This changes the following module directories:
* webrtc/modules/audio_conference_mixer/interface
* webrtc/modules/interface
* webrtc/modules/media_file/interface
* webrtc/modules/rtp_rtcp/interface
* webrtc/modules/utility/interface
To avoid breaking downstream, I followed this recipe:
1. Copy the interface dir to a new sibling directory: include
2. Update the header guards in the include directory to match the style guide.
3. Update the header guards in the interface directory to match the ones in include. This is required to avoid getting redefinitions in the not-yet-updated downstream code.
4. Add a pragma warning in the header files in the interface dir. Example:
#pragma message("WARNING: webrtc/modules/interface is DEPRECATED; "
"use webrtc/modules/include")
5. Search for all source references to webrtc/modules/interface and update them to webrtc/modules/include (*.c*,*.h,*.mm,*.S)
6. Update all GYP+GN files. This required manual inspection since many subdirectories of webrtc/modules referenced the interface dir using ../interface etc(*.gyp*,*.gn*)
BUG=5095
TESTED=Passing compile-trybots with --clobber flag:
git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel -m tryserver.webrtc
R=stefan@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1417683006 .
Cr-Commit-Position: refs/heads/master@{#10500}
331 lines
11 KiB
C++
331 lines
11 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 <sstream>
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#include "gflags/gflags.h"
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#include "testing/gtest/include/gtest/gtest.h"
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#include "webrtc/base/checks.h"
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#include "webrtc/base/scoped_ptr.h"
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#include "webrtc/call.h"
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#include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
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#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
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#include "webrtc/system_wrappers/include/clock.h"
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#include "webrtc/system_wrappers/include/sleep.h"
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#include "webrtc/test/encoder_settings.h"
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#include "webrtc/test/null_transport.h"
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#include "webrtc/test/fake_decoder.h"
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#include "webrtc/test/rtp_file_reader.h"
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#include "webrtc/test/run_loop.h"
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#include "webrtc/test/run_test.h"
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#include "webrtc/test/video_capturer.h"
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#include "webrtc/test/video_renderer.h"
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#include "webrtc/typedefs.h"
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#include "webrtc/video_decoder.h"
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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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static bool ValidatePayloadType(const char* flagname, int32_t payload_type) {
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return payload_type > 0 && payload_type <= 127;
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}
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DEFINE_int32(payload_type, 0, "Payload type");
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static int PayloadType() { return static_cast<int>(FLAGS_payload_type); }
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static const bool payload_dummy =
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google::RegisterFlagValidator(&FLAGS_payload_type, &ValidatePayloadType);
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// Flag for SSRC.
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static bool ValidateSsrc(const char* flagname, uint64_t ssrc) {
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return ssrc > 0 && ssrc <= 0xFFFFFFFFu;
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}
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DEFINE_uint64(ssrc, 0, "Incoming SSRC");
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static uint32_t Ssrc() { return static_cast<uint32_t>(FLAGS_ssrc); }
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static const bool ssrc_dummy =
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google::RegisterFlagValidator(&FLAGS_ssrc, &ValidateSsrc);
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static bool ValidateOptionalPayloadType(const char* flagname,
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int32_t payload_type) {
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return payload_type == -1 || ValidatePayloadType(flagname, payload_type);
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}
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// Flag for RED payload type.
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DEFINE_int32(red_payload_type, -1, "RED payload type");
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static int RedPayloadType() {
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return static_cast<int>(FLAGS_red_payload_type);
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}
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static const bool red_dummy =
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google::RegisterFlagValidator(&FLAGS_red_payload_type,
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&ValidateOptionalPayloadType);
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// Flag for ULPFEC payload type.
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DEFINE_int32(fec_payload_type, -1, "ULPFEC payload type");
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static int FecPayloadType() {
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return static_cast<int>(FLAGS_fec_payload_type);
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}
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static const bool fec_dummy =
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google::RegisterFlagValidator(&FLAGS_fec_payload_type,
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&ValidateOptionalPayloadType);
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// Flag for abs-send-time id.
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static bool ValidateRtpHeaderExtensionId(const char* flagname,
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int32_t extension_id) {
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return extension_id >= -1 || extension_id < 15;
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}
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DEFINE_int32(abs_send_time_id, -1, "RTP extension ID for abs-send-time");
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static int AbsSendTimeId() { return static_cast<int>(FLAGS_abs_send_time_id); }
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static const bool abs_send_time_dummy =
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google::RegisterFlagValidator(&FLAGS_abs_send_time_id,
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&ValidateRtpHeaderExtensionId);
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// Flag for transmission-offset id.
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DEFINE_int32(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>(FLAGS_transmission_offset_id);
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}
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static const bool timestamp_offset_dummy =
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google::RegisterFlagValidator(&FLAGS_transmission_offset_id,
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&ValidateRtpHeaderExtensionId);
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// Flag for rtpdump input file.
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bool ValidateInputFilenameNotEmpty(const char* flagname,
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const std::string& string) {
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return !string.empty();
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}
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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>(FLAGS_input_file);
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}
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static const bool input_file_dummy =
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google::RegisterFlagValidator(&FLAGS_input_file,
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&ValidateInputFilenameNotEmpty);
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// Flag for raw output files.
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DEFINE_string(out_base, "", "Basename (excluding .yuv) for raw output");
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static std::string OutBase() {
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return static_cast<std::string>(FLAGS_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>(FLAGS_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>(FLAGS_codec); }
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} // namespace flags
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static const uint32_t kReceiverLocalSsrc = 0x123456;
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class FileRenderPassthrough : public VideoRenderer {
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public:
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FileRenderPassthrough(const std::string& basename, VideoRenderer* renderer)
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: basename_(basename),
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renderer_(renderer),
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file_(nullptr),
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count_(0),
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last_width_(0),
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last_height_(0) {}
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~FileRenderPassthrough() {
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if (file_ != nullptr)
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fclose(file_);
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}
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private:
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void RenderFrame(const VideoFrame& video_frame,
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int time_to_render_ms) override {
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if (renderer_ != nullptr)
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renderer_->RenderFrame(video_frame, time_to_render_ms);
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if (basename_.empty())
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return;
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if (last_width_ != video_frame.width() ||
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last_height_ != video_frame.height()) {
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if (file_ != nullptr)
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fclose(file_);
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std::stringstream filename;
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filename << basename_;
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if (++count_ > 1)
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filename << '-' << count_;
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filename << '_' << video_frame.width() << 'x' << video_frame.height()
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<< ".yuv";
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file_ = fopen(filename.str().c_str(), "wb");
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if (file_ == nullptr) {
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fprintf(stderr,
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"Couldn't open file for writing: %s\n",
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filename.str().c_str());
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}
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}
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last_width_ = video_frame.width();
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last_height_ = video_frame.height();
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if (file_ == nullptr)
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return;
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PrintVideoFrame(video_frame, file_);
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}
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bool IsTextureSupported() const override { return false; }
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const std::string basename_;
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VideoRenderer* const renderer_;
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FILE* file_;
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size_t count_;
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int last_width_;
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int last_height_;
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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_ != nullptr);
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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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rtc::scoped_ptr<test::VideoRenderer> playback_video(
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test::VideoRenderer::Create("Playback Video", 640, 480));
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FileRenderPassthrough file_passthrough(flags::OutBase(),
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playback_video.get());
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rtc::scoped_ptr<Call> call(Call::Create(Call::Config()));
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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.fec.ulpfec_payload_type = flags::FecPayloadType();
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receive_config.rtp.fec.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(
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RtpExtension(RtpExtension::kTOffset, 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::kAbsSendTime, flags::AbsSendTimeId()));
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}
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receive_config.renderer = &file_passthrough;
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VideoSendStream::Config::EncoderSettings encoder_settings;
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encoder_settings.payload_name = flags::Codec();
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encoder_settings.payload_type = flags::PayloadType();
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VideoReceiveStream::Decoder decoder;
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rtc::scoped_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(encoder_settings);
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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(receive_config);
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rtc::scoped_ptr<test::RtpFileReader> rtp_reader(test::RtpFileReader::Create(
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test::RtpFileReader::kRtpDump, flags::InputFile()));
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if (rtp_reader.get() == nullptr) {
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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.get() == nullptr) {
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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.get() == nullptr) {
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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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uint32_t last_time_ms = 0;
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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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test::RtpPacket packet;
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if (!rtp_reader->NextPacket(&packet))
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break;
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++num_packets;
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switch (call->Receiver()->DeliverPacket(webrtc::MediaType::ANY, packet.data,
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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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rtc::scoped_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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break;
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}
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if (last_time_ms != 0 && last_time_ms != packet.time_ms) {
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SleepMs(packet.time_ms - last_time_ms);
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}
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last_time_ms = packet.time_ms;
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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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google::ParseCommandLineFlags(&argc, &argv, true);
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webrtc::test::RunTest(webrtc::RtpReplay);
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return 0;
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}
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