Delete redundant logic for setting is_first_packet_in_frame

A value for this flag was derived in RtpReceiverImpl::IncomingRtpPacket.
For audio, it was never used, and for video, it was overridden by
the result from RtpDepacketizer::ParsedPayload.

Bug: webrtc:7135
Change-Id: I597a57ca77d13b9a9145a9ee5e6624c1986777b9
Reviewed-on: https://webrtc-review.googlesource.com/3660
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19985}

modules/rtp_rtcp/source/rtp_receiver_audio.cc

@@ -133,8 +133,7 @@ int32_t RTPReceiverAudio::ParseRtpPacket(WebRtcRTPHeader* rtp_header,
                                          bool is_red,
                                          const uint8_t* payload,
                                          size_t payload_length,
-                                         int64_t timestamp_ms,
-                                         bool is_first_packet) {
+                                         int64_t timestamp_ms) {
   TRACE_EVENT2(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"), "Audio::ParseRtp",
                "seqnum", rtp_header->header.sequenceNumber, "timestamp",
                rtp_header->header.timestamp);

modules/rtp_rtcp/source/rtp_receiver_audio.h

@@ -49,8 +49,7 @@ class RTPReceiverAudio : public RTPReceiverStrategy,
                          bool is_red,
                          const uint8_t* packet,
                          size_t payload_length,
-                         int64_t timestamp_ms,
-                         bool is_first_packet) override;
+                         int64_t timestamp_ms) override;
 
   RTPAliveType ProcessDeadOrAlive(uint16_t last_payload_length) const override;
 

modules/rtp_rtcp/source/rtp_receiver_impl.cc

@@ -65,14 +65,11 @@ RtpReceiverImpl::RtpReceiverImpl(Clock* clock,
       rtp_payload_registry_(rtp_payload_registry),
       rtp_media_receiver_(rtp_media_receiver),
       cb_rtp_feedback_(incoming_messages_callback),
-      last_receive_time_(0),
-      last_received_payload_length_(0),
       ssrc_(0),
       num_csrcs_(0),
       current_remote_csrc_(),
       last_received_timestamp_(0),
-      last_received_frame_time_ms_(-1),
-      last_received_sequence_number_(0) {
+      last_received_frame_time_ms_(-1) {
   assert(incoming_messages_callback);
 
   memset(current_remote_csrc_, 0, sizeof(current_remote_csrc_));
@@ -169,23 +166,9 @@ bool RtpReceiverImpl::IncomingRtpPacket(
           : rtc::Optional<uint8_t>();
   UpdateSources(audio_level);
 
-  size_t payload_data_length = payload_length - rtp_header.paddingLength;
-
-  bool is_first_packet_in_frame = false;
-  {
-    rtc::CritScope lock(&critical_section_rtp_receiver_);
-    if (HaveReceivedFrame()) {
-      is_first_packet_in_frame =
-          last_received_sequence_number_ + 1 == rtp_header.sequenceNumber &&
-          last_received_timestamp_ != rtp_header.timestamp;
-    } else {
-      is_first_packet_in_frame = true;
-    }
-  }
-
   int32_t ret_val = rtp_media_receiver_->ParseRtpPacket(
       &webrtc_rtp_header, payload_specific, is_red, payload, payload_length,
-      clock_->TimeInMilliseconds(), is_first_packet_in_frame);
+      clock_->TimeInMilliseconds());
 
   if (ret_val < 0) {
     return false;
@@ -194,15 +177,11 @@ bool RtpReceiverImpl::IncomingRtpPacket(
   {
     rtc::CritScope lock(&critical_section_rtp_receiver_);
 
-    last_receive_time_ = clock_->TimeInMilliseconds();
-    last_received_payload_length_ = payload_data_length;
-
     if (in_order) {
       if (last_received_timestamp_ != rtp_header.timestamp) {
         last_received_timestamp_ = rtp_header.timestamp;
         last_received_frame_time_ms_ = clock_->TimeInMilliseconds();
       }
-      last_received_sequence_number_ = rtp_header.sequenceNumber;
     }
   }
   return true;
@@ -285,7 +264,6 @@ void RtpReceiverImpl::CheckSSRCChanged(const RTPHeader& rtp_header) {
       new_ssrc = true;
 
       last_received_timestamp_ = 0;
-      last_received_sequence_number_ = 0;
       last_received_frame_time_ms_ = -1;
 
       // Do we have a SSRC? Then the stream is restarted.

modules/rtp_rtcp/source/rtp_receiver_impl.h

@@ -89,8 +89,6 @@ class RtpReceiverImpl : public RtpReceiver {
   RtpFeedback* cb_rtp_feedback_;
 
   rtc::CriticalSection critical_section_rtp_receiver_;
-  int64_t last_receive_time_;
-  size_t last_received_payload_length_;
 
   // SSRCs.
   uint32_t ssrc_;
@@ -99,7 +97,6 @@ class RtpReceiverImpl : public RtpReceiver {
 
   uint32_t last_received_timestamp_;
   int64_t last_received_frame_time_ms_;
-  uint16_t last_received_sequence_number_;
 
   std::unordered_map<uint32_t, std::list<RtpSource>::iterator>
       iterator_by_csrc_;

modules/rtp_rtcp/source/rtp_receiver_strategy.h

@@ -36,15 +36,13 @@ class RTPReceiverStrategy {
   // Implementations are encouraged to use the provided packet buffer and RTP
   // header as arguments to the callback; implementations are also allowed to
   // make changes in the data as necessary. The specific_payload argument
-  // provides audio or video-specific data. The is_first_packet argument is true
-  // if this packet is either the first packet ever or the first in its frame.
+  // provides audio or video-specific data.
   virtual int32_t ParseRtpPacket(WebRtcRTPHeader* rtp_header,
                                  const PayloadUnion& specific_payload,
                                  bool is_red,
                                  const uint8_t* payload,
                                  size_t payload_length,
-                                 int64_t timestamp_ms,
-                                 bool is_first_packet) = 0;
+                                 int64_t timestamp_ms) = 0;
 
   virtual TelephoneEventHandler* GetTelephoneEventHandler() = 0;
 

modules/rtp_rtcp/source/rtp_receiver_video.cc

@@ -54,8 +54,7 @@ int32_t RTPReceiverVideo::ParseRtpPacket(WebRtcRTPHeader* rtp_header,
                                          bool is_red,
                                          const uint8_t* payload,
                                          size_t payload_length,
-                                         int64_t timestamp_ms,
-                                         bool is_first_packet) {
+                                         int64_t timestamp_ms) {
   TRACE_EVENT2(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"), "Video::ParseRtp",
                "seqnum", rtp_header->header.sequenceNumber, "timestamp",
                rtp_header->header.timestamp);
@@ -82,7 +81,6 @@ int32_t RTPReceiverVideo::ParseRtpPacket(WebRtcRTPHeader* rtp_header,
     return -1;
   }
 
-  rtp_header->type.Video.is_first_packet_in_frame = is_first_packet;
   RtpDepacketizer::ParsedPayload parsed_payload;
   if (!depacketizer->Parse(&parsed_payload, payload, payload_data_length))
     return -1;

modules/rtp_rtcp/source/rtp_receiver_video.h

@@ -30,8 +30,7 @@ class RTPReceiverVideo : public RTPReceiverStrategy {
                          bool is_red,
                          const uint8_t* packet,
                          size_t packet_length,
-                         int64_t timestamp,
-                         bool is_first_packet) override;
+                         int64_t timestamp) override;
 
   TelephoneEventHandler* GetTelephoneEventHandler() override { return NULL; }