Adding upstream version 110.0.1.upstream/110.0.1upstream

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>

1 files changed, 216 insertions, 0 deletions

diff --git a/third_party/libwebrtc/media/engine/webrtc_media_engine.cc b/third_party/libwebrtc/media/engine/webrtc_media_engine.cc
new file mode 100644
index 0000000000..d0d6de2034
--- /dev/null
+++ b/third_party/libwebrtc/media/engine/webrtc_media_engine.cc
@@ -0,0 +1,216 @@
+/*
+ *  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 <map>
+#include <memory>
+#include <utility>
+
+#include "absl/algorithm/container.h"
+#include "absl/strings/match.h"
+#include "media/engine/webrtc_voice_engine.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, 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