Filtering audio playout events with SSRC in NetEq RTP player.

Bug: webrtc:9259
Change-Id: I0b88aa6a7b49bd786637c7ffd9b94c92c608c841
Reviewed-on: https://webrtc-review.googlesource.com/76141
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23414}

modules/audio_coding/neteq/tools/neteq_packet_source_input.cc

@@ -60,6 +60,12 @@ std::unique_ptr<NetEqInput::PacketData> NetEqPacketSourceInput::PopPacket() {
   return packet_data;
 }
 
+void NetEqPacketSourceInput::SelectSsrc(uint32_t ssrc) {
+  source()->SelectSsrc(ssrc);
+  if (packet_ && packet_->header().ssrc != ssrc)
+    LoadNextPacket();
+}
+
 NetEqRtpDumpInput::NetEqRtpDumpInput(const std::string& file_name,
                                      const RtpHeaderExtensionMap& hdr_ext_map)
     : source_(RtpFileSource::Create(file_name)) {

modules/audio_coding/neteq/tools/neteq_packet_source_input.h

@@ -33,6 +33,7 @@ class NetEqPacketSourceInput : public NetEqInput {
   std::unique_ptr<PacketData> PopPacket() override;
   rtc::Optional<RTPHeader> NextHeader() const override;
   bool ended() const override { return !next_output_event_ms_; }
+  void SelectSsrc(uint32_t);
 
  protected:
   virtual PacketSource* source() = 0;

modules/audio_coding/neteq/tools/neteq_rtpplay.cc

@@ -226,56 +226,6 @@ rtc::Optional<int> CodecSampleRate(uint8_t payload_type) {
   return rtc::nullopt;
 }
 
-// Class to let through only the packets with a given SSRC. Should be used as an
-// outer layer on another NetEqInput object.
-class FilterSsrcInput : public NetEqInput {
- public:
-  FilterSsrcInput(std::unique_ptr<NetEqInput> source, uint32_t ssrc)
-      : source_(std::move(source)), ssrc_(ssrc) {
-    FindNextWithCorrectSsrc();
-    RTC_CHECK(source_->NextHeader()) << "Found no packet with SSRC = 0x"
-                                     << std::hex << ssrc_;
-  }
-
-  // All methods but PopPacket() simply relay to the |source_| object.
-  rtc::Optional<int64_t> NextPacketTime() const override {
-    return source_->NextPacketTime();
-  }
-  rtc::Optional<int64_t> NextOutputEventTime() const override {
-    return source_->NextOutputEventTime();
-  }
-
-  // Returns the next packet, and throws away upcoming packets that do not match
-  // the desired SSRC.
-  std::unique_ptr<PacketData> PopPacket() override {
-    std::unique_ptr<PacketData> packet_to_return = source_->PopPacket();
-    RTC_DCHECK(!packet_to_return || packet_to_return->header.ssrc == ssrc_);
-    // Pre-fetch the next packet with correct SSRC. Hence, |source_| will always
-    // be have a valid packet (or empty if no more packets are available) when
-    // this method returns.
-    FindNextWithCorrectSsrc();
-    return packet_to_return;
-  }
-
-  void AdvanceOutputEvent() override { source_->AdvanceOutputEvent(); }
-
-  bool ended() const override { return source_->ended(); }
-
-  rtc::Optional<RTPHeader> NextHeader() const override {
-    return source_->NextHeader();
-  }
-
- private:
-  void FindNextWithCorrectSsrc() {
-    while (source_->NextHeader() && source_->NextHeader()->ssrc != ssrc_) {
-      source_->PopPacket();
-    }
-  }
-
-  std::unique_ptr<NetEqInput> source_;
-  uint32_t ssrc_;
-};
-
 // A callback class which prints whenver the inserted packet stream changes
 // the SSRC.
 class SsrcSwitchDetector : public NetEqPostInsertPacket {
@@ -386,14 +336,13 @@ int RunTest(int argc, char* argv[]) {
   if (strlen(FLAG_ssrc) > 0) {
     uint32_t ssrc;
     RTC_CHECK(ParseSsrc(FLAG_ssrc, &ssrc)) << "Flag verification has failed.";
-    input.reset(new FilterSsrcInput(std::move(input), ssrc));
+    static_cast<NetEqPacketSourceInput*>(input.get())->SelectSsrc(ssrc);
   }
 
   // Check the sample rate.
   rtc::Optional<int> sample_rate_hz;
   std::set<std::pair<int, uint32_t>> discarded_pt_and_ssrc;
-  while (input->NextHeader()) {
+  while (rtc::Optional<RTPHeader> first_rtp_header = input->NextHeader()) {
-    rtc::Optional<RTPHeader> first_rtp_header = input->NextHeader();
     RTC_DCHECK(first_rtp_header);
     sample_rate_hz = CodecSampleRate(first_rtp_header->payloadType);
     if (sample_rate_hz) {

modules/audio_coding/neteq/tools/rtc_event_log_source.cc

@@ -87,9 +87,11 @@ int64_t RtcEventLogSource::NextAudioOutputEventMs() {
         ParsedRtcEventLogNew::AUDIO_PLAYOUT_EVENT) {
       LoggedAudioPlayoutEvent playout_event =
           parsed_stream_.GetAudioPlayout(audio_output_index_);
+      if (!(use_ssrc_filter_ && playout_event.ssrc != ssrc_)) {
         audio_output_index_++;
         return playout_event.timestamp_us / 1000;
       }
+    }
     audio_output_index_++;
   }
   return std::numeric_limits<int64_t>::max();