#include "base/threading/non_thread_safe.h"
#include "base/time/tick_clock.h"
#include "base/time/time.h"
+#include "media/base/audio_bus.h"
#include "media/cast/cast_config.h"
+#include "media/cast/cast_environment.h"
+#include "media/cast/logging/logging_defines.h"
#include "media/cast/rtcp/rtcp.h"
-#include "media/cast/transport/rtp_sender/rtp_sender.h"
-
-namespace media {
-class AudioBus;
-}
+#include "media/cast/rtp_timestamp_helper.h"
namespace media {
namespace cast {
class AudioEncoder;
-class LocalRtcpAudioSenderFeedback;
-class LocalRtpSenderStatistics;
-// This class is not thread safe.
-// It's only called from the main cast thread.
-class AudioSender : public base::NonThreadSafe,
+// Not thread safe. Only called from the main cast thread.
+// This class owns all objects related to sending audio, objects that create RTP
+// packets, congestion control, audio encoder, parsing and sending of
+// RTCP packets.
+// Additionally it posts a bunch of delayed tasks to the main thread for various
+// timeouts.
+class AudioSender : public RtcpSenderFeedback,
+ public base::NonThreadSafe,
public base::SupportsWeakPtr<AudioSender> {
public:
AudioSender(scoped_refptr<CastEnvironment> cast_environment,
virtual ~AudioSender();
CastInitializationStatus InitializationResult() const {
- return initialization_status_;
+ return cast_initialization_status_;
}
- // The |audio_bus| must be valid until the |done_callback| is called.
- // The callback is called from the main cast thread as soon as the encoder is
- // done with |audio_bus|; it does not mean that the encoded data has been
- // sent out.
- void InsertAudio(const AudioBus* audio_bus,
- const base::TimeTicks& recorded_time,
- const base::Closure& done_callback);
+ // Note: It is not guaranteed that |audio_frame| will actually be encoded and
+ // sent, if AudioSender detects too many frames in flight. Therefore, clients
+ // should be careful about the rate at which this method is called.
+ //
+ // Note: It is invalid to call this method if InitializationResult() returns
+ // anything but STATUS_AUDIO_INITIALIZED.
+ void InsertAudio(scoped_ptr<AudioBus> audio_bus,
+ const base::TimeTicks& recorded_time);
// Only called from the main cast thread.
void IncomingRtcpPacket(scoped_ptr<Packet> packet);
protected:
- void SendEncodedAudioFrame(
- scoped_ptr<transport::EncodedAudioFrame> audio_frame,
- const base::TimeTicks& recorded_time);
+ // Protected for testability.
+ virtual void OnReceivedCastFeedback(const RtcpCastMessage& cast_feedback)
+ OVERRIDE;
private:
- friend class LocalRtcpAudioSenderFeedback;
+ // Schedule and execute periodic sending of RTCP report.
+ void ScheduleNextRtcpReport();
+ void SendRtcpReport(bool schedule_future_reports);
+
+ // Schedule and execute periodic checks for re-sending packets. If no
+ // acknowledgements have been received for "too long," AudioSender will
+ // speculatively re-send certain packets of an unacked frame to kick-start
+ // re-transmission. This is a last resort tactic to prevent the session from
+ // getting stuck after a long outage.
+ void ScheduleNextResendCheck();
+ void ResendCheck();
+ void ResendForKickstart();
+
+ // Returns true if there are too many frames in flight, as defined by the
+ // configured target playout delay plus simple logic. When this is true,
+ // InsertAudio() will silenty drop frames instead of sending them to the audio
+ // encoder.
+ bool AreTooManyFramesInFlight() const;
+
+ // Called by the |audio_encoder_| with the next EncodedFrame to send.
+ void SendEncodedAudioFrame(scoped_ptr<transport::EncodedFrame> audio_frame);
+
+ const scoped_refptr<CastEnvironment> cast_environment_;
+
+ // The total amount of time between a frame's capture/recording on the sender
+ // and its playback on the receiver (i.e., shown to a user). This is fixed as
+ // a value large enough to give the system sufficient time to encode,
+ // transmit/retransmit, receive, decode, and render; given its run-time
+ // environment (sender/receiver hardware performance, network conditions,
+ // etc.).
+ const base::TimeDelta target_playout_delay_;
+
+ // Sends encoded frames over the configured transport (e.g., UDP). In
+ // Chromium, this could be a proxy that first sends the frames from a renderer
+ // process to the browser process over IPC, with the browser process being
+ // responsible for "packetizing" the frames and pushing packets into the
+ // network layer.
+ transport::CastTransportSender* const transport_sender_;
- void SendEncodedAudioFrameToTransport(
- scoped_ptr<transport::EncodedAudioFrame> audio_frame,
- const base::TimeTicks& recorded_time);
+ // Maximum number of outstanding frames before the encoding and sending of
+ // new frames shall halt.
+ const int max_unacked_frames_;
- void ResendPackets(
- const MissingFramesAndPacketsMap& missing_frames_and_packets);
+ // Encodes AudioBuses into EncodedFrames.
+ scoped_ptr<AudioEncoder> audio_encoder_;
+ const int configured_encoder_bitrate_;
- void ResendPacketsOnTransportThread(
- const transport::MissingFramesAndPacketsMap& missing_packets);
+ // Manages sending/receiving of RTCP packets, including sender/receiver
+ // reports.
+ Rtcp rtcp_;
- void ScheduleNextRtcpReport();
- void SendRtcpReport();
+ // Records lip-sync (i.e., mapping of RTP <--> NTP timestamps), and
+ // extrapolates this mapping to any other point in time.
+ RtpTimestampHelper rtp_timestamp_helper_;
- void InitializeTimers();
+ // Counts how many RTCP reports are being "aggressively" sent (i.e., one per
+ // frame) at the start of the session. Once a threshold is reached, RTCP
+ // reports are instead sent at the configured interval + random drift.
+ int num_aggressive_rtcp_reports_sent_;
- base::WeakPtrFactory<AudioSender> weak_factory_;
+ // This is "null" until the first frame is sent. Thereafter, this tracks the
+ // last time any frame was sent or re-sent.
+ base::TimeTicks last_send_time_;
- scoped_refptr<CastEnvironment> cast_environment_;
- transport::CastTransportSender* const transport_sender_;
- scoped_refptr<AudioEncoder> audio_encoder_;
- scoped_ptr<LocalRtpSenderStatistics> rtp_audio_sender_statistics_;
- scoped_ptr<LocalRtcpAudioSenderFeedback> rtcp_feedback_;
- Rtcp rtcp_;
- bool timers_initialized_;
- CastInitializationStatus initialization_status_;
+ // The ID of the last frame sent. Logic throughout AudioSender assumes this
+ // can safely wrap-around. This member is invalid until
+ // |!last_send_time_.is_null()|.
+ uint32 last_sent_frame_id_;
+
+ // The ID of the latest (not necessarily the last) frame that has been
+ // acknowledged. Logic throughout AudioSender assumes this can safely
+ // wrap-around. This member is invalid until |!last_send_time_.is_null()|.
+ uint32 latest_acked_frame_id_;
+
+ // Counts the number of duplicate ACK that are being received. When this
+ // number reaches a threshold, the sender will take this as a sign that the
+ // receiver hasn't yet received the first packet of the next frame. In this
+ // case, AudioSender will trigger a re-send of the next frame.
+ int duplicate_ack_counter_;
+
+ // If this sender is ready for use, this is STATUS_AUDIO_INITIALIZED.
+ CastInitializationStatus cast_initialization_status_;
+
+ // This is a "good enough" mapping for finding the RTP timestamp associated
+ // with a video frame. The key is the lowest 8 bits of frame id (which is
+ // what is sent via RTCP). This map is used for logging purposes.
+ RtpTimestamp frame_id_to_rtp_timestamp_[256];
+
+ // NOTE: Weak pointers must be invalidated before all other member variables.
+ base::WeakPtrFactory<AudioSender> weak_factory_;
DISALLOW_COPY_AND_ASSIGN(AudioSender);
};
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