Migrate video tests and tool to VideoRtpDepacketizer interface
Bug: webrtc:11152
Change-Id: I1e7868ca88b162db8615cb4903bd89d3daac4827
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161452
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30085}
diff --git a/video/BUILD.gn b/video/BUILD.gn
index cf9ce46..0f9829e 100644
--- a/video/BUILD.gn
+++ b/video/BUILD.gn
@@ -286,6 +286,7 @@
"../modules/audio_device:windows_core_audio_utility",
"../modules/audio_mixer:audio_mixer_impl",
"../modules/rtp_rtcp",
+ "../modules/rtp_rtcp:rtp_rtcp_format",
"../modules/video_coding",
"../modules/video_coding:video_coding_utility",
"../modules/video_coding:webrtc_h264",
diff --git a/video/picture_id_tests.cc b/video/picture_id_tests.cc
index da952f7..19c1141 100644
--- a/video/picture_id_tests.cc
+++ b/video/picture_id_tests.cc
@@ -16,7 +16,7 @@
#include "call/simulated_network.h"
#include "media/engine/internal_encoder_factory.h"
#include "media/engine/simulcast_encoder_adapter.h"
-#include "modules/rtp_rtcp/source/rtp_format.h"
+#include "modules/rtp_rtcp/source/create_video_rtp_depacketizer.h"
#include "modules/rtp_rtcp/source/rtp_packet.h"
#include "modules/video_coding/codecs/vp8/include/vp8.h"
#include "modules/video_coding/codecs/vp9/include/vp9.h"
@@ -43,7 +43,7 @@
public:
explicit PictureIdObserver(VideoCodecType codec_type)
: test::RtpRtcpObserver(test::CallTest::kDefaultTimeoutMs),
- codec_type_(codec_type),
+ depacketizer_(CreateVideoRtpDepacketizer(codec_type)),
max_expected_picture_id_gap_(0),
max_expected_tl0_idx_gap_(0),
num_ssrcs_to_observe_(1) {}
@@ -88,43 +88,32 @@
rtp_packet.Ssrc() == test::CallTest::kVideoSendSsrcs[2])
<< "Unknown SSRC sent.";
- rtc::ArrayView<const uint8_t> rtp_payload = rtp_packet.payload();
- if (rtp_payload.empty()) {
+ if (rtp_packet.payload_size() == 0) {
return false; // Padding packet.
}
parsed->timestamp = rtp_packet.Timestamp();
parsed->ssrc = rtp_packet.Ssrc();
- std::unique_ptr<RtpDepacketizer> depacketizer(
- RtpDepacketizer::Create(codec_type_));
- RtpDepacketizer::ParsedPayload parsed_payload;
- EXPECT_TRUE(depacketizer->Parse(&parsed_payload, rtp_payload.data(),
- rtp_payload.size()));
+ absl::optional<VideoRtpDepacketizer::ParsedRtpPayload> parsed_payload =
+ depacketizer_->Parse(rtp_packet.PayloadBuffer());
+ EXPECT_TRUE(parsed_payload);
- switch (codec_type_) {
- case kVideoCodecVP8: {
- const auto& vp8_header = absl::get<RTPVideoHeaderVP8>(
- parsed_payload.video_header().video_type_header);
- parsed->picture_id = vp8_header.pictureId;
- parsed->tl0_pic_idx = vp8_header.tl0PicIdx;
- parsed->temporal_idx = vp8_header.temporalIdx;
- break;
- }
- case kVideoCodecVP9: {
- const auto& vp9_header = absl::get<RTPVideoHeaderVP9>(
- parsed_payload.video_header().video_type_header);
- parsed->picture_id = vp9_header.picture_id;
- parsed->tl0_pic_idx = vp9_header.tl0_pic_idx;
- parsed->temporal_idx = vp9_header.temporal_idx;
- break;
- }
- default:
- RTC_NOTREACHED();
- break;
+ if (const auto* vp8_header = absl::get_if<RTPVideoHeaderVP8>(
+ &parsed_payload->video_header.video_type_header)) {
+ parsed->picture_id = vp8_header->pictureId;
+ parsed->tl0_pic_idx = vp8_header->tl0PicIdx;
+ parsed->temporal_idx = vp8_header->temporalIdx;
+ } else if (const auto* vp9_header = absl::get_if<RTPVideoHeaderVP9>(
+ &parsed_payload->video_header.video_type_header)) {
+ parsed->picture_id = vp9_header->picture_id;
+ parsed->tl0_pic_idx = vp9_header->tl0_pic_idx;
+ parsed->temporal_idx = vp9_header->temporal_idx;
+ } else {
+ RTC_NOTREACHED();
}
- parsed->frame_type = parsed_payload.video_header().frame_type;
+ parsed->frame_type = parsed_payload->video_header.frame_type;
return true;
}
@@ -208,7 +197,7 @@
}
rtc::CriticalSection crit_;
- const VideoCodecType codec_type_;
+ const std::unique_ptr<VideoRtpDepacketizer> depacketizer_;
std::map<uint32_t, ParsedPacket> last_observed_packet_ RTC_GUARDED_BY(crit_);
std::map<uint32_t, size_t> num_packets_sent_ RTC_GUARDED_BY(crit_);
int max_expected_picture_id_gap_ RTC_GUARDED_BY(crit_);
diff --git a/video/video_analyzer.cc b/video/video_analyzer.cc
index 018ec8b..f2822ef 100644
--- a/video/video_analyzer.cc
+++ b/video/video_analyzer.cc
@@ -16,8 +16,8 @@
#include "absl/flags/flag.h"
#include "absl/flags/parse.h"
#include "common_video/libyuv/include/webrtc_libyuv.h"
-#include "modules/rtp_rtcp/source/rtp_format.h"
-#include "modules/rtp_rtcp/source/rtp_utility.h"
+#include "modules/rtp_rtcp/source/create_video_rtp_depacketizer.h"
+#include "modules/rtp_rtcp/source/rtp_packet.h"
#include "rtc_base/cpu_time.h"
#include "rtc_base/format_macros.h"
#include "rtc_base/memory_usage.h"
@@ -109,6 +109,8 @@
is_quick_test_enabled_(is_quick_test_enabled),
quit_(false),
done_(true, false),
+ vp8_depacketizer_(CreateVideoRtpDepacketizer(kVideoCodecVP8)),
+ vp9_depacketizer_(CreateVideoRtpDepacketizer(kVideoCodecVP9)),
clock_(clock),
start_ms_(clock->TimeInMilliseconds()),
task_queue_(task_queue) {
@@ -221,18 +223,18 @@
rtp_file_writer_->WritePacket(&p);
}
- RtpUtility::RtpHeaderParser parser(packet.cdata(), packet.size());
- RTPHeader header;
- parser.Parse(&header);
- if (!IsFlexfec(header.payloadType) && (header.ssrc == ssrc_to_analyze_ ||
- header.ssrc == rtx_ssrc_to_analyze_)) {
+ RtpPacket rtp_packet;
+ rtp_packet.Parse(packet);
+ if (!IsFlexfec(rtp_packet.PayloadType()) &&
+ (rtp_packet.Ssrc() == ssrc_to_analyze_ ||
+ rtp_packet.Ssrc() == rtx_ssrc_to_analyze_)) {
// Ignore FlexFEC timestamps, to avoid collisions with media timestamps.
// (FlexFEC and media are sent on different SSRCs, which have different
// timestamps spaces.)
// Also ignore packets from wrong SSRC, but include retransmits.
rtc::CritScope lock(&crit_);
int64_t timestamp =
- wrap_handler_.Unwrap(header.timestamp - rtp_timestamp_delta_);
+ wrap_handler_.Unwrap(rtp_packet.Timestamp() - rtp_timestamp_delta_);
recv_times_[timestamp] = clock_->CurrentNtpInMilliseconds();
}
@@ -262,32 +264,31 @@
bool VideoAnalyzer::SendRtp(const uint8_t* packet,
size_t length,
const PacketOptions& options) {
- RtpUtility::RtpHeaderParser parser(packet, length);
- RTPHeader header;
- parser.Parse(&header);
+ RtpPacket rtp_packet;
+ rtp_packet.Parse(packet, length);
int64_t current_time = clock_->CurrentNtpInMilliseconds();
bool result = transport_->SendRtp(packet, length, options);
{
rtc::CritScope lock(&crit_);
- if (rtp_timestamp_delta_ == 0 && header.ssrc == ssrc_to_analyze_) {
+ if (rtp_timestamp_delta_ == 0 && rtp_packet.Ssrc() == ssrc_to_analyze_) {
RTC_CHECK(static_cast<bool>(first_sent_timestamp_));
- rtp_timestamp_delta_ = header.timestamp - *first_sent_timestamp_;
+ rtp_timestamp_delta_ = rtp_packet.Timestamp() - *first_sent_timestamp_;
}
- if (!IsFlexfec(header.payloadType) && header.ssrc == ssrc_to_analyze_) {
+ if (!IsFlexfec(rtp_packet.PayloadType()) &&
+ rtp_packet.Ssrc() == ssrc_to_analyze_) {
// Ignore FlexFEC timestamps, to avoid collisions with media timestamps.
// (FlexFEC and media are sent on different SSRCs, which have different
// timestamps spaces.)
// Also ignore packets from wrong SSRC and retransmits.
int64_t timestamp =
- wrap_handler_.Unwrap(header.timestamp - rtp_timestamp_delta_);
+ wrap_handler_.Unwrap(rtp_packet.Timestamp() - rtp_timestamp_delta_);
send_times_[timestamp] = current_time;
- if (IsInSelectedSpatialAndTemporalLayer(packet, length, header)) {
- encoded_frame_sizes_[timestamp] +=
- length - (header.headerLength + header.paddingLength);
+ if (IsInSelectedSpatialAndTemporalLayer(rtp_packet)) {
+ encoded_frame_sizes_[timestamp] += rtp_packet.payload_size();
}
}
}
@@ -428,44 +429,31 @@
}
bool VideoAnalyzer::IsInSelectedSpatialAndTemporalLayer(
- const uint8_t* packet,
- size_t length,
- const RTPHeader& header) {
- if (header.payloadType != test::CallTest::kPayloadTypeVP9 &&
- header.payloadType != test::CallTest::kPayloadTypeVP8) {
- return true;
- } else {
- // Get VP8 and VP9 specific header to check layers indexes.
- const uint8_t* payload = packet + header.headerLength;
- const size_t payload_length = length - header.headerLength;
- const size_t payload_data_length = payload_length - header.paddingLength;
- const bool is_vp8 = header.payloadType == test::CallTest::kPayloadTypeVP8;
- std::unique_ptr<RtpDepacketizer> depacketizer(
- RtpDepacketizer::Create(is_vp8 ? kVideoCodecVP8 : kVideoCodecVP9));
- RtpDepacketizer::ParsedPayload parsed_payload;
- bool result =
- depacketizer->Parse(&parsed_payload, payload, payload_data_length);
- RTC_DCHECK(result);
+ const RtpPacket& rtp_packet) {
+ if (rtp_packet.PayloadType() == test::CallTest::kPayloadTypeVP8) {
+ auto parsed_payload = vp8_depacketizer_->Parse(rtp_packet.PayloadBuffer());
+ RTC_DCHECK(parsed_payload);
+ const auto& vp8_header = absl::get<RTPVideoHeaderVP8>(
+ parsed_payload->video_header.video_type_header);
+ int temporal_idx = vp8_header.temporalIdx;
+ return selected_tl_ < 0 || temporal_idx == kNoTemporalIdx ||
+ temporal_idx <= selected_tl_;
+ }
- int temporal_idx;
- int spatial_idx;
- if (is_vp8) {
- temporal_idx = absl::get<RTPVideoHeaderVP8>(
- parsed_payload.video_header().video_type_header)
- .temporalIdx;
- spatial_idx = kNoTemporalIdx;
- } else {
- const auto& vp9_header = absl::get<RTPVideoHeaderVP9>(
- parsed_payload.video_header().video_type_header);
- temporal_idx = vp9_header.temporal_idx;
- spatial_idx = vp9_header.spatial_idx;
- }
-
+ if (rtp_packet.PayloadType() == test::CallTest::kPayloadTypeVP9) {
+ auto parsed_payload = vp9_depacketizer_->Parse(rtp_packet.PayloadBuffer());
+ RTC_DCHECK(parsed_payload);
+ const auto& vp9_header = absl::get<RTPVideoHeaderVP9>(
+ parsed_payload->video_header.video_type_header);
+ int temporal_idx = vp9_header.temporal_idx;
+ int spatial_idx = vp9_header.spatial_idx;
return (selected_tl_ < 0 || temporal_idx == kNoTemporalIdx ||
temporal_idx <= selected_tl_) &&
(selected_sl_ < 0 || spatial_idx == kNoSpatialIdx ||
spatial_idx <= selected_sl_);
}
+
+ return true;
}
void VideoAnalyzer::PollStats() {
diff --git a/video/video_analyzer.h b/video/video_analyzer.h
index 1069abf..7d60e7c 100644
--- a/video/video_analyzer.h
+++ b/video/video_analyzer.h
@@ -18,6 +18,8 @@
#include "api/task_queue/task_queue_base.h"
#include "api/video/video_source_interface.h"
+#include "modules/rtp_rtcp/source/rtp_packet.h"
+#include "modules/rtp_rtcp/source/video_rtp_depacketizer.h"
#include "rtc_base/event.h"
#include "rtc_base/numerics/running_statistics.h"
#include "rtc_base/platform_thread.h"
@@ -173,9 +175,7 @@
VideoFrame frame;
};
- bool IsInSelectedSpatialAndTemporalLayer(const uint8_t* packet,
- size_t length,
- const RTPHeader& header);
+ bool IsInSelectedSpatialAndTemporalLayer(const RtpPacket& rtp_packet);
void AddFrameComparison(const VideoFrame& reference,
const VideoFrame& render,
@@ -296,6 +296,8 @@
bool quit_ RTC_GUARDED_BY(comparison_lock_);
rtc::Event done_;
+ std::unique_ptr<VideoRtpDepacketizer> vp8_depacketizer_;
+ std::unique_ptr<VideoRtpDepacketizer> vp9_depacketizer_;
std::unique_ptr<test::RtpFileWriter> rtp_file_writer_;
Clock* const clock_;
const int64_t start_ms_;
