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      1 /*
      2  *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
      3  *
      4  *  Use of this source code is governed by a BSD-style license
      5  *  that can be found in the LICENSE file in the root of the source
      6  *  tree. An additional intellectual property rights grant can be found
      7  *  in the file PATENTS.  All contributing project authors may
      8  *  be found in the AUTHORS file in the root of the source tree.
      9  */
     10 
     11 #ifndef WEBRTC_MODULES_AUDIO_CODING_ACM2_INITIAL_DELAY_MANAGER_H_
     12 #define WEBRTC_MODULES_AUDIO_CODING_ACM2_INITIAL_DELAY_MANAGER_H_
     13 
     14 #include "webrtc/base/scoped_ptr.h"
     15 #include "webrtc/modules/include/module_common_types.h"
     16 
     17 namespace webrtc {
     18 
     19 namespace acm2 {
     20 
     21 class InitialDelayManager {
     22  public:
     23   enum PacketType {
     24     kUndefinedPacket, kCngPacket, kAvtPacket, kAudioPacket, kSyncPacket };
     25 
     26   // Specifies a stream of sync-packets.
     27   struct SyncStream {
     28     SyncStream()
     29         : num_sync_packets(0),
     30           receive_timestamp(0),
     31           timestamp_step(0) {
     32       memset(&rtp_info, 0, sizeof(rtp_info));
     33     }
     34 
     35     int num_sync_packets;
     36 
     37     // RTP header of the first sync-packet in the sequence.
     38     WebRtcRTPHeader rtp_info;
     39 
     40     // Received timestamp of the first sync-packet in the sequence.
     41     uint32_t receive_timestamp;
     42 
     43     // Samples per packet.
     44     uint32_t timestamp_step;
     45   };
     46 
     47   InitialDelayManager(int initial_delay_ms, int late_packet_threshold);
     48 
     49   // Update with the last received RTP header, |header|, and received timestamp,
     50   // |received_timestamp|. |type| indicates the packet type. If codec is changed
     51   // since the last time |new_codec| should be true. |sample_rate_hz| is the
     52   // decoder's sampling rate in Hz. |header| has a field to store sampling rate
     53   // but we are not sure if that is properly set at the send side, and |header|
     54   // is declared constant in the caller of this function
     55   // (AcmReceiver::InsertPacket()). |sync_stream| contains information required
     56   // to generate a stream of sync packets.
     57   void UpdateLastReceivedPacket(const WebRtcRTPHeader& header,
     58                                 uint32_t receive_timestamp,
     59                                 PacketType type,
     60                                 bool new_codec,
     61                                 int sample_rate_hz,
     62                                 SyncStream* sync_stream);
     63 
     64   // Based on the last received timestamp and given the current timestamp,
     65   // sequence of late (or perhaps missing) packets is computed.
     66   void LatePackets(uint32_t timestamp_now, SyncStream* sync_stream);
     67 
     68   // Get playout timestamp.
     69   // Returns true if the timestamp is valid (when buffering), otherwise false.
     70   bool GetPlayoutTimestamp(uint32_t* playout_timestamp);
     71 
     72   // True if buffered audio is less than the given initial delay (specified at
     73   // the constructor). Buffering might be disabled by the client of this class.
     74   bool buffering() { return buffering_; }
     75 
     76   // Disable buffering in the class.
     77   void DisableBuffering();
     78 
     79   // True if any packet received for buffering.
     80   bool PacketBuffered() { return last_packet_type_ != kUndefinedPacket; }
     81 
     82  private:
     83   static const uint8_t kInvalidPayloadType = 0xFF;
     84 
     85   // Update playout timestamps. While buffering, this is about
     86   // |initial_delay_ms| millisecond behind the latest received timestamp.
     87   void UpdatePlayoutTimestamp(const RTPHeader& current_header,
     88                               int sample_rate_hz);
     89 
     90   // Record an RTP headr and related parameter
     91   void RecordLastPacket(const WebRtcRTPHeader& rtp_info,
     92                         uint32_t receive_timestamp,
     93                         PacketType type);
     94 
     95   PacketType last_packet_type_;
     96   WebRtcRTPHeader last_packet_rtp_info_;
     97   uint32_t last_receive_timestamp_;
     98   uint32_t timestamp_step_;
     99   uint8_t audio_payload_type_;
    100   const int initial_delay_ms_;
    101   int buffered_audio_ms_;
    102   bool buffering_;
    103 
    104   // During the initial phase where packets are being accumulated and silence
    105   // is played out, |playout_ts| is a timestamp which is equal to
    106   // |initial_delay_ms_| milliseconds earlier than the most recently received
    107   // RTP timestamp.
    108   uint32_t playout_timestamp_;
    109 
    110   // If the number of late packets exceed this value (computed based on current
    111   // timestamp and last received timestamp), sequence of sync-packets is
    112   // specified.
    113   const int late_packet_threshold_;
    114 };
    115 
    116 }  // namespace acm2
    117 
    118 }  // namespace webrtc
    119 
    120 #endif  // WEBRTC_MODULES_AUDIO_CODING_ACM2_INITIAL_DELAY_MANAGER_H_
    121