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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
Download Patch File Download Diff File Expand all files Collapse all files

18
modules/audio_coding/neteq/neteq_unittest.cc
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@ -36,6 +36,7 @@
#include "rtc_base/string_encode.h"
#include "rtc_base/strings/string_builder.h"
#include "rtc_base/system/arch.h"
#include "system_wrappers/include/clock.h"
#include "test/field_trial.h"
#include "test/gtest.h"
#include "test/testsupport/file_utils.h"
@ -288,11 +289,11 @@ class NetEqDecodingTest : public ::testing::Test {
void DuplicateCng();
SimulatedClock clock_;
NetEq* neteq_;
NetEq::Config config_;
std::unique_ptr<test::RtpFileSource> rtp_source_;
std::unique_ptr<test::Packet> packet_;
unsigned int sim_clock_;
AudioFrame out_frame_;
int output_sample_rate_;
int algorithmic_delay_ms_;
@ -306,16 +307,16 @@ const size_t NetEqDecodingTest::kBlockSize32kHz;
const int NetEqDecodingTest::kInitSampleRateHz;
NetEqDecodingTest::NetEqDecodingTest()
: neteq_(NULL),
: clock_(0),
neteq_(NULL),
config_(),
sim_clock_(0),
output_sample_rate_(kInitSampleRateHz),
algorithmic_delay_ms_(0) {
config_.sample_rate_hz = kInitSampleRateHz;
}
void NetEqDecodingTest::SetUp() {
neteq_ = NetEq::Create(config_, CreateBuiltinAudioDecoderFactory());
neteq_ = NetEq::Create(config_, &clock_, CreateBuiltinAudioDecoderFactory());
NetEqNetworkStatistics stat;
ASSERT_EQ(0, neteq_->NetworkStatistics(&stat));
algorithmic_delay_ms_ = stat.current_buffer_size_ms;
@ -333,7 +334,7 @@ void NetEqDecodingTest::OpenInputFile(const std::string& rtp_file) {
void NetEqDecodingTest::Process() {
// Check if time to receive.
while (packet_ && sim_clock_ >= packet_->time_ms()) {
while (packet_ && clock_.TimeInMilliseconds() >= packet_->time_ms()) {
if (packet_->payload_length_bytes() > 0) {
#ifndef WEBRTC_CODEC_ISAC
// Ignore payload type 104 (iSAC-swb) if ISAC is not supported.
@ -363,7 +364,7 @@ void NetEqDecodingTest::Process() {
EXPECT_EQ(output_sample_rate_, neteq_->last_output_sample_rate_hz());
// Increase time.
sim_clock_ += kTimeStepMs;
clock_.AdvanceTimeMilliseconds(kTimeStepMs);
}
void NetEqDecodingTest::DecodeAndCompare(
@ -394,7 +395,7 @@ void NetEqDecodingTest::DecodeAndCompare(
output.AddResult(out_frame_.data(), out_frame_.samples_per_channel_));
// Query the network statistics API once per second
if (sim_clock_ % 1000 == 0) {
if (clock_.TimeInMilliseconds() % 1000 == 0) {
// Process NetworkStatistics.
NetEqNetworkStatistics current_network_stats;
ASSERT_EQ(0, neteq_->NetworkStatistics(&current_network_stats));
@ -1435,7 +1436,8 @@ class NetEqDecodingTestTwoInstances : public NetEqDecodingTest {
}
void CreateSecondInstance() {
neteq2_.reset(NetEq::Create(config2_, CreateBuiltinAudioDecoderFactory()));
neteq2_.reset(
NetEq::Create(config2_, &clock_, CreateBuiltinAudioDecoderFactory()));
ASSERT_TRUE(neteq2_);
LoadDecoders(neteq2_.get());
}