasing_prio) {
RTC_DCHECK(extensions);
bool found = false;
for (const char* uri : extensions_decreasing_prio) {
auto it = std::find_if(
extensions->begin(), extensions->end(),
[uri](const webrtc::RtpExtension& rhs) { return rhs.uri == uri; });
if (it != extensions->end()) {
if (found) {
extensions->erase(it);
}
found = true;
}
}
}
} // namespace
bool ValidateRtpExtensions(
const std::vector<webrtc::RtpExtension>& extensions) {
bool id_used[1 + webrtc::RtpExtension::kMaxId] = {false};
for (const auto& extension : extensions) {
if (extension.id < webrtc::RtpExtension::kMinId ||
extension.id > webrtc::RtpExtension::kMaxId) {
RTC_LOG(LS_ERROR) << "Bad RTP extension ID: " << extension.ToString();
return false;
}
if (id_used[extension.id]) {
RTC_LOG(LS_ERROR) << "Duplicate RTP extension ID: "
<< extension.ToString();
return false;
}
id_used[extension.id] = true;
}
return true;
}
std::vector<webrtc::RtpExtension> FilterRtpExtensions(
const std::vector<webrtc::RtpExtension>& extensions,
bool (*supported)(const std::string&),
bool filter_redundant_extensions) {
RTC_DCHECK(ValidateRtpExtensions(extensions));
RTC_DCHECK(supported);
std::vector<webrtc::RtpExtension> result;
// Ignore any extensions that we don't recognize.
for (const auto& extension : extensions) {
if (supported(extension.uri)) {
result.push_back(extension);
} else {
RTC_LOG(LS_WARNING) << "Unsupported RTP extension: "
<< extension.ToString();
}
}
// Sort by name, ascending (prioritise encryption), so that we don't reset
// extensions if they were specified in a different order (also allows us
// to use std::unique below).
std::sort(
result.begin(), result.end(),
[](const webrtc::RtpExtension& rhs, const webrtc::RtpExtension& lhs) {
return rhs.encrypt == lhs.encrypt ? rhs.uri < lhs.uri
: rhs.encrypt > lhs.encrypt;
});
// Remove unnecessary extensions (used on send side).
if (filter_redundant_extensions) {
auto it = std::unique(
result.begin(), result.end(),
[](const webrtc::RtpExtension& rhs, const webrtc::RtpExtension& lhs) {
return rhs.uri == lhs.uri && rhs.encrypt == lhs.encrypt;
});
result.erase(it, result.end());
// Keep just the highest priority extension of any in the following list.
static const char* const kBweExtensionPriorities[] = {
webrtc::RtpExtension::kTransportSequenceNumberUri,
webrtc::RtpExtension::kAbsSendTimeUri,
webrtc::RtpExtension::kTimestampOffsetUri};
DiscardRedundantExtensions(&result, kBweExtensionPriorities);
}
return result;
}
webrtc::BitrateConstraints GetBitrateConfigForCodec(const Codec& codec) {
webrtc::BitrateConstraints config;
int bitrate_kbps = 0;
if (codec.GetParam(kCodecParamMinBitrate, &bitrate_kbps) &&
bitrate_kbps > 0) {
config.min_bitrate_bps = bitrate_kbps * 1000;
} else {
config.min_bitrate_bps = 0;
}
if (codec.GetParam(kCodecParamStartBitrate, &bitrate_kbps) &&
bitrate_kbps > 0) {
config.start_bitrate_bps = bitrate_kbps * 1000;
} else {
// Do not reconfigure start bitrate unless it's specified and positive.
config.start_bitrate_bps = -1;
}
if (codec.GetParam(kCodecParamMaxBitrate, &bitrate_kbps) &&
bitrate_kbps > 0) {
config.max_bitrate_bps = bitrate_kbps * 1000;
} else {
config.max_bitrate_bps = -1;
}
return config;
}
} // namespace cricket
Reference in New Issue View Git Blame Copy Permalink