#include "media/cast/cast_config.h"
#include "media/cast/cast_environment.h"
#include "media/cast/cast_receiver.h"
+#include "media/cast/framer/framer.h"
#include "media/cast/rtcp/receiver_rtcp_event_subscriber.h"
-#include "media/cast/rtcp/rtcp.h" // RtcpCastMessage
+#include "media/cast/rtcp/rtcp.h"
#include "media/cast/rtp_receiver/rtp_receiver.h"
-#include "media/cast/rtp_receiver/rtp_receiver_defines.h" // RtpCastHeader
+#include "media/cast/rtp_receiver/rtp_receiver_defines.h"
#include "media/cast/transport/utility/transport_encryption_handler.h"
namespace media {
namespace cast {
class AudioDecoder;
-class Framer;
-class LocalRtpAudioFeedback;
-class RtpReceiver;
-class RtpReceiverStatistics;
-
-struct DecodedAudioCallbackData {
- DecodedAudioCallbackData();
- ~DecodedAudioCallbackData();
- int number_of_10ms_blocks;
- int desired_frequency;
- AudioFrameDecodedCallback callback;
-};
-// This class is not thread safe. Should only be called from the Main cast
+// AudioReceiver receives packets out-of-order while clients make requests for
+// complete frames in-order. (A frame consists of one or more packets.)
+//
+// AudioReceiver also includes logic for computing the playout time for each
+// frame, accounting for a constant targeted playout delay. The purpose of the
+// playout delay is to provide a fixed window of time between the capture event
+// on the sender and the playout on the receiver. This is important because
+// each step of the pipeline (i.e., encode frame, then transmit/retransmit from
+// the sender, then receive and re-order packets on the receiver, then decode
+// frame) can vary in duration and is typically very hard to predict.
+// Heuristics will determine when the targeted playout delay is insufficient in
+// the current environment; and the receiver can then increase the playout
+// delay, notifying the sender, to account for the extra variance.
+// TODO(miu): Make the last sentence true. http://crbug.com/360111
+//
+// Two types of frames can be requested: 1) A frame of decoded audio data; or 2)
+// a frame of still-encoded audio data, to be passed into an external audio
+// decoder. Each request for a frame includes a callback which AudioReceiver
+// guarantees will be called at some point in the future unless the
+// AudioReceiver is destroyed. Clients should generally limit the number of
+// outstanding requests (perhaps to just one or two).
+//
+// This class is not thread safe. Should only be called from the Main cast
// thread.
-class AudioReceiver : public base::NonThreadSafe,
- public base::SupportsWeakPtr<AudioReceiver>,
- public RtpReceiver {
+class AudioReceiver : public RtpReceiver,
+ public RtpPayloadFeedback,
+ public base::NonThreadSafe,
+ public base::SupportsWeakPtr<AudioReceiver> {
public:
AudioReceiver(scoped_refptr<CastEnvironment> cast_environment,
const AudioReceiverConfig& audio_config,
virtual ~AudioReceiver();
- // Extract a raw audio frame from the cast receiver.
- // Actual decoding will be preformed on a designated audio_decoder thread.
- void GetRawAudioFrame(int number_of_10ms_blocks,
- int desired_frequency,
- const AudioFrameDecodedCallback& callback);
-
- // Extract an encoded audio frame from the cast receiver.
+ // Request a decoded audio frame. The audio signal data returned in the
+ // callback will have the sampling rate and number of channels as requested in
+ // the configuration that was passed to the ctor.
+ //
+ // The given |callback| is guaranteed to be run at some point in the future,
+ // even if to respond with NULL at shutdown time.
+ void GetRawAudioFrame(const AudioFrameDecodedCallback& callback);
+
+ // Request an encoded audio frame.
+ //
+ // The given |callback| is guaranteed to be run at some point in the future,
+ // even if to respond with NULL at shutdown time.
void GetEncodedAudioFrame(const AudioFrameEncodedCallback& callback);
- // Should only be called from the main cast thread.
+ // Deliver another packet, possibly a duplicate, and possibly out-of-order.
void IncomingPacket(scoped_ptr<Packet> packet);
// Update target audio delay used to compute the playout time. Rtcp
// will also be updated (will be included in all outgoing reports).
void SetTargetDelay(base::TimeDelta target_delay);
+ protected:
+ friend class AudioReceiverTest; // Invokes OnReceivedPayloadData().
+
virtual void OnReceivedPayloadData(const uint8* payload_data,
size_t payload_size,
const RtpCastHeader& rtp_header) OVERRIDE;
- private:
- friend class LocalRtpAudioFeedback;
-
- void CastFeedback(const RtcpCastMessage& cast_message);
-
- // Time to pull out the audio even though we are missing data.
- void PlayoutTimeout();
-
- bool PostEncodedAudioFrame(
- const AudioFrameEncodedCallback& callback,
- bool next_frame,
- scoped_ptr<transport::EncodedAudioFrame>* encoded_frame);
+ // RtpPayloadFeedback implementation.
+ virtual void CastFeedback(const RtcpCastMessage& cast_message) OVERRIDE;
- // Actual decoding implementation - should be called under the audio decoder
- // thread.
- void DecodeAudioFrameThread(int number_of_10ms_blocks,
- int desired_frequency,
- const AudioFrameDecodedCallback callback);
- void ReturnDecodedFrameWithPlayoutDelay(
- scoped_ptr<PcmAudioFrame> audio_frame,
- uint32 rtp_timestamp,
- const AudioFrameDecodedCallback callback);
+ private:
+ // Processes ready-to-consume packets from |framer_|, decrypting each packet's
+ // payload data, and then running the enqueued callbacks in order (one for
+ // each packet). This method may post a delayed task to re-invoke itself in
+ // the future to wait for missing/incomplete frames.
+ void EmitAvailableEncodedFrames();
+
+ // Clears the |is_waiting_for_consecutive_frame_| flag and invokes
+ // EmitAvailableEncodedFrames().
+ void EmitAvailableEncodedFramesAfterWaiting();
+
+ // Feeds an EncodedAudioFrame into |audio_decoder_|. GetRawAudioFrame() uses
+ // this as a callback for GetEncodedAudioFrame().
+ void DecodeEncodedAudioFrame(
+ const AudioFrameDecodedCallback& callback,
+ scoped_ptr<transport::EncodedAudioFrame> encoded_frame,
+ const base::TimeTicks& playout_time);
// Return the playout time based on the current time and rtp timestamp.
base::TimeTicks GetPlayoutTime(base::TimeTicks now, uint32 rtp_timestamp);
void InitializeTimers();
- // Decrypts the data within the |audio_frame| and replaces the data with the
- // decrypted string.
- bool DecryptAudioFrame(scoped_ptr<transport::EncodedAudioFrame>* audio_frame);
-
// Schedule the next RTCP report.
void ScheduleNextRtcpReport();
// Actually send the next cast message.
void SendNextCastMessage();
- scoped_refptr<CastEnvironment> cast_environment_;
+ // Receives an AudioBus from |audio_decoder_|, logs the event, and passes the
+ // data on by running the given |callback|. This method is static to ensure
+ // it can be called after an AudioReceiver instance is destroyed.
+ // DecodeEncodedAudioFrame() uses this as a callback for
+ // AudioDecoder::DecodeFrame().
+ static void EmitRawAudioFrame(
+ const scoped_refptr<CastEnvironment>& cast_environment,
+ const AudioFrameDecodedCallback& callback,
+ uint32 frame_id,
+ uint32 rtp_timestamp,
+ const base::TimeTicks& playout_time,
+ scoped_ptr<AudioBus> audio_bus,
+ bool is_continuous);
+
+ const scoped_refptr<CastEnvironment> cast_environment_;
// Subscribes to raw events.
// Processes raw audio events to be sent over to the cast sender via RTCP.
const transport::AudioCodec codec_;
const int frequency_;
base::TimeDelta target_delay_delta_;
- scoped_ptr<Framer> audio_buffer_;
+ Framer framer_;
scoped_ptr<AudioDecoder> audio_decoder_;
- scoped_ptr<LocalRtpAudioFeedback> incoming_payload_feedback_;
- scoped_ptr<Rtcp> rtcp_;
+ Rtcp rtcp_;
base::TimeDelta time_offset_;
base::TimeTicks time_first_incoming_packet_;
uint32 first_incoming_rtp_timestamp_;
transport::TransportEncryptionHandler decryptor_;
- base::TimeTicks last_playout_time_;
- std::list<AudioFrameEncodedCallback> queued_encoded_callbacks_;
- std::list<DecodedAudioCallbackData> queued_decoded_callbacks_;
+ // Outstanding callbacks to run to deliver on client requests for frames.
+ std::list<AudioFrameEncodedCallback> frame_request_queue_;
+
+ // True while there's an outstanding task to re-invoke
+ // EmitAvailableEncodedFrames().
+ bool is_waiting_for_consecutive_frame_;
// This mapping allows us to log kAudioAckSent as a frame event. In addition
// it allows the event to be transmitted via RTCP.
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