Move talk/media to webrtc/media

talk/media/webrtc/webrtcmediaengine.cc

-/*
- * libjingle
- * Copyright 2014 Google Inc.
- *
- * Redistribution and use in source and binary forms, with or without
- * modification, are permitted provided that the following conditions are met:
- *
- *  1. Redistributions of source code must retain the above copyright notice,
- *     this list of conditions and the following disclaimer.
- *  2. Redistributions in binary form must reproduce the above copyright notice,
- *     this list of conditions and the following disclaimer in the documentation
- *     and/or other materials provided with the distribution.
- *  3. The name of the author may not be used to endorse or promote products
- *     derived from this software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
- * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
- * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
- * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
- * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
- * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
- * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
- * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
- * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
- * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
- */
-
-#include "talk/media/webrtc/webrtcmediaengine.h"
-
-#include <algorithm>
-
-#include "talk/media/webrtc/webrtcvideoengine2.h"
-#include "talk/media/webrtc/webrtcvoiceengine.h"
-
-namespace cricket {
-
-class WebRtcMediaEngine2
-    : public CompositeMediaEngine<WebRtcVoiceEngine, WebRtcVideoEngine2> {
- public:
-  WebRtcMediaEngine2(webrtc::AudioDeviceModule* adm,
-                     WebRtcVideoEncoderFactory* encoder_factory,
-                     WebRtcVideoDecoderFactory* decoder_factory) {
-    voice_.SetAudioDeviceModule(adm);
-    video_.SetExternalDecoderFactory(decoder_factory);
-    video_.SetExternalEncoderFactory(encoder_factory);
-  }
-};
-
-}  // namespace cricket
-
-cricket::MediaEngineInterface* CreateWebRtcMediaEngine(
-    webrtc::AudioDeviceModule* adm,
-    cricket::WebRtcVideoEncoderFactory* encoder_factory,
-    cricket::WebRtcVideoDecoderFactory* decoder_factory) {
-  return new cricket::WebRtcMediaEngine2(adm, encoder_factory,
-                                         decoder_factory);
-}
-
-void DestroyWebRtcMediaEngine(cricket::MediaEngineInterface* media_engine) {
-  delete media_engine;
-}
-
-namespace cricket {
-
-// Used by PeerConnectionFactory to create a media engine passed into
-// ChannelManager.
-MediaEngineInterface* WebRtcMediaEngineFactory::Create(
-    webrtc::AudioDeviceModule* adm,
-    WebRtcVideoEncoderFactory* encoder_factory,
-    WebRtcVideoDecoderFactory* decoder_factory) {
-  return CreateWebRtcMediaEngine(adm, encoder_factory, decoder_factory);
-}
-
-namespace {
-// Remove mutually exclusive extensions with lower priority.
-void DiscardRedundantExtensions(
-    std::vector<webrtc::RtpExtension>* extensions,
-    rtc::ArrayView<const char*> extensions_decreasing_prio) {
-  RTC_DCHECK(extensions);
-  bool found = false;
-  for (const char* name : extensions_decreasing_prio) {
-    auto it = std::find_if(extensions->begin(), extensions->end(),
-        [name](const webrtc::RtpExtension& rhs) {
-          return rhs.name == name;
-        });
-    if (it != extensions->end()) {
-      if (found) {
-        extensions->erase(it);
-      }
-      found = true;
-    }
-  }
-}
-}  // namespace
-
-bool ValidateRtpExtensions(const std::vector<RtpHeaderExtension>& extensions) {
-  bool id_used[14] = {false};
-  for (const auto& extension : extensions) {
-    if (extension.id <= 0 || extension.id >= 15) {
-      LOG(LS_ERROR) << "Bad RTP extension ID: " << extension.ToString();
-      return false;
-    }
-    if (id_used[extension.id - 1]) {
-      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<RtpHeaderExtension>& 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.uri, extension.id});
-    } else {
-      LOG(LS_WARNING) << "Unsupported RTP extension: " << extension.ToString();
-    }
-  }
-
-  // Sort by name, ascending, 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.name < lhs.name;
-      });
-
-  // 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.name == lhs.name;
-        });
-    result.erase(it, result.end());
-
-    // Keep just the highest priority extension of any in the following list.
-    static const char* kBweExtensionPriorities[] = {
-      kRtpTransportSequenceNumberHeaderExtension,
-      kRtpAbsoluteSenderTimeHeaderExtension,
-      kRtpTimestampOffsetHeaderExtension
-    };
-    DiscardRedundantExtensions(&result, kBweExtensionPriorities);
-  }
-
-  return result;
-}
-}  // namespace cricket