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// Copyright 2014 The Chromium Authors. All rights reserved.
// Use of this source code is governed by a BSD-style license that can be
// found in the LICENSE file.
#ifndef MEDIA_CAST_SENDER_AUDIO_SENDER_H_
#define MEDIA_CAST_SENDER_AUDIO_SENDER_H_
#include "base/callback.h"
#include "base/memory/ref_counted.h"
#include "base/memory/scoped_ptr.h"
#include "base/memory/weak_ptr.h"
#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/sender/frame_sender.h"
namespace media {
namespace cast {
class AudioEncoder;
// 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 FrameSender,
public base::NonThreadSafe,
public base::SupportsWeakPtr<AudioSender> {
public:
AudioSender(scoped_refptr<CastEnvironment> cast_environment,
const AudioSenderConfig& audio_config,
CastTransportSender* const transport_sender);
virtual ~AudioSender();
CastInitializationStatus InitializationResult() const {
return cast_initialization_status_;
}
// 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);
protected:
// Protected for testability.
void OnReceivedCastFeedback(const RtcpCastMessage& cast_feedback);
private:
// 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<EncodedFrame> audio_frame);
// Encodes AudioBuses into EncodedFrames.
scoped_ptr<AudioEncoder> audio_encoder_;
const int configured_encoder_bitrate_;
// 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_;
// 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_;
// 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);
};
} // namespace cast
} // namespace media
#endif // MEDIA_CAST_SENDER_AUDIO_SENDER_H_
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