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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/webrtc_media_engine.h"
#include <algorithm>
#include <map>
#include <memory>
#include <string>
#include <utility>
#include "absl/algorithm/container.h"
#include "absl/strings/match.h"
#include "api/transport/field_trial_based_config.h"
#include "media/base/media_constants.h"
#include "media/engine/webrtc_voice_engine.h"
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
#ifdef HAVE_WEBRTC_VIDEO
#include "media/engine/webrtc_video_engine.h"
#else
#include "media/engine/null_webrtc_video_engine.h"
#endif
namespace cricket {
std::unique_ptr<MediaEngineInterface> CreateMediaEngine(
MediaEngineDependencies dependencies) {
// TODO(sprang): Make populating `dependencies.trials` mandatory and remove
// these fallbacks.
std::unique_ptr<webrtc::FieldTrialsView> fallback_trials(
dependencies.trials ? nullptr : new webrtc::FieldTrialBasedConfig());
const webrtc::FieldTrialsView& trials =
dependencies.trials ? *dependencies.trials : *fallback_trials;
auto audio_engine = std::make_unique<WebRtcVoiceEngine>(
dependencies.task_queue_factory, dependencies.adm.get(),
std::move(dependencies.audio_encoder_factory),
std::move(dependencies.audio_decoder_factory),
std::move(dependencies.audio_mixer),
std::move(dependencies.audio_processing),
dependencies.audio_frame_processor,
std::move(dependencies.owned_audio_frame_processor), trials);
#ifdef HAVE_WEBRTC_VIDEO
auto video_engine = std::make_unique<WebRtcVideoEngine>(
std::move(dependencies.video_encoder_factory),
std::move(dependencies.video_decoder_factory), trials);
#else
auto video_engine = std::make_unique<NullWebRtcVideoEngine>();
#endif
return std::make_unique<CompositeMediaEngine>(std::move(fallback_trials),
std::move(audio_engine),
std::move(video_engine));
}
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 = absl::c_find_if(
*extensions,
[uri](const webrtc::RtpExtension& rhs) { return rhs.uri == uri; });
if (it != extensions->end()) {
if (found) {
extensions->erase(it);
}
found = true;
}
}
}
} // namespace
bool ValidateRtpExtensions(
rtc::ArrayView<const webrtc::RtpExtension> extensions,
rtc::ArrayView<const webrtc::RtpExtension> old_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;
}
// Validate the extension list against the already negotiated extensions.
// Re-registering is OK, re-mapping (either same URL at new ID or same
// ID used with new URL) is an illegal remap.
// This is required in order to avoid a crash when registering an
// extension. A better structure would use the registered extensions
// in the RTPSender. This requires spinning through:
//
// WebRtcVoiceMediaChannel::::WebRtcAudioSendStream::stream_ (pointer)
// AudioSendStream::rtp_rtcp_module_ (pointer)
// ModuleRtpRtcpImpl2::rtp_sender_ (pointer)
// RtpSenderContext::packet_generator (struct member)
// RTPSender::rtp_header_extension_map_ (class member)
//
// Getting at this seems like a hard slog.
if (!old_extensions.empty()) {
absl::string_view urimap[1 + webrtc::RtpExtension::kMaxId];
std::map<absl::string_view, int> idmap;
for (const auto& old_extension : old_extensions) {
urimap[old_extension.id] = old_extension.uri;
idmap[old_extension.uri] = old_extension.id;
}
for (const auto& extension : extensions) {
if (!urimap[extension.id].empty() &&
urimap[extension.id] != extension.uri) {
RTC_LOG(LS_ERROR) << "Extension negotiation failure: " << extension.id
<< " was mapped to " << urimap[extension.id]
<< " but is proposed changed to " << extension.uri;
return false;
}
const auto& it = idmap.find(extension.uri);
if (it != idmap.end() && it->second != extension.id) {
RTC_LOG(LS_ERROR) << "Extension negotation failure: " << extension.uri
<< " was identified by " << it->second
<< " but is proposed changed to " << extension.id;
return false;
}
}
}
return true;
}
std::vector<webrtc::RtpExtension> FilterRtpExtensions(
const std::vector<webrtc::RtpExtension>& extensions,
bool (*supported)(absl::string_view),
bool filter_redundant_extensions,
const webrtc::FieldTrialsView& trials) {
// Don't check against old parameters; this should have been done earlier.
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).
absl::c_sort(result, [](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 lists.
if (absl::StartsWith(trials.Lookup("WebRTC-FilterAbsSendTimeExtension"),
"Enabled")) {
static const char* const kBweExtensionPriorities[] = {
webrtc::RtpExtension::kTransportSequenceNumberUri,
webrtc::RtpExtension::kAbsSendTimeUri,
webrtc::RtpExtension::kTimestampOffsetUri};
DiscardRedundantExtensions(&result, kBweExtensionPriorities);
} else {
static const char* const kBweExtensionPriorities[] = {
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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