audio_coding: remove "main" directory

webrtc/modules/audio_coding/main/acm2/initial_delay_manager.h

-/*
- *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
- *
- *  Use of this source code is governed by a BSD-style license
- *  that can be found in the LICENSE file in the root of the source
- *  tree. An additional intellectual property rights grant can be found
- *  in the file PATENTS.  All contributing project authors may
- *  be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_INITIAL_DELAY_MANAGER_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_INITIAL_DELAY_MANAGER_H_
-
-#include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/include/module_common_types.h"
-
-namespace webrtc {
-
-namespace acm2 {
-
-class InitialDelayManager {
- public:
-  enum PacketType {
-    kUndefinedPacket, kCngPacket, kAvtPacket, kAudioPacket, kSyncPacket };
-
-  // Specifies a stream of sync-packets.
-  struct SyncStream {
-    SyncStream()
-        : num_sync_packets(0),
-          receive_timestamp(0),
-          timestamp_step(0) {
-      memset(&rtp_info, 0, sizeof(rtp_info));
-    }
-
-    int num_sync_packets;
-
-    // RTP header of the first sync-packet in the sequence.
-    WebRtcRTPHeader rtp_info;
-
-    // Received timestamp of the first sync-packet in the sequence.
-    uint32_t receive_timestamp;
-
-    // Samples per packet.
-    uint32_t timestamp_step;
-  };
-
-  InitialDelayManager(int initial_delay_ms, int late_packet_threshold);
-
-  // Update with the last received RTP header, |header|, and received timestamp,
-  // |received_timestamp|. |type| indicates the packet type. If codec is changed
-  // since the last time |new_codec| should be true. |sample_rate_hz| is the
-  // decoder's sampling rate in Hz. |header| has a field to store sampling rate
-  // but we are not sure if that is properly set at the send side, and |header|
-  // is declared constant in the caller of this function
-  // (AcmReceiver::InsertPacket()). |sync_stream| contains information required
-  // to generate a stream of sync packets.
-  void UpdateLastReceivedPacket(const WebRtcRTPHeader& header,
-                                uint32_t receive_timestamp,
-                                PacketType type,
-                                bool new_codec,
-                                int sample_rate_hz,
-                                SyncStream* sync_stream);
-
-  // Based on the last received timestamp and given the current timestamp,
-  // sequence of late (or perhaps missing) packets is computed.
-  void LatePackets(uint32_t timestamp_now, SyncStream* sync_stream);
-
-  // Get playout timestamp.
-  // Returns true if the timestamp is valid (when buffering), otherwise false.
-  bool GetPlayoutTimestamp(uint32_t* playout_timestamp);
-
-  // True if buffered audio is less than the given initial delay (specified at
-  // the constructor). Buffering might be disabled by the client of this class.
-  bool buffering() { return buffering_; }
-
-  // Disable buffering in the class.
-  void DisableBuffering();
-
-  // True if any packet received for buffering.
-  bool PacketBuffered() { return last_packet_type_ != kUndefinedPacket; }
-
- private:
-  static const uint8_t kInvalidPayloadType = 0xFF;
-
-  // Update playout timestamps. While buffering, this is about
-  // |initial_delay_ms| millisecond behind the latest received timestamp.
-  void UpdatePlayoutTimestamp(const RTPHeader& current_header,
-                              int sample_rate_hz);
-
-  // Record an RTP headr and related parameter
-  void RecordLastPacket(const WebRtcRTPHeader& rtp_info,
-                        uint32_t receive_timestamp,
-                        PacketType type);
-
-  PacketType last_packet_type_;
-  WebRtcRTPHeader last_packet_rtp_info_;
-  uint32_t last_receive_timestamp_;
-  uint32_t timestamp_step_;
-  uint8_t audio_payload_type_;
-  const int initial_delay_ms_;
-  int buffered_audio_ms_;
-  bool buffering_;
-
-  // During the initial phase where packets are being accumulated and silence
-  // is played out, |playout_ts| is a timestamp which is equal to
-  // |initial_delay_ms_| milliseconds earlier than the most recently received
-  // RTP timestamp.
-  uint32_t playout_timestamp_;
-
-  // If the number of late packets exceed this value (computed based on current
-  // timestamp and last received timestamp), sequence of sync-packets is
-  // specified.
-  const int late_packet_threshold_;
-};
-
-}  // namespace acm2
-
-}  // namespace webrtc
-
-#endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_INITIAL_DELAY_MANAGER_H_