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Add plumbing of RtpPacketInfos to each AudioFrame as input for SourceTracker.

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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}
This commit is contained in:
Chen Xing
2019-07-01 17:16:32 +02:00
committed by Commit Bot
parent 62eb89d221
commit 3e8ef940fe
23 changed files with 195 additions and 26 deletions
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13
modules/audio_coding/neteq/neteq_impl.h
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@ -11,11 +11,15 @@
#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"
@ -34,6 +38,7 @@ namespace webrtc {
class Accelerate;
class BackgroundNoise;
class BufferLevelFilter;
class Clock;
class ComfortNoise;
class DecisionLogic;
class DecoderDatabase;
@ -93,11 +98,13 @@ class NetEqImpl : public webrtc::NetEq {
// 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.
explicit Dependencies(
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;
@ -338,6 +345,8 @@ class NetEqImpl : public webrtc::NetEq {
// 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_
@ -403,6 +412,8 @@ class NetEqImpl : public webrtc::NetEq {
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.