Moved call.h and most of api/call/* into call/
BUG=webrtc:6716
Review-Url: https://codereview.webrtc.org/2550273003
Cr-Commit-Position: refs/heads/master@{#15460}
diff --git a/webrtc/call.h b/webrtc/call.h
index 26f8c82..afea9dd 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"