Merge changes I7bbf776e,I1b827825

am: fe8b4a6579

* commit 'fe8b4a657979b49e1701bd92f6d5814a99e0b2be': (7237 commits)
  WIP: Changes after merge commit 'cb3f9bd'
  Make the nonlinear beamformer steerable
  Utilize bitrate above codec max to protect video.
  Enable VP9 internal resize by default.
  Filter overlapping RTP header extensions.
  Make VCMEncodedFrameCallback const.
  MediaCodecVideoEncoder: Add number of quality resolution downscales to Encoded callback.
  Remove redudant encoder rate calls.
  Create isolate files for nonparallel tests.
  Register header extensions in RtpRtcpObserver to avoid log spam.
  Make an enum class out of NetEqDecoder, and hide the neteq_decoders_ table
  ACM: Move NACK functionality inside NetEq
  Fix chromium-style warnings in webrtc/sound/.
  Create a 'webrtc_nonparallel_tests' target.
  Update scalability structure data according to updates in the RTP payload profile.
  audio_coding: rename interface -> include
  Rewrote perform_action_on_all_files to be parallell.
  Update reference indices according to updates in the RTP payload profile.
  Disable P2PTransport...TestFailoverControlledSide on Memcheck
  pass clangcl compile options to ignore warnings in gflags.cc
  ...

1 files changed, 242 insertions, 0 deletions

diff --git a/webrtc/modules/audio_coding/main/acm2/initial_delay_manager.cc b/webrtc/modules/audio_coding/main/acm2/initial_delay_manager.cc
new file mode 100644
index 0000000000..786fb2e527
--- /dev/null
+++ b/webrtc/modules/audio_coding/main/acm2/initial_delay_manager.cc
@@ -0,0 +1,242 @@
+/*
+ *  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/main/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