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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

8
modules/audio_coding/neteq/neteq_stereo_unittest.cc
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@ -22,6 +22,7 @@
#include "modules/audio_coding/neteq/tools/input_audio_file.h"
#include "modules/audio_coding/neteq/tools/rtp_generator.h"
#include "rtc_base/strings/string_builder.h"
#include "system_wrappers/include/clock.h"
#include "test/gtest.h"
#include "test/testsupport/file_utils.h"
@ -57,6 +58,7 @@ class NetEqStereoTest : public ::testing::TestWithParam<TestParameters> {
frame_size_samples_(
static_cast<size_t>(frame_size_ms_ * samples_per_ms_)),
output_size_samples_(10 * samples_per_ms_),
clock_(0),
rtp_generator_mono_(samples_per_ms_),
rtp_generator_(samples_per_ms_),
payload_size_bytes_(0),
@ -67,8 +69,8 @@ class NetEqStereoTest : public ::testing::TestWithParam<TestParameters> {
config.sample_rate_hz = sample_rate_hz_;
rtc::scoped_refptr<AudioDecoderFactory> factory =
CreateBuiltinAudioDecoderFactory();
neteq_mono_ = NetEq::Create(config, factory);
neteq_ = NetEq::Create(config, factory);
neteq_mono_ = NetEq::Create(config, &clock_, factory);
neteq_ = NetEq::Create(config, &clock_, factory);
input_ = new int16_t[frame_size_samples_];
encoded_ = new uint8_t[2 * frame_size_samples_];
input_multi_channel_ = new int16_t[frame_size_samples_ * num_channels_];
@ -196,6 +198,7 @@ class NetEqStereoTest : public ::testing::TestWithParam<TestParameters> {
ASSERT_NO_FATAL_FAILURE(VerifyOutput(output_size_samples_));
time_now += kTimeStepMs;
clock_.AdvanceTimeMilliseconds(kTimeStepMs);
}
}
@ -205,6 +208,7 @@ class NetEqStereoTest : public ::testing::TestWithParam<TestParameters> {
const int frame_size_ms_;
const size_t frame_size_samples_;
const size_t output_size_samples_;
SimulatedClock clock_;
NetEq* neteq_mono_;
NetEq* neteq_;
test::RtpGenerator rtp_generator_mono_;