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/*
* 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_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,
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>>
video_args(
(std::unique_ptr<WebRtcVideoEncoderFactory>(video_encoder_factory)),
(std::unique_ptr<WebRtcVideoDecoderFactory>(video_decoder_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 CreateWebRtcMediaEngine(adm, audio_encoder_factory,
audio_decoder_factory, video_encoder_factory,
video_decoder_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 CreateWebRtcMediaEngine(
adm, audio_encoder_factory, audio_decoder_factory, video_encoder_factory,
video_decoder_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) {
#ifdef HAVE_WEBRTC_VIDEO
typedef WebRtcVideoEngine VideoEngine;
std::tuple<std::unique_ptr<webrtc::VideoEncoderFactory>,
std::unique_ptr<webrtc::VideoDecoderFactory>>
video_args(std::move(video_encoder_factory),
std::move(video_decoder_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[14] = {false};
for (const auto& extension : extensions) {
if (extension.id <= 0 || extension.id >= 15) {
RTC_LOG(LS_ERROR) << "Bad RTP extension ID: " << extension.ToString();
return false;
}
if (id_used[extension.id - 1]) {
RTC_LOG(LS_ERROR) << "Duplicate RTP extension ID: "
<< extension.ToString();
return false;
}
id_used[extension.id - 1] = 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
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