Revert "Add plumbing of RtpPacketInfos to each AudioFrame as input for SourceTracker."
This reverts commit 3e8ef940fe86cf6285afb80e68d2a0bedc631b9f.
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}
diff --git a/modules/audio_coding/neteq/neteq_impl.h b/modules/audio_coding/neteq/neteq_impl.h
index 9e1af10..39a4df6 100644
--- a/modules/audio_coding/neteq/neteq_impl.h
+++ b/modules/audio_coding/neteq/neteq_impl.h
@@ -11,15 +11,11 @@
#ifndef MODULES_AUDIO_CODING_NETEQ_NETEQ_IMPL_H_
#define MODULES_AUDIO_CODING_NETEQ_NETEQ_IMPL_H_
-#include <map>
#include <memory>
#include <string>
-#include <utility>
-#include <vector>
#include "absl/types/optional.h"
#include "api/audio/audio_frame.h"
-#include "api/rtp_packet_info.h"
#include "modules/audio_coding/neteq/audio_multi_vector.h"
#include "modules/audio_coding/neteq/defines.h" // Modes, Operations
#include "modules/audio_coding/neteq/expand_uma_logger.h"
@@ -38,7 +34,6 @@
class Accelerate;
class BackgroundNoise;
class BufferLevelFilter;
-class Clock;
class ComfortNoise;
class DecisionLogic;
class DecoderDatabase;
@@ -92,13 +87,11 @@
// before sending the struct to the NetEqImpl constructor. However, there
// are dependencies between some of the classes inside the struct, so
// swapping out one may make it necessary to re-create another one.
- Dependencies(
+ explicit Dependencies(
const NetEq::Config& config,
- Clock* clock,
const rtc::scoped_refptr<AudioDecoderFactory>& decoder_factory);
~Dependencies();
- Clock* const clock;
std::unique_ptr<TickTimer> tick_timer;
std::unique_ptr<StatisticsCalculator> stats;
std::unique_ptr<BufferLevelFilter> buffer_level_filter;
@@ -339,8 +332,6 @@
// Creates DecisionLogic object with the mode given by |playout_mode_|.
virtual void CreateDecisionLogic() RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
- Clock* const clock_;
-
rtc::CriticalSection crit_sect_;
const std::unique_ptr<TickTimer> tick_timer_ RTC_GUARDED_BY(crit_sect_);
const std::unique_ptr<BufferLevelFilter> buffer_level_filter_
@@ -406,8 +397,6 @@
std::unique_ptr<TickTimer::Stopwatch> generated_noise_stopwatch_
RTC_GUARDED_BY(crit_sect_);
std::vector<uint32_t> last_decoded_timestamps_ RTC_GUARDED_BY(crit_sect_);
- std::vector<RtpPacketInfo> last_decoded_packet_infos_
- RTC_GUARDED_BY(crit_sect_);
ExpandUmaLogger expand_uma_logger_ RTC_GUARDED_BY(crit_sect_);
ExpandUmaLogger speech_expand_uma_logger_ RTC_GUARDED_BY(crit_sect_);
bool no_time_stretching_ RTC_GUARDED_BY(crit_sect_); // Only used for test.