This reverts commit 6c2c13af06b32778b86950681758a7970d1c5d9e. Reason for revert: Intend to investigate and fix perf problems. Original change's description: > Revert "Reland "Move rtp-specific config out of EncoderSettings."" > > This reverts commit 04dd1768625eb2241d1fb97fd0137897e703e266. > > Reason for revert: Regression in ramp up perf tests. > > Original change's description: > > Reland "Move rtp-specific config out of EncoderSettings." > > > > This is a reland of bc900cb1d1810fcf678fe41cf1e3966daa39c88c > > > > Original change's description: > > > Move rtp-specific config out of EncoderSettings. > > > > > > In VideoSendStream::Config, move payload_name and payload_type from > > > EncoderSettings to Rtp. > > > > > > EncoderSettings now contains configuration for VideoStreamEncoder only, > > > and should perhaps be renamed in a follow up cl. It's no longer > > > passed as an argument to VideoCodecInitializer::SetupCodec. > > > > > > The latter then needs a different way to know the codec type, > > > which is provided by a new codec_type member in VideoEncoderConfig. > > > > > > Bug: webrtc:8830 > > > Change-Id: Ifcc691aef1ee6a95e43c0452c5e630d92a511cd6 > > > Reviewed-on: https://webrtc-review.googlesource.com/62062 > > > Commit-Queue: Niels Moller <nisse@webrtc.org> > > > Reviewed-by: Magnus Jedvert <magjed@webrtc.org> > > > Reviewed-by: Stefan Holmer <stefan@webrtc.org> > > > Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> > > > Cr-Commit-Position: refs/heads/master@{#22532} > > > > Bug: webrtc:8830 > > Change-Id: If88ef7d57cdaa4fae3c7b2a97ea5a6e1b833e019 > > Reviewed-on: https://webrtc-review.googlesource.com/63721 > > Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> > > Reviewed-by: Stefan Holmer <stefan@webrtc.org> > > Commit-Queue: Niels Moller <nisse@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#22595} > > TBR=brandtr@webrtc.org,magjed@webrtc.org,nisse@webrtc.org,stefan@webrtc.org > > Bug: webrtc:8830,chromium:827080 > Change-Id: Iaaf146de91ec5c0d741b8efdf143f7e173084fef > Reviewed-on: https://webrtc-review.googlesource.com/65520 > Commit-Queue: Niels Moller <nisse@webrtc.org> > Reviewed-by: Niels Moller <nisse@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#22677} TBR=brandtr@webrtc.org,magjed@webrtc.org,nisse@webrtc.org,stefan@webrtc.org # Not skipping CQ checks because original CL landed > 1 day ago. Bug: webrtc:8830, chromium:827080 Change-Id: I9b62987bf5daced90dfeb3ebb6739c80117c487f Reviewed-on: https://webrtc-review.googlesource.com/66862 Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22751}
376 lines
12 KiB
C++
376 lines
12 KiB
C++
/*
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* Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include <stdio.h>
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#include <map>
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#include <memory>
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#include <sstream>
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#include "api/video_codecs/video_decoder.h"
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#include "call/call.h"
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#include "common_video/libyuv/include/webrtc_libyuv.h"
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#include "logging/rtc_event_log/rtc_event_log.h"
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#include "modules/rtp_rtcp/include/rtp_header_parser.h"
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#include "rtc_base/checks.h"
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#include "rtc_base/flags.h"
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#include "rtc_base/string_to_number.h"
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#include "rtc_base/timeutils.h"
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#include "system_wrappers/include/clock.h"
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#include "system_wrappers/include/sleep.h"
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#include "test/call_test.h"
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#include "test/encoder_settings.h"
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#include "test/fake_decoder.h"
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#include "test/gtest.h"
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#include "test/null_transport.h"
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#include "test/rtp_file_reader.h"
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#include "test/run_loop.h"
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#include "test/run_test.h"
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#include "test/testsupport/frame_writer.h"
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#include "test/video_capturer.h"
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#include "test/video_renderer.h"
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#include "typedefs.h" // NOLINT(build/include)
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namespace {
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static bool ValidatePayloadType(int32_t payload_type) {
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return payload_type > 0 && payload_type <= 127;
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}
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static bool ValidateSsrc(const char* ssrc_string) {
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return rtc::StringToNumber<uint32_t>(ssrc_string).has_value();
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}
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static bool ValidateOptionalPayloadType(int32_t payload_type) {
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return payload_type == -1 || ValidatePayloadType(payload_type);
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}
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static bool ValidateRtpHeaderExtensionId(int32_t extension_id) {
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return extension_id >= -1 && extension_id < 15;
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}
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bool ValidateInputFilenameNotEmpty(const std::string& string) {
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return !string.empty();
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}
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} // namespace
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namespace webrtc {
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namespace flags {
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// TODO(pbos): Multiple receivers.
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// Flag for payload type.
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DEFINE_int(media_payload_type,
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test::CallTest::kPayloadTypeVP8,
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"Media payload type");
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static int MediaPayloadType() {
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return static_cast<int>(FLAG_media_payload_type);
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}
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// Flag for RED payload type.
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DEFINE_int(red_payload_type,
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test::CallTest::kRedPayloadType,
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"RED payload type");
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static int RedPayloadType() {
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return static_cast<int>(FLAG_red_payload_type);
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}
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// Flag for ULPFEC payload type.
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DEFINE_int(ulpfec_payload_type,
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test::CallTest::kUlpfecPayloadType,
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"ULPFEC payload type");
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static int UlpfecPayloadType() {
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return static_cast<int>(FLAG_ulpfec_payload_type);
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}
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DEFINE_int(media_payload_type_rtx,
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test::CallTest::kSendRtxPayloadType,
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"Media over RTX payload type");
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static int MediaPayloadTypeRtx() {
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return static_cast<int>(FLAG_media_payload_type_rtx);
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}
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DEFINE_int(red_payload_type_rtx,
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test::CallTest::kRtxRedPayloadType,
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"RED over RTX payload type");
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static int RedPayloadTypeRtx() {
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return static_cast<int>(FLAG_red_payload_type_rtx);
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}
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// Flag for SSRC.
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const std::string& DefaultSsrc() {
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static const std::string ssrc = std::to_string(
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test::CallTest::kVideoSendSsrcs[0]);
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return ssrc;
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}
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DEFINE_string(ssrc, DefaultSsrc().c_str(), "Incoming SSRC");
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static uint32_t Ssrc() {
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return rtc::StringToNumber<uint32_t>(FLAG_ssrc).value();
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}
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const std::string& DefaultSsrcRtx() {
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static const std::string ssrc_rtx = std::to_string(
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test::CallTest::kSendRtxSsrcs[0]);
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return ssrc_rtx;
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}
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DEFINE_string(ssrc_rtx, DefaultSsrcRtx().c_str(), "Incoming RTX SSRC");
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static uint32_t SsrcRtx() {
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return rtc::StringToNumber<uint32_t>(FLAG_ssrc_rtx).value();
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}
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// Flag for abs-send-time id.
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DEFINE_int(abs_send_time_id, -1, "RTP extension ID for abs-send-time");
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static int AbsSendTimeId() { return static_cast<int>(FLAG_abs_send_time_id); }
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// Flag for transmission-offset id.
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DEFINE_int(transmission_offset_id,
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-1,
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"RTP extension ID for transmission-offset");
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static int TransmissionOffsetId() {
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return static_cast<int>(FLAG_transmission_offset_id);
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}
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// Flag for rtpdump input file.
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DEFINE_string(input_file, "", "input file");
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static std::string InputFile() {
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return static_cast<std::string>(FLAG_input_file);
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}
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// Flag for raw output files.
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DEFINE_string(out_base, "", "Basename (excluding .jpg) for raw output");
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static std::string OutBase() {
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return static_cast<std::string>(FLAG_out_base);
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}
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DEFINE_string(decoder_bitstream_filename, "", "Decoder bitstream output file");
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static std::string DecoderBitstreamFilename() {
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return static_cast<std::string>(FLAG_decoder_bitstream_filename);
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}
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// Flag for video codec.
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DEFINE_string(codec, "VP8", "Video codec");
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static std::string Codec() { return static_cast<std::string>(FLAG_codec); }
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DEFINE_bool(help, false, "Print this message.");
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} // namespace flags
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static const uint32_t kReceiverLocalSsrc = 0x123456;
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class FileRenderPassthrough : public rtc::VideoSinkInterface<VideoFrame> {
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public:
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FileRenderPassthrough(const std::string& basename,
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rtc::VideoSinkInterface<VideoFrame>* renderer)
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: basename_(basename), renderer_(renderer), file_(nullptr), count_(0) {}
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~FileRenderPassthrough() {
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if (file_)
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fclose(file_);
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}
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private:
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void OnFrame(const VideoFrame& video_frame) override {
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if (renderer_)
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renderer_->OnFrame(video_frame);
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if (basename_.empty())
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return;
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std::stringstream filename;
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filename << basename_ << count_++ << "_" << video_frame.timestamp()
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<< ".jpg";
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test::JpegFrameWriter frame_writer(filename.str());
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RTC_CHECK(frame_writer.WriteFrame(video_frame, 100));
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}
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const std::string basename_;
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rtc::VideoSinkInterface<VideoFrame>* const renderer_;
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FILE* file_;
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size_t count_;
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};
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class DecoderBitstreamFileWriter : public EncodedFrameObserver {
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public:
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explicit DecoderBitstreamFileWriter(const char* filename)
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: file_(fopen(filename, "wb")) {
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RTC_DCHECK(file_);
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}
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~DecoderBitstreamFileWriter() { fclose(file_); }
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virtual void EncodedFrameCallback(const EncodedFrame& encoded_frame) {
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fwrite(encoded_frame.data_, 1, encoded_frame.length_, file_);
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}
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private:
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FILE* file_;
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};
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void RtpReplay() {
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std::stringstream window_title;
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window_title << "Playback Video (" << flags::InputFile() << ")";
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std::unique_ptr<test::VideoRenderer> playback_video(
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test::VideoRenderer::Create(window_title.str().c_str(), 640, 480));
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FileRenderPassthrough file_passthrough(flags::OutBase(),
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playback_video.get());
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webrtc::RtcEventLogNullImpl event_log;
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std::unique_ptr<Call> call(Call::Create(Call::Config(&event_log)));
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test::NullTransport transport;
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VideoReceiveStream::Config receive_config(&transport);
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receive_config.rtp.remote_ssrc = flags::Ssrc();
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receive_config.rtp.local_ssrc = kReceiverLocalSsrc;
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receive_config.rtp.rtx_ssrc = flags::SsrcRtx();
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receive_config.rtp
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.rtx_associated_payload_types[flags::MediaPayloadTypeRtx()] =
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flags::MediaPayloadType();
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receive_config.rtp.rtx_associated_payload_types[flags::RedPayloadTypeRtx()] =
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flags::RedPayloadType();
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receive_config.rtp.ulpfec_payload_type = flags::UlpfecPayloadType();
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receive_config.rtp.red_payload_type = flags::RedPayloadType();
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receive_config.rtp.nack.rtp_history_ms = 1000;
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if (flags::TransmissionOffsetId() != -1) {
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receive_config.rtp.extensions.push_back(RtpExtension(
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RtpExtension::kTimestampOffsetUri, flags::TransmissionOffsetId()));
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}
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if (flags::AbsSendTimeId() != -1) {
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receive_config.rtp.extensions.push_back(
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RtpExtension(RtpExtension::kAbsSendTimeUri, flags::AbsSendTimeId()));
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}
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receive_config.renderer = &file_passthrough;
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VideoReceiveStream::Decoder decoder;
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std::unique_ptr<DecoderBitstreamFileWriter> bitstream_writer;
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if (!flags::DecoderBitstreamFilename().empty()) {
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bitstream_writer.reset(new DecoderBitstreamFileWriter(
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flags::DecoderBitstreamFilename().c_str()));
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receive_config.pre_decode_callback = bitstream_writer.get();
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}
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decoder = test::CreateMatchingDecoder(flags::MediaPayloadType(),
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flags::Codec());
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if (!flags::DecoderBitstreamFilename().empty()) {
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// Replace with a null decoder if we're writing the bitstream to a file
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// instead.
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delete decoder.decoder;
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decoder.decoder = new test::FakeNullDecoder();
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}
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receive_config.decoders.push_back(decoder);
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VideoReceiveStream* receive_stream =
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call->CreateVideoReceiveStream(std::move(receive_config));
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std::unique_ptr<test::RtpFileReader> rtp_reader(test::RtpFileReader::Create(
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test::RtpFileReader::kRtpDump, flags::InputFile()));
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if (!rtp_reader) {
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rtp_reader.reset(test::RtpFileReader::Create(test::RtpFileReader::kPcap,
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flags::InputFile()));
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if (!rtp_reader) {
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fprintf(stderr,
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"Couldn't open input file as either a rtpdump or .pcap. Note "
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"that .pcapng is not supported.\nTrying to interpret the file as "
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"length/packet interleaved.\n");
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rtp_reader.reset(test::RtpFileReader::Create(
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test::RtpFileReader::kLengthPacketInterleaved, flags::InputFile()));
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if (!rtp_reader) {
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fprintf(stderr,
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"Unable to open input file with any supported format\n");
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return;
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}
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}
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}
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receive_stream->Start();
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int64_t replay_start_ms = -1;
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int num_packets = 0;
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std::map<uint32_t, int> unknown_packets;
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while (true) {
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int64_t now_ms = rtc::TimeMillis();
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if (replay_start_ms == -1)
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replay_start_ms = now_ms;
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test::RtpPacket packet;
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if (!rtp_reader->NextPacket(&packet))
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break;
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int64_t deliver_in_ms = replay_start_ms + packet.time_ms - now_ms;
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if (deliver_in_ms > 0)
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SleepMs(deliver_in_ms);
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++num_packets;
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switch (call->Receiver()->DeliverPacket(
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webrtc::MediaType::VIDEO,
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rtc::CopyOnWriteBuffer(packet.data, packet.length), PacketTime())) {
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case PacketReceiver::DELIVERY_OK:
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break;
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case PacketReceiver::DELIVERY_UNKNOWN_SSRC: {
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RTPHeader header;
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std::unique_ptr<RtpHeaderParser> parser(RtpHeaderParser::Create());
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parser->Parse(packet.data, packet.length, &header);
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if (unknown_packets[header.ssrc] == 0)
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fprintf(stderr, "Unknown SSRC: %u!\n", header.ssrc);
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++unknown_packets[header.ssrc];
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break;
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}
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case PacketReceiver::DELIVERY_PACKET_ERROR: {
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fprintf(stderr, "Packet error, corrupt packets or incorrect setup?\n");
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RTPHeader header;
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std::unique_ptr<RtpHeaderParser> parser(RtpHeaderParser::Create());
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parser->Parse(packet.data, packet.length, &header);
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fprintf(stderr, "Packet len=%zu pt=%u seq=%u ts=%u ssrc=0x%8x\n",
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packet.length, header.payloadType, header.sequenceNumber,
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header.timestamp, header.ssrc);
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break;
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}
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}
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}
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fprintf(stderr, "num_packets: %d\n", num_packets);
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for (std::map<uint32_t, int>::const_iterator it = unknown_packets.begin();
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it != unknown_packets.end();
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++it) {
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fprintf(
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stderr, "Packets for unknown ssrc '%u': %d\n", it->first, it->second);
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}
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call->DestroyVideoReceiveStream(receive_stream);
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delete decoder.decoder;
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}
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} // namespace webrtc
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int main(int argc, char* argv[]) {
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::testing::InitGoogleTest(&argc, argv);
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if (rtc::FlagList::SetFlagsFromCommandLine(&argc, argv, true)) {
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return 1;
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}
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if (webrtc::flags::FLAG_help) {
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rtc::FlagList::Print(nullptr, false);
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return 0;
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}
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RTC_CHECK(ValidatePayloadType(webrtc::flags::FLAG_media_payload_type));
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RTC_CHECK(ValidatePayloadType(webrtc::flags::FLAG_media_payload_type_rtx));
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RTC_CHECK(ValidateOptionalPayloadType(webrtc::flags::FLAG_red_payload_type));
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RTC_CHECK(
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ValidateOptionalPayloadType(webrtc::flags::FLAG_red_payload_type_rtx));
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RTC_CHECK(
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ValidateOptionalPayloadType(webrtc::flags::FLAG_ulpfec_payload_type));
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RTC_CHECK(ValidateSsrc(webrtc::flags::FLAG_ssrc));
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RTC_CHECK(ValidateSsrc(webrtc::flags::FLAG_ssrc_rtx));
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RTC_CHECK(ValidateRtpHeaderExtensionId(webrtc::flags::FLAG_abs_send_time_id));
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RTC_CHECK(ValidateRtpHeaderExtensionId(
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webrtc::flags::FLAG_transmission_offset_id));
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RTC_CHECK(ValidateInputFilenameNotEmpty(webrtc::flags::FLAG_input_file));
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webrtc::test::RunTest(webrtc::RtpReplay);
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return 0;
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}
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