gh-217 Add broken code for review

Author: ForeverASilver

src/tests/roc_pipeline/test_receiver_source.cpp

@@ -10,6 +10,7 @@
 
 #include "test_helpers/control_writer.h"
 #include "test_helpers/frame_reader.h"
+#include "test_helpers/packet_sender.h"
 #include "test_helpers/packet_writer.h"
 
 #include "roc_address/interface.h"
@@ -21,6 +22,10 @@
 #include "roc_pipeline/receiver_source.h"
 #include "roc_rtp/format_map.h"
 
+#include "roc_fec/composer.h"
+#include "roc_fec/headers.h"
+#include "roc_fec/writer.h"
+
 namespace roc {
 namespace pipeline {
 
@@ -57,6 +62,11 @@ packet::PacketFactory packet_factory(arena);
 
 rtp::FormatMap format_map(arena);
 
+fec::Composer<fec::LDPC_Source_PayloadID, fec::Source, fec::Footer>
+    ldpc_source_composer(&rtp_composer);
+fec::Composer<fec::LDPC_Repair_PayloadID, fec::Repair, fec::Header>
+    ldpc_repair_composer(NULL);
+
 ReceiverSlot* create_slot(ReceiverSource& source) {
     ReceiverSlot* slot = source.create_slot();
     CHECK(slot);
@@ -1744,6 +1754,93 @@ IGNORE_TEST(receiver_source, timestamp_mapping_remixing) {
     CHECK(first_ts);
 }
 
+packet::PacketPtr fill_one_packet(size_t sn,
+                                  size_t fec_payload_size = 193,
+                                  packet::IComposer* composer = NULL) {
+    CHECK(fec_payload_size > sizeof(rtp::Header));
+    const size_t rtp_payload_size = fec_payload_size - sizeof(rtp::Header);
+
+    packet::PacketPtr pp = packet_factory.new_packet();
+    CHECK(pp);
+
+    core::Slice<uint8_t> bp = byte_buffer_factory.new_buffer();
+    CHECK(bp);
+
+    if (!composer) {
+        composer = &ldpc_source_composer;
+    }
+    CHECK(composer->prepare(*pp, bp, rtp_payload_size));
+
+    pp->set_data(bp);
+
+    UNSIGNED_LONGS_EQUAL(rtp_payload_size, pp->rtp()->payload.size());
+    UNSIGNED_LONGS_EQUAL(fec_payload_size, pp->fec()->payload.size());
+
+    pp->add_flags(packet::Packet::FlagAudio);
+
+    pp->rtp()->source = 555;
+    pp->rtp()->payload_type = rtp::PayloadType_L16_Stereo;
+    pp->rtp()->seqnum = packet::seqnum_t(sn);
+    pp->rtp()->stream_timestamp = packet::stream_timestamp_t(sn * 10);
+
+    for (size_t i = 0; i < rtp_payload_size; i++) {
+        pp->rtp()->payload.data()[i] = uint8_t(sn + i);
+    }
+
+    return pp;
+}
+
+TEST(receiver_source, packet_buffer) {
+    packet::PacketPtr source_packets[20];
+
+    enum { Rate = SampleRate, Chans = Chans_Stereo, MaxPackets = 10 };
+
+    init(Rate, Chans, Rate, Chans);
+
+    ReceiverConfig config = make_config();
+    config.default_session.prebuf_len = 0;
+    ReceiverSource receiver(config, format_map, packet_factory, byte_buffer_factory,
+                            sample_buffer_factory, arena);
+    CHECK(receiver.is_valid());
+
+    ReceiverSlot* slot = create_slot(receiver);
+    CHECK(slot);
+
+    packet::IWriter* endpoint1_writer =
+        create_endpoint(slot, address::Iface_AudioSource, proto1);
+    CHECK(endpoint1_writer);
+
+    fec::WriterConfig writerConfig;
+    fec::CodecConfig codecConfig;
+    codecConfig.scheme = packet::FEC_LDPC_Staircase;
+
+    // setup encoder
+    core::ScopedPtr<fec::IBlockEncoder> encoder(
+        fec::CodecMap::instance().new_encoder(codecConfig, byte_buffer_factory, arena),
+        arena);
+    CHECK(encoder);
+
+    // setup fec writer
+    fec::Writer fec_writer(writerConfig, packet::FEC_LDPC_Staircase, *encoder,
+                           *endpoint1_writer, ldpc_source_composer, ldpc_repair_composer,
+                           packet_factory, byte_buffer_factory, arena);
+    CHECK(fec_writer.is_valid());
+
+    test::PacketWriter packet_writer(arena, fec_writer, ldpc_source_composer, format_map,
+                                     packet_factory, byte_buffer_factory, PayloadType_Ch2,
+                                     src1, dst1);
+
+    // setup reader
+    test::FrameReader frame_reader(receiver, sample_buffer_factory);
+
+    packet_writer.write_packets(writerConfig.n_source_packets, SamplesPerPacket,
+                                output_sample_spec);
+
+    receiver.refresh(frame_reader.refresh_ts());
+    frame_reader.read_zero_samples(SamplesPerFrame, output_sample_spec);
+    UNSIGNED_LONGS_EQUAL(0, receiver.num_sessions());
+}
+
 TEST(receiver_source, metrics_sessions) {
     enum { Rate = SampleRate, Chans = Chans_Stereo, MaxSess = 10 };