Split out RtpSource from libjingle_peerconnection_api

And moved declaration into a new api directory, as
api/transport/rtp/rtp_source.h.

Bug: webrtc:8733
Change-Id: Ia73b7b0630e6065de4707a37633adddfa00a2b8a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150880
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29039}

api/BUILD.gn

@@ -177,6 +177,7 @@ rtc_static_library("libjingle_peerconnection_api") {
     "transport:network_control",
     "transport/media:audio_interfaces",
     "transport/media:video_interfaces",
+    "transport/rtp:rtp_source",
     "units:data_rate",
     "units:timestamp",
     "video:encoded_image",

api/rtp_receiver_interface.cc

@@ -12,21 +12,6 @@
 
 namespace webrtc {
 
-RtpSource::RtpSource(int64_t timestamp_ms,
-                     uint32_t source_id,
-                     RtpSourceType source_type,
-                     absl::optional<uint8_t> audio_level,
-                     uint32_t rtp_timestamp)
-    : timestamp_ms_(timestamp_ms),
-      source_id_(source_id),
-      source_type_(source_type),
-      audio_level_(audio_level),
-      rtp_timestamp_(rtp_timestamp) {}
-
-RtpSource::RtpSource(const RtpSource&) = default;
-RtpSource& RtpSource::operator=(const RtpSource&) = default;
-RtpSource::~RtpSource() = default;
-
 std::vector<std::string> RtpReceiverInterface::stream_ids() const {
   return {};
 }

api/rtp_receiver_interface.h

@@ -24,63 +24,12 @@
 #include "api/proxy.h"
 #include "api/rtp_parameters.h"
 #include "api/scoped_refptr.h"
+#include "api/transport/rtp/rtp_source.h"
 #include "rtc_base/deprecation.h"
 #include "rtc_base/ref_count.h"
 
 namespace webrtc {
 
-enum class RtpSourceType {
-  SSRC,
-  CSRC,
-};
-
-class RtpSource {
- public:
-  RtpSource() = delete;
-
-  RtpSource(int64_t timestamp_ms,
-            uint32_t source_id,
-            RtpSourceType source_type,
-            absl::optional<uint8_t> audio_level,
-            uint32_t rtp_timestamp);
-
-  RtpSource(const RtpSource&);
-  RtpSource& operator=(const RtpSource&);
-  ~RtpSource();
-
-  int64_t timestamp_ms() const { return timestamp_ms_; }
-  void update_timestamp_ms(int64_t timestamp_ms) {
-    RTC_DCHECK_LE(timestamp_ms_, timestamp_ms);
-    timestamp_ms_ = timestamp_ms;
-  }
-
-  // The identifier of the source can be the CSRC or the SSRC.
-  uint32_t source_id() const { return source_id_; }
-
-  // The source can be either a contributing source or a synchronization source.
-  RtpSourceType source_type() const { return source_type_; }
-
-  absl::optional<uint8_t> audio_level() const { return audio_level_; }
-  void set_audio_level(const absl::optional<uint8_t>& level) {
-    audio_level_ = level;
-  }
-
-  uint32_t rtp_timestamp() const { return rtp_timestamp_; }
-
-  bool operator==(const RtpSource& o) const {
-    return timestamp_ms_ == o.timestamp_ms() && source_id_ == o.source_id() &&
-           source_type_ == o.source_type() && audio_level_ == o.audio_level_ &&
-           rtp_timestamp_ == o.rtp_timestamp();
-  }
-
- private:
-  int64_t timestamp_ms_;
-  uint32_t source_id_;
-  RtpSourceType source_type_;
-  absl::optional<uint8_t> audio_level_;
-  uint32_t rtp_timestamp_;
-};
-
 class RtpReceiverObserverInterface {
  public:
   // Note: Currently if there are multiple RtpReceivers of the same media type,

api/transport/rtp/BUILD.gn

@@ -0,0 +1,20 @@
+# Copyright (c) 2019 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.
+
+import("../../../webrtc.gni")
+
+rtc_source_set("rtp_source") {
+  visibility = [ "*" ]
+  sources = [
+    "rtp_source.h",
+  ]
+  deps = [
+    "../../../rtc_base:checks",
+    "//third_party/abseil-cpp/absl/types:optional",
+  ]
+}

api/transport/rtp/rtp_source.h

@@ -0,0 +1,80 @@
+/*
+ *  Copyright 2019 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 API_TRANSPORT_RTP_RTP_SOURCE_H_
+#define API_TRANSPORT_RTP_RTP_SOURCE_H_
+
+#include <stdint.h>
+
+#include "absl/types/optional.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+enum class RtpSourceType {
+  SSRC,
+  CSRC,
+};
+
+class RtpSource {
+ public:
+  RtpSource() = delete;
+
+  RtpSource(int64_t timestamp_ms,
+            uint32_t source_id,
+            RtpSourceType source_type,
+            absl::optional<uint8_t> audio_level,
+            uint32_t rtp_timestamp)
+      : timestamp_ms_(timestamp_ms),
+        source_id_(source_id),
+        source_type_(source_type),
+        audio_level_(audio_level),
+        rtp_timestamp_(rtp_timestamp) {}
+
+  RtpSource(const RtpSource&) = default;
+  RtpSource& operator=(const RtpSource&) = default;
+  ~RtpSource() = default;
+
+  int64_t timestamp_ms() const { return timestamp_ms_; }
+  void update_timestamp_ms(int64_t timestamp_ms) {
+    RTC_DCHECK_LE(timestamp_ms_, timestamp_ms);
+    timestamp_ms_ = timestamp_ms;
+  }
+
+  // The identifier of the source can be the CSRC or the SSRC.
+  uint32_t source_id() const { return source_id_; }
+
+  // The source can be either a contributing source or a synchronization source.
+  RtpSourceType source_type() const { return source_type_; }
+
+  absl::optional<uint8_t> audio_level() const { return audio_level_; }
+  void set_audio_level(const absl::optional<uint8_t>& level) {
+    audio_level_ = level;
+  }
+
+  uint32_t rtp_timestamp() const { return rtp_timestamp_; }
+
+  bool operator==(const RtpSource& o) const {
+    return timestamp_ms_ == o.timestamp_ms() && source_id_ == o.source_id() &&
+           source_type_ == o.source_type() && audio_level_ == o.audio_level_ &&
+           rtp_timestamp_ == o.rtp_timestamp();
+  }
+
+ private:
+  int64_t timestamp_ms_;
+  uint32_t source_id_;
+  RtpSourceType source_type_;
+  absl::optional<uint8_t> audio_level_;
+  uint32_t rtp_timestamp_;
+};
+
+}  // namespace webrtc
+
+#endif  // API_TRANSPORT_RTP_RTP_SOURCE_H_

audio/BUILD.gn

@@ -54,6 +54,7 @@ rtc_static_library("audio") {
     "../api/audio_codecs:audio_codecs_api",
     "../api/rtc_event_log",
     "../api/task_queue",
+    "../api/transport/rtp:rtp_source",
     "../call:bitrate_allocator",
     "../call:call_interfaces",
     "../call:rtp_interfaces",

audio/channel_receive.cc

@@ -18,6 +18,7 @@
 #include <vector>
 
 #include "absl/memory/memory.h"
+#include "api/crypto/frame_decryptor_interface.h"
 #include "api/rtc_event_log/rtc_event_log.h"
 #include "audio/audio_level.h"
 #include "audio/channel_send.h"

audio/channel_receive.h

@@ -24,7 +24,7 @@
 #include "api/crypto/crypto_options.h"
 #include "api/media_transport_config.h"
 #include "api/media_transport_interface.h"
-#include "api/rtp_receiver_interface.h"
+#include "api/transport/rtp/rtp_source.h"
 #include "call/rtp_packet_sink_interface.h"
 #include "call/syncable.h"
 #include "modules/audio_coding/include/audio_coding_module_typedefs.h"

call/BUILD.gn

@@ -43,6 +43,7 @@ rtc_source_set("call_interfaces") {
     "../api/audio_codecs:audio_codecs_api",
     "../api/task_queue",
     "../api/transport:network_control",
+    "../api/transport/rtp:rtp_source",
     "../modules/audio_device",
     "../modules/audio_processing",
     "../modules/audio_processing:api",
@@ -284,6 +285,7 @@ rtc_source_set("video_stream_api") {
     "../api:rtp_headers",
     "../api:rtp_parameters",
     "../api:transport_api",
+    "../api/transport/rtp:rtp_source",
     "../api/video:video_frame",
     "../api/video:video_rtp_headers",
     "../api/video:video_stream_encoder",
@@ -449,6 +451,7 @@ if (rtc_include_tests) {
       "../modules/audio_device:audio_device_impl",
       "../modules/audio_mixer:audio_mixer_impl",
       "../modules/rtp_rtcp",
+      "../rtc_base",
       "../rtc_base:checks",
       "../rtc_base:rtc_base_approved",
       "../system_wrappers",

call/audio_receive_stream.h

@@ -20,15 +20,15 @@
 #include "api/audio_codecs/audio_decoder_factory.h"
 #include "api/call/transport.h"
 #include "api/crypto/crypto_options.h"
+#include "api/crypto/frame_decryptor_interface.h"
 #include "api/media_transport_config.h"
 #include "api/rtp_parameters.h"
-#include "api/rtp_receiver_interface.h"
 #include "api/scoped_refptr.h"
+#include "api/transport/rtp/rtp_source.h"
 #include "call/rtp_config.h"
 
 namespace webrtc {
 class AudioSinkInterface;
-class FrameDecryptorInterface;
 
 class AudioReceiveStream {
  public:

call/call_perf_tests.cc

@@ -28,6 +28,7 @@
 #include "modules/audio_device/include/test_audio_device.h"
 #include "modules/audio_mixer/audio_mixer_impl.h"
 #include "rtc_base/checks.h"
+#include "rtc_base/thread.h"
 #include "rtc_base/thread_annotations.h"
 #include "system_wrappers/include/metrics.h"
 #include "test/call_test.h"

call/video_receive_stream.h

@@ -19,12 +19,14 @@
 
 #include "api/call/transport.h"
 #include "api/crypto/crypto_options.h"
+#include "api/crypto/frame_decryptor_interface.h"
 #include "api/media_transport_config.h"
 #include "api/media_transport_interface.h"
 #include "api/rtp_headers.h"
 #include "api/rtp_parameters.h"
-#include "api/rtp_receiver_interface.h"
+#include "api/transport/rtp/rtp_source.h"
 #include "api/video/video_content_type.h"
+#include "api/video/video_frame.h"
 #include "api/video/video_sink_interface.h"
 #include "api/video/video_timing.h"
 #include "api/video_codecs/sdp_video_format.h"
@@ -34,7 +36,6 @@
 
 namespace webrtc {
 
-class FrameDecryptorInterface;
 class RtpPacketSinkInterface;
 class VideoDecoderFactory;
 

media/BUILD.gn

@@ -78,6 +78,7 @@ rtc_static_library("rtc_media_base") {
     "../api:rtp_parameters",
     "../api:scoped_refptr",
     "../api/audio_codecs:audio_codecs_api",
+    "../api/transport/rtp:rtp_source",
     "../api/video:video_bitrate_allocation",
     "../api/video:video_bitrate_allocator_factory",
     "../api/video:video_frame",
@@ -264,6 +265,7 @@ rtc_static_library("rtc_audio_video") {
     "../api/audio:audio_mixer_api",
     "../api/audio_codecs:audio_codecs_api",
     "../api/task_queue",
+    "../api/transport/rtp:rtp_source",
     "../api/video:video_bitrate_allocation",
     "../api/video:video_bitrate_allocator_factory",
     "../api/video:video_codec_constants",

media/base/media_channel.h

@@ -25,7 +25,7 @@
 #include "api/media_transport_config.h"
 #include "api/rtc_error.h"
 #include "api/rtp_parameters.h"
-#include "api/rtp_receiver_interface.h"
+#include "api/transport/rtp/rtp_source.h"
 #include "api/video/video_content_type.h"
 #include "api/video/video_sink_interface.h"
 #include "api/video/video_source_interface.h"
@@ -42,6 +42,7 @@
 #include "rtc_base/async_packet_socket.h"
 #include "rtc_base/buffer.h"
 #include "rtc_base/copy_on_write_buffer.h"
+#include "rtc_base/critical_section.h"
 #include "rtc_base/dscp.h"
 #include "rtc_base/logging.h"
 #include "rtc_base/network_route.h"

media/engine/webrtc_voice_engine.h

@@ -17,9 +17,9 @@
 #include <vector>
 
 #include "api/audio_codecs/audio_encoder_factory.h"
-#include "api/rtp_receiver_interface.h"
 #include "api/scoped_refptr.h"
 #include "api/task_queue/task_queue_factory.h"
+#include "api/transport/rtp/rtp_source.h"
 #include "call/audio_state.h"
 #include "call/call.h"
 #include "media/base/media_engine.h"

modules/rtp_rtcp/BUILD.gn

@@ -228,6 +228,7 @@ rtc_static_library("rtp_rtcp") {
     "../../api/rtc_event_log",
     "../../api/transport:field_trial_based_config",
     "../../api/transport:webrtc_key_value_config",
+    "../../api/transport/rtp:rtp_source",
     "../../api/units:time_delta",
     "../../api/units:timestamp",
     "../../api/video:video_bitrate_allocation",

modules/rtp_rtcp/source/source_tracker.h

@@ -19,7 +19,7 @@
 
 #include "absl/types/optional.h"
 #include "api/rtp_packet_infos.h"
-#include "api/rtp_receiver_interface.h"
+#include "api/transport/rtp/rtp_source.h"
 #include "rtc_base/critical_section.h"
 #include "rtc_base/time_utils.h"
 #include "system_wrappers/include/clock.h"

pc/BUILD.gn

@@ -541,6 +541,7 @@ if (rtc_include_tests) {
       "../api/rtc_event_log",
       "../api/rtc_event_log:rtc_event_log_factory",
       "../api/task_queue:default_task_queue_factory",
+      "../api/transport/rtp:rtp_source",
       "../api/units:time_delta",
       "../api/video:builtin_video_bitrate_allocator_factory",
       "../logging:fake_rtc_event_log",

pc/track_media_info_map_unittest.cc

@@ -16,8 +16,8 @@
 #include <utility>
 #include <vector>
 
-#include "api/rtp_receiver_interface.h"
 #include "api/rtp_sender_interface.h"
+#include "api/transport/rtp/rtp_source.h"
 #include "media/base/media_channel.h"
 #include "pc/audio_track.h"
 #include "pc/test/fake_video_track_source.h"

test/scenario/stats_collection.cc

@@ -12,6 +12,7 @@
 
 #include "common_video/libyuv/include/webrtc_libyuv.h"
 #include "rtc_base/memory_usage.h"
+#include "rtc_base/thread.h"
 
 namespace webrtc {
 namespace test {