webrtc/modules/audio_coding/neteq/red_payload_splitter.cc
Alessio Bazzica 8f319a3472 Reland "Reland "Add plumbing of RtpPacketInfos to each AudioFrame as input for SourceTracker.""
This reverts commit fab3460a82.

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 commit 9973933d2e.
> 
> 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 commit 24192c267a.
> > 
> > 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. Commit 1796a820f6 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 commit 3e8ef940fe.
> > >
> > > 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}
2019-07-24 16:47:13 +00:00

175 lines
6.9 KiB
C++

/*
* Copyright (c) 2012 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 "modules/audio_coding/neteq/red_payload_splitter.h"
#include <assert.h>
#include <stddef.h>
#include <cstdint>
#include <list>
#include <utility>
#include <vector>
#include "modules/audio_coding/neteq/decoder_database.h"
#include "modules/audio_coding/neteq/packet.h"
#include "rtc_base/buffer.h"
#include "rtc_base/logging.h"
#include "rtc_base/numerics/safe_conversions.h"
namespace webrtc {
// The method loops through a list of packets {A, B, C, ...}. Each packet is
// split into its corresponding RED payloads, {A1, A2, ...}, which is
// temporarily held in the list |new_packets|.
// When the first packet in |packet_list| has been processed, the orignal packet
// is replaced by the new ones in |new_packets|, so that |packet_list| becomes:
// {A1, A2, ..., B, C, ...}. The method then continues with B, and C, until all
// the original packets have been replaced by their split payloads.
bool RedPayloadSplitter::SplitRed(PacketList* packet_list) {
// Too many RED blocks indicates that something is wrong. Clamp it at some
// reasonable value.
const size_t kMaxRedBlocks = 32;
bool ret = true;
PacketList::iterator it = packet_list->begin();
while (it != packet_list->end()) {
const Packet& red_packet = *it;
assert(!red_packet.payload.empty());
const uint8_t* payload_ptr = red_packet.payload.data();
// Read RED headers (according to RFC 2198):
//
// 0 1 2 3
// 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
// |F| block PT | timestamp offset | block length |
// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
// Last RED header:
// 0 1 2 3 4 5 6 7
// +-+-+-+-+-+-+-+-+
// |0| Block PT |
// +-+-+-+-+-+-+-+-+
struct RedHeader {
uint8_t payload_type;
uint32_t timestamp;
size_t payload_length;
};
std::vector<RedHeader> new_headers;
bool last_block = false;
size_t sum_length = 0;
while (!last_block) {
RedHeader new_header;
// Check the F bit. If F == 0, this was the last block.
last_block = ((*payload_ptr & 0x80) == 0);
// Bits 1 through 7 are payload type.
new_header.payload_type = payload_ptr[0] & 0x7F;
if (last_block) {
// No more header data to read.
++sum_length; // Account for RED header size of 1 byte.
new_header.timestamp = red_packet.timestamp;
new_header.payload_length = red_packet.payload.size() - sum_length;
payload_ptr += 1; // Advance to first payload byte.
} else {
// Bits 8 through 21 are timestamp offset.
int timestamp_offset =
(payload_ptr[1] << 6) + ((payload_ptr[2] & 0xFC) >> 2);
new_header.timestamp = red_packet.timestamp - timestamp_offset;
// Bits 22 through 31 are payload length.
new_header.payload_length =
((payload_ptr[2] & 0x03) << 8) + payload_ptr[3];
payload_ptr += 4; // Advance to next RED header.
}
sum_length += new_header.payload_length;
sum_length += 4; // Account for RED header size of 4 bytes.
// Store in new list of packets.
new_headers.push_back(new_header);
}
if (new_headers.size() <= kMaxRedBlocks) {
// Populate the new packets with payload data.
// |payload_ptr| now points at the first payload byte.
PacketList new_packets; // An empty list to store the split packets in.
for (size_t i = 0; i != new_headers.size(); ++i) {
const auto& new_header = new_headers[i];
size_t payload_length = new_header.payload_length;
if (payload_ptr + payload_length >
red_packet.payload.data() + red_packet.payload.size()) {
// The block lengths in the RED headers do not match the overall
// packet length. Something is corrupt. Discard this and the remaining
// payloads from this packet.
RTC_LOG(LS_WARNING) << "SplitRed length mismatch";
ret = false;
break;
}
Packet new_packet;
new_packet.timestamp = new_header.timestamp;
new_packet.payload_type = new_header.payload_type;
new_packet.sequence_number = red_packet.sequence_number;
new_packet.priority.red_level =
rtc::dchecked_cast<int>((new_headers.size() - 1) - i);
new_packet.payload.SetData(payload_ptr, payload_length);
new_packet.packet_info = RtpPacketInfo(
/*ssrc=*/red_packet.packet_info.ssrc(),
/*csrcs=*/std::vector<uint32_t>(),
/*rtp_timestamp=*/new_packet.timestamp,
/*audio_level=*/absl::nullopt,
/*receive_time_ms=*/red_packet.packet_info.receive_time_ms());
new_packets.push_front(std::move(new_packet));
payload_ptr += payload_length;
}
// Insert new packets into original list, before the element pointed to by
// iterator |it|.
packet_list->splice(it, std::move(new_packets));
} else {
RTC_LOG(LS_WARNING) << "SplitRed too many blocks: " << new_headers.size();
ret = false;
}
// Remove |it| from the packet list. This operation effectively moves the
// iterator |it| to the next packet in the list. Thus, we do not have to
// increment it manually.
it = packet_list->erase(it);
}
return ret;
}
void RedPayloadSplitter::CheckRedPayloads(
PacketList* packet_list,
const DecoderDatabase& decoder_database) {
int main_payload_type = -1;
for (auto it = packet_list->begin(); it != packet_list->end(); /* */) {
uint8_t this_payload_type = it->payload_type;
if (decoder_database.IsRed(this_payload_type)) {
it = packet_list->erase(it);
continue;
}
if (!decoder_database.IsDtmf(this_payload_type) &&
!decoder_database.IsComfortNoise(this_payload_type)) {
if (main_payload_type == -1) {
// This is the first packet in the list which is non-DTMF non-CNG.
main_payload_type = this_payload_type;
} else {
if (this_payload_type != main_payload_type) {
// We do not allow redundant payloads of a different type.
// Remove |it| from the packet list. This operation effectively
// moves the iterator |it| to the next packet in the list. Thus, we
// do not have to increment it manually.
it = packet_list->erase(it);
continue;
}
}
}
++it;
}
}
} // namespace webrtc