ACM: Removed the code for InitialDelayManager

webrtc/modules/audio_coding/acm2/initial_delay_manager.cc

Index: webrtc/modules/audio_coding/acm2/initial_delay_manager.cc
diff --git a/webrtc/modules/audio_coding/acm2/initial_delay_manager.cc b/webrtc/modules/audio_coding/acm2/initial_delay_manager.cc
deleted file mode 100644
index 0c31b83eb313006fb42bf8e6fde12925a708ebda..0000000000000000000000000000000000000000
--- a/webrtc/modules/audio_coding/acm2/initial_delay_manager.cc
+++ /dev/null
@@ -1,242 +0,0 @@
-/*
- *  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.
- */
-
-#include "webrtc/modules/audio_coding/acm2/initial_delay_manager.h"
-
-namespace webrtc {
-
-namespace acm2 {
-
-InitialDelayManager::InitialDelayManager(int initial_delay_ms,
-                                         int late_packet_threshold)
-    : last_packet_type_(kUndefinedPacket),
-      last_receive_timestamp_(0),
-      timestamp_step_(0),
-      audio_payload_type_(kInvalidPayloadType),
-      initial_delay_ms_(initial_delay_ms),
-      buffered_audio_ms_(0),
-      buffering_(true),
-      playout_timestamp_(0),
-      late_packet_threshold_(late_packet_threshold) {
-  last_packet_rtp_info_.header.payloadType = kInvalidPayloadType;
-  last_packet_rtp_info_.header.ssrc = 0;
-  last_packet_rtp_info_.header.sequenceNumber = 0;
-  last_packet_rtp_info_.header.timestamp = 0;
-}
-
-void InitialDelayManager::UpdateLastReceivedPacket(
-    const WebRtcRTPHeader& rtp_info,
-    uint32_t receive_timestamp,
-    PacketType type,
-    bool new_codec,
-    int sample_rate_hz,
-    SyncStream* sync_stream) {
-  assert(sync_stream);
-
-  // If payload of audio packets is changing |new_codec| has to be true.
-  assert(!(!new_codec && type == kAudioPacket &&
-         rtp_info.header.payloadType != audio_payload_type_));
-
-  // Just shorthands.
-  const RTPHeader* current_header = &rtp_info.header;
-  RTPHeader* last_header = &last_packet_rtp_info_.header;
-
-  // Don't do anything if getting DTMF. The chance of DTMF in applications where
-  // initial delay is required is very low (we don't know of any). This avoids a
-  // lot of corner cases. The effect of ignoring DTMF packet is minimal. Note
-  // that DTMFs are inserted into NetEq just not accounted here.
-  if (type == kAvtPacket ||
-      (last_packet_type_ != kUndefinedPacket &&
-      !IsNewerSequenceNumber(current_header->sequenceNumber,
-                             last_header->sequenceNumber))) {
-    sync_stream->num_sync_packets = 0;
-    return;
-  }
-
-  // Either if it is a new packet or the first packet record and set variables.
-  if (new_codec ||
-      last_packet_rtp_info_.header.payloadType == kInvalidPayloadType) {
-    timestamp_step_ = 0;
-    if (type == kAudioPacket)
-      audio_payload_type_ = rtp_info.header.payloadType;
-    else
-      audio_payload_type_ = kInvalidPayloadType;  // Invalid.
-
-    RecordLastPacket(rtp_info, receive_timestamp, type);
-    sync_stream->num_sync_packets = 0;
-    buffered_audio_ms_ = 0;
-    buffering_ = true;
-
-    // If |buffering_| is set then |playout_timestamp_| should have correct
-    // value.
-    UpdatePlayoutTimestamp(*current_header, sample_rate_hz);
-    return;
-  }
-
-  uint32_t timestamp_increase = current_header->timestamp -
-      last_header->timestamp;
-
-  // |timestamp_increase| is invalid if this is the first packet. The effect is
-  // that |buffered_audio_ms_| is not increased.
-  if (last_packet_type_ == kUndefinedPacket) {
-    timestamp_increase = 0;
-  }
-
-  if (buffering_) {
-    buffered_audio_ms_ += timestamp_increase * 1000 / sample_rate_hz;
-
-    // A timestamp that reflects the initial delay, while buffering.
-    UpdatePlayoutTimestamp(*current_header, sample_rate_hz);
-
-    if (buffered_audio_ms_ >= initial_delay_ms_)
-      buffering_ = false;
-  }
-
-  if (current_header->sequenceNumber == last_header->sequenceNumber + 1) {
-    // Two consecutive audio packets, the previous packet-type is audio, so we
-    // can update |timestamp_step_|.
-    if (last_packet_type_ == kAudioPacket)
-      timestamp_step_ = timestamp_increase;
-    RecordLastPacket(rtp_info, receive_timestamp, type);
-    sync_stream->num_sync_packets = 0;
-    return;
-  }
-
-  uint16_t packet_gap = current_header->sequenceNumber -
-      last_header->sequenceNumber - 1;
-
-  // For smooth transitions leave a gap between audio and sync packets.
-  sync_stream->num_sync_packets = last_packet_type_ == kSyncPacket ?
-      packet_gap - 1 : packet_gap - 2;
-
-  // Do nothing if we haven't received any audio packet.
-  if (sync_stream->num_sync_packets > 0 &&
-      audio_payload_type_ != kInvalidPayloadType) {
-    if (timestamp_step_ == 0) {
-      // Make an estimate for |timestamp_step_| if it is not updated, yet.
-      assert(packet_gap > 0);
-      timestamp_step_ = timestamp_increase / (packet_gap + 1);
-    }
-    sync_stream->timestamp_step = timestamp_step_;
-
-    // Build the first sync-packet based on the current received packet.
-    memcpy(&sync_stream->rtp_info, &rtp_info, sizeof(rtp_info));
-    sync_stream->rtp_info.header.payloadType = audio_payload_type_;
-
-    uint16_t sequence_number_update = sync_stream->num_sync_packets + 1;
-    uint32_t timestamp_update = timestamp_step_ * sequence_number_update;
-
-    // Rewind sequence number and timestamps. This will give a more accurate
-    // description of the missing packets.
-    //
-    // Note that we leave a gap between the last packet in sync-stream and the
-    // current received packet, so it should be compensated for in the following
-    // computation of timestamps and sequence number.
-    sync_stream->rtp_info.header.sequenceNumber -= sequence_number_update;
-    sync_stream->receive_timestamp = receive_timestamp - timestamp_update;
-    sync_stream->rtp_info.header.timestamp -= timestamp_update;
-    sync_stream->rtp_info.header.payloadType = audio_payload_type_;
-  } else {
-    sync_stream->num_sync_packets = 0;
-  }
-
-  RecordLastPacket(rtp_info, receive_timestamp, type);
-  return;
-}
-
-void InitialDelayManager::RecordLastPacket(const WebRtcRTPHeader& rtp_info,
-                                           uint32_t receive_timestamp,
-                                           PacketType type) {
-  last_packet_type_ = type;
-  last_receive_timestamp_ = receive_timestamp;
-  memcpy(&last_packet_rtp_info_, &rtp_info, sizeof(rtp_info));
-}
-
-void InitialDelayManager::LatePackets(
-    uint32_t timestamp_now, SyncStream* sync_stream) {
-  assert(sync_stream);
-  sync_stream->num_sync_packets = 0;
-
-  // If there is no estimate of timestamp increment, |timestamp_step_|, then
-  // we cannot estimate the number of late packets.
-  // If the last packet has been CNG, estimating late packets is not meaningful,
-  // as a CNG packet is on unknown length.
-  // We can set a higher threshold if the last packet is CNG and continue
-  // execution, but this is how ACM1 code was written.
-  if (timestamp_step_ <= 0 ||
-      last_packet_type_ == kCngPacket ||
-      last_packet_type_ == kUndefinedPacket ||
-      audio_payload_type_ == kInvalidPayloadType)  // No audio packet received.
-    return;
-
-  int num_late_packets = (timestamp_now - last_receive_timestamp_) /
-      timestamp_step_;
-
-  if (num_late_packets < late_packet_threshold_)
-    return;
-
-  int sync_offset = 1;  // One gap at the end of the sync-stream.
-  if (last_packet_type_ != kSyncPacket) {
-    ++sync_offset;  // One more gap at the beginning of the sync-stream.
-    --num_late_packets;
-  }
-  uint32_t timestamp_update = sync_offset * timestamp_step_;
-
-  sync_stream->num_sync_packets = num_late_packets;
-  if (num_late_packets == 0)
-    return;
-
-  // Build the first sync-packet in the sync-stream.
-  memcpy(&sync_stream->rtp_info, &last_packet_rtp_info_,
-         sizeof(last_packet_rtp_info_));
-
-  // Increase sequence number and timestamps.
-  sync_stream->rtp_info.header.sequenceNumber += sync_offset;
-  sync_stream->rtp_info.header.timestamp += timestamp_update;
-  sync_stream->receive_timestamp = last_receive_timestamp_ + timestamp_update;
-  sync_stream->timestamp_step = timestamp_step_;
-
-  // Sync-packets have audio payload-type.
-  sync_stream->rtp_info.header.payloadType = audio_payload_type_;
-
-  uint16_t sequence_number_update = num_late_packets + sync_offset - 1;
-  timestamp_update = sequence_number_update * timestamp_step_;
-
-  // Fake the last RTP, assuming the caller will inject the whole sync-stream.
-  last_packet_rtp_info_.header.timestamp += timestamp_update;
-  last_packet_rtp_info_.header.sequenceNumber += sequence_number_update;
-  last_packet_rtp_info_.header.payloadType = audio_payload_type_;
-  last_receive_timestamp_ += timestamp_update;
-
-  last_packet_type_ = kSyncPacket;
-  return;
-}
-
-bool InitialDelayManager::GetPlayoutTimestamp(uint32_t* playout_timestamp) {
-  if (!buffering_) {
-    return false;
-  }
-  *playout_timestamp = playout_timestamp_;
-  return true;
-}
-
-void InitialDelayManager::DisableBuffering() {
-  buffering_ = false;
-}
-
-void InitialDelayManager::UpdatePlayoutTimestamp(
-    const RTPHeader& current_header, int sample_rate_hz) {
-  playout_timestamp_ = current_header.timestamp - static_cast<uint32_t>(
-      initial_delay_ms_ * sample_rate_hz / 1000);
-}
-
-}  // namespace acm2
-
-}  // namespace webrtc