Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(411)

Unified Diff: webrtc/modules/audio_coding/main/acm2/initial_delay_manager.cc

Issue 1481493004: audio_coding: remove "main" directory (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Rebased Created 5 years, 1 month ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View side-by-side diff with in-line comments
Download patch
Index: webrtc/modules/audio_coding/main/acm2/initial_delay_manager.cc
diff --git a/webrtc/modules/audio_coding/main/acm2/initial_delay_manager.cc b/webrtc/modules/audio_coding/main/acm2/initial_delay_manager.cc
deleted file mode 100644
index 786fb2e5275140bab7bf131dbf653cc32eef6280..0000000000000000000000000000000000000000
--- a/webrtc/modules/audio_coding/main/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/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

Powered by Google App Engine
This is Rietveld 408576698