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Unified Diff: webrtc/modules/audio_coding/main/acm2/nack.h

Issue 1410073006: ACM: Move NACK functionality inside NetEq (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: rebase Created 5 years, 2 months ago
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Index: webrtc/modules/audio_coding/main/acm2/nack.h
diff --git a/webrtc/modules/audio_coding/main/acm2/nack.h b/webrtc/modules/audio_coding/main/acm2/nack.h
deleted file mode 100644
index 4b22fa123b04769c8370ea77c587c107e34de78f..0000000000000000000000000000000000000000
--- a/webrtc/modules/audio_coding/main/acm2/nack.h
+++ /dev/null
@@ -1,213 +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.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_NACK_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_NACK_H_
-
-#include <vector>
-#include <map>
-
-#include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/audio_coding/main/include/audio_coding_module_typedefs.h"
-#include "webrtc/test/testsupport/gtest_prod_util.h"
-
-//
-// The Nack class keeps track of the lost packets, an estimate of time-to-play
-// for each packet is also given.
-//
-// Every time a packet is pushed into NetEq, LastReceivedPacket() has to be
-// called to update the NACK list.
-//
-// Every time 10ms audio is pulled from NetEq LastDecodedPacket() should be
-// called, and time-to-play is updated at that moment.
-//
-// If packet N is received, any packet prior to |N - NackThreshold| which is not
-// arrived is considered lost, and should be labeled as "missing" (the size of
-// the list might be limited and older packet eliminated from the list). Packets
-// |N - NackThreshold|, |N - NackThreshold + 1|, ..., |N - 1| are considered
-// "late." A "late" packet with sequence number K is changed to "missing" any
-// time a packet with sequence number newer than |K + NackList| is arrived.
-//
-// The Nack class has to know about the sample rate of the packets to compute
-// time-to-play. So sample rate should be set as soon as the first packet is
-// received. If there is a change in the receive codec (sender changes codec)
-// then Nack should be reset. This is because NetEQ would flush its buffer and
-// re-transmission is meaning less for old packet. Therefore, in that case,
-// after reset the sampling rate has to be updated.
-//
-// Thread Safety
-// =============
-// Please note that this class in not thread safe. The class must be protected
-// if different APIs are called from different threads.
-//
-namespace webrtc {
-
-namespace acm2 {
-
-class Nack {
- public:
- // A limit for the size of the NACK list.
- static const size_t kNackListSizeLimit = 500; // 10 seconds for 20 ms frame
- // packets.
- // Factory method.
- static Nack* Create(int nack_threshold_packets);
-
- ~Nack();
-
- // Set a maximum for the size of the NACK list. If the last received packet
- // has sequence number of N, then NACK list will not contain any element
- // with sequence number earlier than N - |max_nack_list_size|.
- //
- // The largest maximum size is defined by |kNackListSizeLimit|
- int SetMaxNackListSize(size_t max_nack_list_size);
-
- // Set the sampling rate.
- //
- // If associated sampling rate of the received packets is changed, call this
- // function to update sampling rate. Note that if there is any change in
- // received codec then NetEq will flush its buffer and NACK has to be reset.
- // After Reset() is called sampling rate has to be set.
- void UpdateSampleRate(int sample_rate_hz);
-
- // Update the sequence number and the timestamp of the last decoded RTP. This
- // API should be called every time 10 ms audio is pulled from NetEq.
- void UpdateLastDecodedPacket(uint16_t sequence_number, uint32_t timestamp);
-
- // Update the sequence number and the timestamp of the last received RTP. This
- // API should be called every time a packet pushed into ACM.
- void UpdateLastReceivedPacket(uint16_t sequence_number, uint32_t timestamp);
-
- // Get a list of "missing" packets which have expected time-to-play larger
- // than the given round-trip-time (in milliseconds).
- // Note: Late packets are not included.
- std::vector<uint16_t> GetNackList(int64_t round_trip_time_ms) const;
-
- // Reset to default values. The NACK list is cleared.
- // |nack_threshold_packets_| & |max_nack_list_size_| preserve their values.
- void Reset();
-
- private:
- // This test need to access the private method GetNackList().
- FRIEND_TEST_ALL_PREFIXES(NackTest, EstimateTimestampAndTimeToPlay);
-
- struct NackElement {
- NackElement(int64_t initial_time_to_play_ms,
- uint32_t initial_timestamp,
- bool missing)
- : time_to_play_ms(initial_time_to_play_ms),
- estimated_timestamp(initial_timestamp),
- is_missing(missing) {}
-
- // Estimated time (ms) left for this packet to be decoded. This estimate is
- // updated every time jitter buffer decodes a packet.
- int64_t time_to_play_ms;
-
- // A guess about the timestamp of the missing packet, it is used for
- // estimation of |time_to_play_ms|. The estimate might be slightly wrong if
- // there has been frame-size change since the last received packet and the
- // missing packet. However, the risk of this is low, and in case of such
- // errors, there will be a minor misestimation in time-to-play of missing
- // packets. This will have a very minor effect on NACK performance.
- uint32_t estimated_timestamp;
-
- // True if the packet is considered missing. Otherwise indicates packet is
- // late.
- bool is_missing;
- };
-
- class NackListCompare {
- public:
- bool operator() (uint16_t sequence_number_old,
- uint16_t sequence_number_new) const {
- return IsNewerSequenceNumber(sequence_number_new, sequence_number_old);
- }
- };
-
- typedef std::map<uint16_t, NackElement, NackListCompare> NackList;
-
- // Constructor.
- explicit Nack(int nack_threshold_packets);
-
- // This API is used only for testing to assess whether time-to-play is
- // computed correctly.
- NackList GetNackList() const;
-
- // Given the |sequence_number_current_received_rtp| of currently received RTP,
- // recognize packets which are not arrive and add to the list.
- void AddToList(uint16_t sequence_number_current_received_rtp);
-
- // This function subtracts 10 ms of time-to-play for all packets in NACK list.
- // This is called when 10 ms elapsed with no new RTP packet decoded.
- void UpdateEstimatedPlayoutTimeBy10ms();
-
- // Given the |sequence_number_current_received_rtp| and
- // |timestamp_current_received_rtp| of currently received RTP update number
- // of samples per packet.
- void UpdateSamplesPerPacket(uint16_t sequence_number_current_received_rtp,
- uint32_t timestamp_current_received_rtp);
-
- // Given the |sequence_number_current_received_rtp| of currently received RTP
- // update the list. That is; some packets will change from late to missing,
- // some packets are inserted as missing and some inserted as late.
- void UpdateList(uint16_t sequence_number_current_received_rtp);
-
- // Packets which are considered late for too long (according to
- // |nack_threshold_packets_|) are flagged as missing.
- void ChangeFromLateToMissing(uint16_t sequence_number_current_received_rtp);
-
- // Packets which have sequence number older that
- // |sequence_num_last_received_rtp_| - |max_nack_list_size_| are removed
- // from the NACK list.
- void LimitNackListSize();
-
- // Estimate timestamp of a missing packet given its sequence number.
- uint32_t EstimateTimestamp(uint16_t sequence_number);
-
- // Compute time-to-play given a timestamp.
- int64_t TimeToPlay(uint32_t timestamp) const;
-
- // If packet N is arrived, any packet prior to N - |nack_threshold_packets_|
- // which is not arrived is considered missing, and should be in NACK list.
- // Also any packet in the range of N-1 and N - |nack_threshold_packets_|,
- // exclusive, which is not arrived is considered late, and should should be
- // in the list of late packets.
- const int nack_threshold_packets_;
-
- // Valid if a packet is received.
- uint16_t sequence_num_last_received_rtp_;
- uint32_t timestamp_last_received_rtp_;
- bool any_rtp_received_; // If any packet received.
-
- // Valid if a packet is decoded.
- uint16_t sequence_num_last_decoded_rtp_;
- uint32_t timestamp_last_decoded_rtp_;
- bool any_rtp_decoded_; // If any packet decoded.
-
- int sample_rate_khz_; // Sample rate in kHz.
-
- // Number of samples per packet. We update this every time we receive a
- // packet, not only for consecutive packets.
- int samples_per_packet_;
-
- // A list of missing packets to be retransmitted. Components of the list
- // contain the sequence number of missing packets and the estimated time that
- // each pack is going to be played out.
- NackList nack_list_;
-
- // NACK list will not keep track of missing packets prior to
- // |sequence_num_last_received_rtp_| - |max_nack_list_size_|.
- size_t max_nack_list_size_;
-};
-
-} // namespace acm2
-
-} // namespace webrtc
-
-#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_NACK_H_
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