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This reverts commitfab3460a82
. Reason for revert: fix downstream instead Original change's description: > Revert "Reland "Add plumbing of RtpPacketInfos to each AudioFrame as input for SourceTracker."" > > This reverts commit9973933d2e
. > > Reason for revert: breaking downstream projects and not reviewed by direct owners > > Original change's description: > > Reland "Add plumbing of RtpPacketInfos to each AudioFrame as input for SourceTracker." > > > > This reverts commit24192c267a
. > > > > Reason for revert: Analyzed the performance regression in more detail. > > > > Most of the regression comes from the extra RtpPacketInfos-related memory allocations in every `NetEq::GetAudio()` call. Commit1796a820f6
has removed roughly 2/3rds of the extra allocations from the impacted perf tests. Remaining perf impact is expected to be about "8 microseconds of CPU time per second" on the Linux benchmarking machines and "15 us per second" on Windows/Mac. > > > > There are options to optimize further but they are unlikely worth doing. Note for example that `NetEqPerformanceTest` uses the PCM codec while the real-world use cases would likely use the much heavier Opus codec. The numbers from `OpusSpeedTest` and `NetEqPerformanceTest` suggest that Opus decoding is about 10x as expensive as NetEq overall. > > > > Original change's description: > > > Revert "Add plumbing of RtpPacketInfos to each AudioFrame as input for SourceTracker." > > > > > > This reverts commit3e8ef940fe
. > > > > > > Reason for revert: This CL causes a performance regression in NetEq, see https://bugs.chromium.org/p/chromium/issues/detail?id=982260. > > > > > > Original change's description: > > > > Add plumbing of RtpPacketInfos to each AudioFrame as input for SourceTracker. > > > > > > > > This change adds the plumbing of RtpPacketInfo from ChannelReceive::OnRtpPacket() to ChannelReceive::GetAudioFrameWithInfo() for audio. It is a step towards replacing the non-spec compliant ContributingSources that updates itself at packet-receive time, with the spec-compliant SourceTracker that will update itself at frame-delivery-to-track time. > > > > > > > > Bug: webrtc:10668 > > > > Change-Id: I03385d6865bbc7bfbef7634f88de820a934f787a > > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/139890 > > > > Reviewed-by: Stefan Holmer <stefan@webrtc.org> > > > > Reviewed-by: Minyue Li <minyue@webrtc.org> > > > > Commit-Queue: Chen Xing <chxg@google.com> > > > > Cr-Commit-Position: refs/heads/master@{#28434} > > > > > > TBR=kwiberg@webrtc.org,stefan@webrtc.org,minyue@webrtc.org,chxg@google.com > > > > > > Bug: webrtc:10668, chromium:982260 > > > Change-Id: I5e2cfde78c59d1123e21869564d76ed3f6193a5c > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145339 > > > Reviewed-by: Ivo Creusen <ivoc@webrtc.org> > > > Commit-Queue: Ivo Creusen <ivoc@webrtc.org> > > > Cr-Commit-Position: refs/heads/master@{#28561} > > > > TBR=kwiberg@webrtc.org,stefan@webrtc.org,ivoc@webrtc.org,minyue@webrtc.org,chxg@google.com > > > > # Not skipping CQ checks because original CL landed > 1 day ago. > > > > Bug: webrtc:10668, chromium:982260 > > Change-Id: Ie375a0b327ee368317bf3a04b2f1415c3a974470 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/146707 > > Reviewed-by: Stefan Holmer <stefan@webrtc.org> > > Commit-Queue: Chen Xing <chxg@google.com> > > Cr-Commit-Position: refs/heads/master@{#28664} > > TBR=kwiberg@webrtc.org,stefan@webrtc.org,ivoc@webrtc.org,minyue@webrtc.org,chxg@google.com > > Change-Id: I652cb0814d83b514d3bee34e65ca3bb693099b22 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:10668, chromium:982260 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/146712 > Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> > Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#28671} TBR=alessiob@webrtc.org,kwiberg@webrtc.org,stefan@webrtc.org,ivoc@webrtc.org,minyue@webrtc.org,chxg@google.com Change-Id: Id43b7b3da79b4f48004b41767482bae1c1fa1e16 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:10668, chromium:982260 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/146713 Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28672}
128 lines
4.6 KiB
C++
128 lines
4.6 KiB
C++
/*
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* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef MODULES_AUDIO_CODING_NETEQ_PACKET_H_
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#define MODULES_AUDIO_CODING_NETEQ_PACKET_H_
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#include <stdint.h>
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#include <list>
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#include <memory>
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#include "api/audio_codecs/audio_decoder.h"
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#include "api/rtp_packet_info.h"
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#include "modules/audio_coding/neteq/tick_timer.h"
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#include "rtc_base/buffer.h"
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#include "rtc_base/checks.h"
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namespace webrtc {
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// Struct for holding RTP packets.
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struct Packet {
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struct Priority {
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Priority() : codec_level(0), red_level(0) {}
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Priority(int codec_level, int red_level)
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: codec_level(codec_level), red_level(red_level) {
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CheckInvariant();
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}
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int codec_level;
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int red_level;
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// Priorities are sorted low-to-high, first on the level the codec
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// prioritizes it, then on the level of RED packet it is; i.e. if it is a
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// primary or secondary payload of a RED packet. For example: with Opus, an
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// Fec packet (which the decoder prioritizes lower than a regular packet)
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// will not be used if there is _any_ RED payload for the same
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// timeframe. The highest priority packet will have levels {0, 0}. Negative
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// priorities are not allowed.
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bool operator<(const Priority& b) const {
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CheckInvariant();
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b.CheckInvariant();
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if (codec_level == b.codec_level)
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return red_level < b.red_level;
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return codec_level < b.codec_level;
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}
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bool operator==(const Priority& b) const {
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CheckInvariant();
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b.CheckInvariant();
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return codec_level == b.codec_level && red_level == b.red_level;
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}
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bool operator!=(const Priority& b) const { return !(*this == b); }
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bool operator>(const Priority& b) const { return b < *this; }
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bool operator<=(const Priority& b) const { return !(b > *this); }
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bool operator>=(const Priority& b) const { return !(b < *this); }
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private:
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void CheckInvariant() const {
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RTC_DCHECK_GE(codec_level, 0);
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RTC_DCHECK_GE(red_level, 0);
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}
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};
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uint32_t timestamp;
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uint16_t sequence_number;
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uint8_t payload_type;
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// Datagram excluding RTP header and header extension.
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rtc::Buffer payload;
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Priority priority;
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RtpPacketInfo packet_info;
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std::unique_ptr<TickTimer::Stopwatch> waiting_time;
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std::unique_ptr<AudioDecoder::EncodedAudioFrame> frame;
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Packet();
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Packet(Packet&& b);
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~Packet();
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// Packets should generally be moved around but sometimes it's useful to make
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// a copy, for example for testing purposes. NOTE: Will only work for
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// un-parsed packets, i.e. |frame| must be unset. The payload will, however,
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// be copied. |waiting_time| will also not be copied.
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Packet Clone() const;
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Packet& operator=(Packet&& b);
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// Comparison operators. Establish a packet ordering based on (1) timestamp,
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// (2) sequence number and (3) redundancy.
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// Timestamp and sequence numbers are compared taking wrap-around into
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// account. For two packets with the same sequence number and timestamp a
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// primary payload is considered "smaller" than a secondary.
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bool operator==(const Packet& rhs) const {
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return (this->timestamp == rhs.timestamp &&
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this->sequence_number == rhs.sequence_number &&
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this->priority == rhs.priority);
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}
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bool operator!=(const Packet& rhs) const { return !operator==(rhs); }
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bool operator<(const Packet& rhs) const {
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if (this->timestamp == rhs.timestamp) {
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if (this->sequence_number == rhs.sequence_number) {
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// Timestamp and sequence numbers are identical - deem the left hand
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// side to be "smaller" (i.e., "earlier") if it has higher priority.
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return this->priority < rhs.priority;
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}
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return (static_cast<uint16_t>(rhs.sequence_number -
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this->sequence_number) < 0xFFFF / 2);
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}
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return (static_cast<uint32_t>(rhs.timestamp - this->timestamp) <
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0xFFFFFFFF / 2);
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}
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bool operator>(const Packet& rhs) const { return rhs.operator<(*this); }
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bool operator<=(const Packet& rhs) const { return !operator>(rhs); }
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bool operator>=(const Packet& rhs) const { return !operator<(rhs); }
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bool empty() const { return !frame && payload.empty(); }
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};
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// A list of packets.
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typedef std::list<Packet> PacketList;
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} // namespace webrtc
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#endif // MODULES_AUDIO_CODING_NETEQ_PACKET_H_
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