Moved call.h and most of api/call/* into call/

webrtc/call.h

Index: webrtc/call.h
diff --git a/webrtc/call.h b/webrtc/call.h
index 26f8c82bd9bca40a5d9ad1bf387d86caaa8c57a6..afea9ddd724056f03ca05e629cf54f74b66183c5 100644
--- a/webrtc/call.h
+++ b/webrtc/call.h
@@ -7,159 +7,7 @@
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
-#ifndef WEBRTC_CALL_H_
-#define WEBRTC_CALL_H_
 
-#include <string>
-#include <vector>
-
-#include "webrtc/api/call/audio_receive_stream.h"
-#include "webrtc/api/call/audio_send_stream.h"
-#include "webrtc/api/call/audio_state.h"
-#include "webrtc/api/call/flexfec_receive_stream.h"
-#include "webrtc/base/networkroute.h"
-#include "webrtc/base/platform_file.h"
-#include "webrtc/base/socket.h"
-#include "webrtc/common_types.h"
-#include "webrtc/video_receive_stream.h"
-#include "webrtc/video_send_stream.h"
-
-namespace webrtc {
-
-class AudioProcessing;
-class RtcEventLog;
-
-const char* Version();
-
-enum class MediaType {
-  ANY,
-  AUDIO,
-  VIDEO,
-  DATA
-};
-
-class PacketReceiver {
- public:
-  enum DeliveryStatus {
-    DELIVERY_OK,
-    DELIVERY_UNKNOWN_SSRC,
-    DELIVERY_PACKET_ERROR,
-  };
-
-  virtual DeliveryStatus DeliverPacket(MediaType media_type,
-                                       const uint8_t* packet,
-                                       size_t length,
-                                       const PacketTime& packet_time) = 0;
-
- protected:
-  virtual ~PacketReceiver() {}
-};
-
-// A Call instance can contain several send and/or receive streams. All streams
-// are assumed to have the same remote endpoint and will share bitrate estimates
-// etc.
-class Call {
- public:
-  struct Config {
-    explicit Config(RtcEventLog* event_log) : event_log(event_log) {
-      RTC_DCHECK(event_log);
-    }
-
-    static const int kDefaultStartBitrateBps;
-
-    // Bitrate config used until valid bitrate estimates are calculated. Also
-    // used to cap total bitrate used.
-    struct BitrateConfig {
-      int min_bitrate_bps = 0;
-      int start_bitrate_bps = kDefaultStartBitrateBps;
-      int max_bitrate_bps = -1;
-    } bitrate_config;
-
-    // AudioState which is possibly shared between multiple calls.
-    // TODO(solenberg): Change this to a shared_ptr once we can use C++11.
-    rtc::scoped_refptr<AudioState> audio_state;
-
-    // Audio Processing Module to be used in this call.
-    // TODO(solenberg): Change this to a shared_ptr once we can use C++11.
-    AudioProcessing* audio_processing = nullptr;
-
-    // RtcEventLog to use for this call. Required.
-    // Use webrtc::RtcEventLog::CreateNull() for a null implementation.
-    RtcEventLog* event_log = nullptr;
-  };
-
-  struct Stats {
-    std::string ToString(int64_t time_ms) const;
-
-    int send_bandwidth_bps = 0;       // Estimated available send bandwidth.
-    int max_padding_bitrate_bps = 0;  // Cumulative configured max padding.
-    int recv_bandwidth_bps = 0;       // Estimated available receive bandwidth.
-    int64_t pacer_delay_ms = 0;
-    int64_t rtt_ms = -1;
-  };
-
-  static Call* Create(const Call::Config& config);
-
-  virtual AudioSendStream* CreateAudioSendStream(
-      const AudioSendStream::Config& config) = 0;
-  virtual void DestroyAudioSendStream(AudioSendStream* send_stream) = 0;
-
-  virtual AudioReceiveStream* CreateAudioReceiveStream(
-      const AudioReceiveStream::Config& config) = 0;
-  virtual void DestroyAudioReceiveStream(
-      AudioReceiveStream* receive_stream) = 0;
-
-  virtual VideoSendStream* CreateVideoSendStream(
-      VideoSendStream::Config config,
-      VideoEncoderConfig encoder_config) = 0;
-  virtual void DestroyVideoSendStream(VideoSendStream* send_stream) = 0;
-
-  virtual VideoReceiveStream* CreateVideoReceiveStream(
-      VideoReceiveStream::Config configuration) = 0;
-  virtual void DestroyVideoReceiveStream(
-      VideoReceiveStream* receive_stream) = 0;
-
-  virtual FlexfecReceiveStream* CreateFlexfecReceiveStream(
-      FlexfecReceiveStream::Config configuration) = 0;
-  virtual void DestroyFlexfecReceiveStream(
-      FlexfecReceiveStream* receive_stream) = 0;
-
-  // All received RTP and RTCP packets for the call should be inserted to this
-  // PacketReceiver. The PacketReceiver pointer is valid as long as the
-  // Call instance exists.
-  virtual PacketReceiver* Receiver() = 0;
-
-  // Returns the call statistics, such as estimated send and receive bandwidth,
-  // pacing delay, etc.
-  virtual Stats GetStats() const = 0;
-
-  // TODO(pbos): Like BitrateConfig above this is currently per-stream instead
-  // of maximum for entire Call. This should be fixed along with the above.
-  // Specifying a start bitrate (>0) will currently reset the current bitrate
-  // estimate. This is due to how the 'x-google-start-bitrate' flag is currently
-  // implemented.
-  virtual void SetBitrateConfig(
-      const Config::BitrateConfig& bitrate_config) = 0;
-
-  // TODO(skvlad): When the unbundled case with multiple streams for the same
-  // media type going over different networks is supported, track the state
-  // for each stream separately. Right now it's global per media type.
-  virtual void SignalChannelNetworkState(MediaType media,
-                                         NetworkState state) = 0;
-
-  virtual void OnTransportOverheadChanged(
-      MediaType media,
-      int transport_overhead_per_packet) = 0;
-
-  virtual void OnNetworkRouteChanged(
-      const std::string& transport_name,
-      const rtc::NetworkRoute& network_route) = 0;
-
-  virtual void OnSentPacket(const rtc::SentPacket& sent_packet) = 0;
-
-  virtual ~Call() {}
-};
-
-}  // namespace webrtc
-
-#endif  // WEBRTC_CALL_H_
+// This file is deprecated. It has been moved to the location below. Please
+// update your includes! See: http://bugs.webrtc.org/6716
+#include "webrtc/call/call.h"