Moving src/webrtc into src/.

In order to eliminate the WebRTC Subtree mirror in Chromium, WebRTC is moving the content of the src/webrtc directory up to the src/ directory. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38 Reviewed-on: https://webrtc-review.googlesource.com/1560 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19845}
2017-09-15 06:15:48 +02:00
parent 6674846b4a
commit bb547203bf
4576 changed files with 1092 additions and 1196 deletions
--- a/modules/rtp_rtcp/include/rtp_payload_registry.h
+++ b/modules/rtp_rtcp/include/rtp_payload_registry.h
@ -0,0 +1,130 @@
+/*
+ *  Copyright (c) 2013 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.
+ */
+
+#ifndef WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_PAYLOAD_REGISTRY_H_
+#define WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_PAYLOAD_REGISTRY_H_
+
+#include <map>
+#include <set>
+
+#include "webrtc/api/audio_codecs/audio_format.h"
+#include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
+#include "webrtc/rtc_base/criticalsection.h"
+
+namespace webrtc {
+
+struct CodecInst;
+class VideoCodec;
+
+class RTPPayloadRegistry {
+ public:
+  RTPPayloadRegistry();
+  ~RTPPayloadRegistry();
+
+  // TODO(magjed): Split RTPPayloadRegistry into separate Audio and Video class
+  // and simplify the code. http://crbug/webrtc/6743.
+
+  // Replace all audio receive payload types with the given map.
+  void SetAudioReceivePayloads(std::map<int, SdpAudioFormat> codecs);
+
+  int32_t RegisterReceivePayload(const CodecInst& audio_codec,
+                                 bool* created_new_payload_type);
+  int32_t RegisterReceivePayload(const VideoCodec& video_codec);
+
+  int32_t DeRegisterReceivePayload(int8_t payload_type);
+
+  int32_t ReceivePayloadType(const CodecInst& audio_codec,
+                             int8_t* payload_type) const;
+  int32_t ReceivePayloadType(const VideoCodec& video_codec,
+                             int8_t* payload_type) const;
+
+  bool RtxEnabled() const;
+
+  void SetRtxSsrc(uint32_t ssrc);
+
+  bool GetRtxSsrc(uint32_t* ssrc) const;
+
+  void SetRtxPayloadType(int payload_type, int associated_payload_type);
+
+  bool IsRed(const RTPHeader& header) const;
+
+  bool GetPayloadSpecifics(uint8_t payload_type, PayloadUnion* payload) const;
+
+  int GetPayloadTypeFrequency(uint8_t payload_type) const;
+
+  const RtpUtility::Payload* PayloadTypeToPayload(uint8_t payload_type) const;
+
+  void ResetLastReceivedPayloadTypes() {
+    rtc::CritScope cs(&crit_sect_);
+    last_received_payload_type_ = -1;
+    last_received_media_payload_type_ = -1;
+  }
+
+  // This sets the payload type of the packets being received from the network
+  // on the media SSRC. For instance if packets are encapsulated with RED, this
+  // payload type will be the RED payload type.
+  void SetIncomingPayloadType(const RTPHeader& header);
+
+  // Returns true if the new media payload type has not changed.
+  bool ReportMediaPayloadType(uint8_t media_payload_type);
+
+  int8_t red_payload_type() const { return GetPayloadTypeWithName("red"); }
+  int8_t ulpfec_payload_type() const {
+    return GetPayloadTypeWithName("ulpfec");
+  }
+  int8_t last_received_payload_type() const {
+    rtc::CritScope cs(&crit_sect_);
+    return last_received_payload_type_;
+  }
+  void set_last_received_payload_type(int8_t last_received_payload_type) {
+    rtc::CritScope cs(&crit_sect_);
+    last_received_payload_type_ = last_received_payload_type;
+  }
+
+  int8_t last_received_media_payload_type() const {
+    rtc::CritScope cs(&crit_sect_);
+    return last_received_media_payload_type_;
+  }
+
+ private:
+  // Prunes the payload type map of the specific payload type, if it exists.
+  void DeregisterAudioCodecOrRedTypeRegardlessOfPayloadType(
+      const CodecInst& audio_codec);
+
+  bool IsRtxInternal(const RTPHeader& header) const;
+  // Returns the payload type for the payload with name |payload_name|, or -1 if
+  // no such payload is registered.
+  int8_t GetPayloadTypeWithName(const char* payload_name) const;
+
+  rtc::CriticalSection crit_sect_;
+  std::map<int, RtpUtility::Payload> payload_type_map_;
+  int8_t incoming_payload_type_;
+  int8_t last_received_payload_type_;
+  int8_t last_received_media_payload_type_;
+  bool rtx_;
+  // Mapping rtx_payload_type_map_[rtx] = associated.
+  std::map<int, int> rtx_payload_type_map_;
+  uint32_t ssrc_rtx_;
+  // Only warn once per payload type, if an RTX packet is received but
+  // no associated payload type found in |rtx_payload_type_map_|.
+  std::set<int> payload_types_with_suppressed_warnings_
+      RTC_GUARDED_BY(crit_sect_);
+
+  // As a first step in splitting this class up in separate cases for audio and
+  // video, DCHECK that no instance is used for both audio and video.
+#if RTC_DCHECK_IS_ON
+  bool used_for_audio_ RTC_GUARDED_BY(crit_sect_) = false;
+  bool used_for_video_ RTC_GUARDED_BY(crit_sect_) = false;
+#endif
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_PAYLOAD_REGISTRY_H_