Use RtpPacketToSend in RtpSenderVideo

webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc

 /*
  *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
 #include "webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h"
 
 #include <string.h>
 
 #include "webrtc/base/logging.h"
 #include "webrtc/base/timeutils.h"
 #include "webrtc/base/trace_event.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/source/rtp_header_extensions.h"
+#include "webrtc/modules/rtp_rtcp/source/rtp_packet_to_send.h"
 #include "webrtc/modules/rtp_rtcp/source/byte_io.h"
 
 namespace webrtc {
 
 static const int kDtmfFrequencyHz = 8000;
 
 RTPSenderAudio::RTPSenderAudio(Clock* clock, RTPSender* rtp_sender)
     : clock_(clock),
       rtp_sender_(rtp_sender),
       packet_size_samples_(160),
@@ skipping 117 matching lines @@
 }
 
 bool RTPSenderAudio::SendAudio(FrameType frame_type,
                                   int8_t payload_type,
                                   uint32_t capture_timestamp,
                                   const uint8_t* payload_data,
                                   size_t data_size,
                                   const RTPFragmentationHeader* fragmentation) {
   // TODO(pwestin) Breakup function in smaller functions.
   size_t payload_size = data_size;
-  size_t max_payload_length = rtp_sender_->MaxPayloadLength();
   uint16_t dtmf_length_ms = 0;
   uint8_t key = 0;
   int red_payload_type;
   uint8_t audio_level_dbov;
   int8_t dtmf_payload_type;
   uint16_t packet_size_samples;
   {
     rtc::CritScope cs(&send_audio_critsect_);
     red_payload_type = red_payload_type_;
     audio_level_dbov = audio_level_dbov_;
@@ skipping 73 matching lines @@
     return true;
   }
   if (payload_size == 0 || payload_data == NULL) {
     if (frame_type == kEmptyFrame) {
       // we don't send empty audio RTP packets
       // no error since we use it to drive DTMF when we use VAD
       return true;
     }
     return false;
   }
-  uint8_t data_buffer[IP_PACKET_SIZE];
+
+  std::unique_ptr<RtpPacketToSend> packet = rtp_sender_->AllocatePacket(false);
+  if (!packet)
+    return false;
   bool marker_bit = MarkerBit(frame_type, payload_type);
+  packet->SetPayloadType(payload_type);
+  packet->SetMarker(marker_bit);
+  packet->SetTimestamp(capture_timestamp);
+  packet->set_capture_time_ms(clock_->TimeInMilliseconds());
+  // Update audio level extension, if included.
+  packet->SetExtension<AudioLevel>(frame_type == kAudioFrameSpeech,
+                                   audio_level_dbov);
 
-  int32_t rtpHeaderLength = 0;
-  uint16_t timestampOffset = 0;
-
-  if (red_payload_type >= 0 && fragmentation && !marker_bit &&
-      fragmentation->fragmentationVectorSize > 1) {
-    // have we configured RED? use its payload type
-    // we need to get the current timestamp to calc the diff
-    uint32_t old_timestamp = rtp_sender_->Timestamp();
-    rtpHeaderLength = rtp_sender_->BuildRtpHeader(data_buffer, red_payload_type,
-                                                  marker_bit, capture_timestamp,
-                                                  clock_->TimeInMilliseconds());
-
-    timestampOffset = uint16_t(rtp_sender_->Timestamp() - old_timestamp);
+  if (fragmentation && fragmentation->fragmentationVectorSize > 0) {
+    // use the fragment info if we have one
+    uint8_t* payload =
+        packet->AllocatePayload(1 + fragmentation->fragmentationLength[0]);
+    if (!payload)
+      return false;
+    payload[0] = fragmentation->fragmentationPlType[0];
+    memcpy(payload + 1, payload_data + fragmentation->fragmentationOffset[0],
+           fragmentation->fragmentationLength[0]);
   } else {
-    rtpHeaderLength = rtp_sender_->BuildRtpHeader(data_buffer, payload_type,
-                                                  marker_bit, capture_timestamp,
-                                                  clock_->TimeInMilliseconds());
-  }
-  if (rtpHeaderLength <= 0) {
-    return false;
-  }
-  if (max_payload_length < (rtpHeaderLength + payload_size)) {
-    // Too large payload buffer.
-    return false;
-  }
-  if (red_payload_type >= 0 &&  // Have we configured RED?
-      fragmentation && fragmentation->fragmentationVectorSize > 1 &&
-      !marker_bit) {
-    if (timestampOffset <= 0x3fff) {
-      if (fragmentation->fragmentationVectorSize != 2) {
-        // we only support 2 codecs when using RED
-        return false;
-      }
-      // only 0x80 if we have multiple blocks
-      data_buffer[rtpHeaderLength++] =
-          0x80 + fragmentation->fragmentationPlType[1];
-      size_t blockLength = fragmentation->fragmentationLength[1];
-
-      // sanity blockLength
-      if (blockLength > 0x3ff) {  // block length 10 bits 1023 bytes
-        return false;
-      }
-      uint32_t REDheader = (timestampOffset << 10) + blockLength;
-      ByteWriter<uint32_t>::WriteBigEndian(data_buffer + rtpHeaderLength,
-                                           REDheader);
-      rtpHeaderLength += 3;
-
-      data_buffer[rtpHeaderLength++] = fragmentation->fragmentationPlType[0];
-      // copy the RED data
-      memcpy(data_buffer + rtpHeaderLength,
-             payload_data + fragmentation->fragmentationOffset[1],
-             fragmentation->fragmentationLength[1]);
-
-      // copy the normal data
-      memcpy(
-          data_buffer + rtpHeaderLength + fragmentation->fragmentationLength[1],
-          payload_data + fragmentation->fragmentationOffset[0],
-          fragmentation->fragmentationLength[0]);
-
-      payload_size = fragmentation->fragmentationLength[0] +
-                     fragmentation->fragmentationLength[1];
-    } else {
-      // silence for too long send only new data
-      data_buffer[rtpHeaderLength++] = fragmentation->fragmentationPlType[0];
-      memcpy(data_buffer + rtpHeaderLength,
-             payload_data + fragmentation->fragmentationOffset[0],
-             fragmentation->fragmentationLength[0]);
-
-      payload_size = fragmentation->fragmentationLength[0];
-    }
-  } else {
-    if (fragmentation && fragmentation->fragmentationVectorSize > 0) {
-      // use the fragment info if we have one
-      data_buffer[rtpHeaderLength++] = fragmentation->fragmentationPlType[0];
-      memcpy(data_buffer + rtpHeaderLength,
-             payload_data + fragmentation->fragmentationOffset[0],
-             fragmentation->fragmentationLength[0]);
-
-      payload_size = fragmentation->fragmentationLength[0];
-    } else {
-      memcpy(data_buffer + rtpHeaderLength, payload_data, payload_size);
-    }
+    uint8_t* payload = packet->AllocatePayload(payload_size);
+    if (!payload)
+      return false;
+    memcpy(payload, payload_data, payload_size);
   }
 
   {
     rtc::CritScope cs(&send_audio_critsect_);
     last_payload_type_ = payload_type;
   }
-  // Update audio level extension, if included.
-  size_t packetSize = payload_size + rtpHeaderLength;
-  RtpUtility::RtpHeaderParser rtp_parser(data_buffer, packetSize);
-  RTPHeader rtp_header;
-  rtp_parser.Parse(&rtp_header);
-  rtp_sender_->UpdateAudioLevel(data_buffer, packetSize, rtp_header,
-                                (frame_type == kAudioFrameSpeech),
-                                audio_level_dbov);
+  // Allocate sequence number.
+  rtp_sender_->PrepareToSend(packet.get());
   TRACE_EVENT_ASYNC_END2("webrtc", "Audio", capture_timestamp, "timestamp",
-                         rtp_sender_->Timestamp(), "seqnum",
-                         rtp_sender_->SequenceNumber());
+                         packet->Timestamp(), "seqnum",
+                         packet->SequenceNumber());
   bool send_result = rtp_sender_->SendToNetwork(
-      data_buffer, payload_size, rtpHeaderLength, rtc::TimeMillis(),
-      kAllowRetransmission, RtpPacketSender::kHighPriority);
+      std::move(packet), kAllowRetransmission, RtpPacketSender::kHighPriority);
   if (first_packet_sent_()) {
     LOG(LS_INFO) << "First audio RTP packet sent to pacer";
   }
   return send_result;
 }
 
 // Audio level magnitude and voice activity flag are set for each RTP packet
 int32_t RTPSenderAudio::SetAudioLevel(uint8_t level_dbov) {
   if (level_dbov > 127) {
     return -1;
@@ skipping 36 matching lines @@
     }
   }
   return AddDTMF(key, time_ms, level);
 }
 
 bool RTPSenderAudio::SendTelephoneEventPacket(bool ended,
                                               int8_t dtmf_payload_type,
                                               uint32_t dtmf_timestamp,
                                               uint16_t duration,
                                               bool marker_bit) {
-  uint8_t dtmfbuffer[IP_PACKET_SIZE];
   uint8_t send_count = 1;
   bool result = true;
 
   if (ended) {
     // resend last packet in an event 3 times
     send_count = 3;
   }
   do {
     // Send DTMF data
-    int32_t header_length = rtp_sender_->BuildRtpHeader(
-        dtmfbuffer, dtmf_payload_type, marker_bit, dtmf_timestamp,
-        clock_->TimeInMilliseconds());
-    if (header_length <= 0)
+    std::unique_ptr<RtpPacketToSend> packet = rtp_sender_->AllocatePacket(true);
+    if (!packet)
       return false;
-
-    // reset CSRC and X bit
-    dtmfbuffer[0] &= 0xe0;
+    packet->SetPayloadType(dtmf_payload_type);
+    packet->SetMarker(marker_bit);
+    packet->SetTimestamp(dtmf_timestamp);
+    packet->set_capture_time_ms(clock_->TimeInMilliseconds());
 
     // Create DTMF data
     /*    From RFC 2833:
 
      0                   1                   2                   3
      0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
     |     event     |E|R| volume    |          duration             |
     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
     */
     // R bit always cleared
     uint8_t R = 0x00;
     uint8_t volume = dtmf_level_;
 
     // First packet un-ended
     uint8_t E = ended ? 0x80 : 0x00;
 
+    uint8_t* dtmfbuffer = packet->AllocatePayload(4);
+    RTC_DCHECK(dtmfbuffer);
     // First byte is Event number, equals key number
-    dtmfbuffer[12] = dtmf_key_;
-    dtmfbuffer[13] = E | R | volume;
-    ByteWriter<uint16_t>::WriteBigEndian(dtmfbuffer + 14, duration);
+    dtmfbuffer[0] = dtmf_key_;
+    dtmfbuffer[1] = E | R | volume;
+    ByteWriter<uint16_t>::WriteBigEndian(dtmfbuffer + 2, duration);
+    if (!rtp_sender_->PrepareToSend(packet.get()))
+      return false;
 
-    TRACE_EVENT_INSTANT2(
-        TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"), "Audio::SendTelephoneEvent",
-        "timestamp", dtmf_timestamp, "seqnum", rtp_sender_->SequenceNumber());
-    result = rtp_sender_->SendToNetwork(dtmfbuffer, 4, 12, rtc::TimeMillis(),
-                                        kAllowRetransmission,
+    TRACE_EVENT_INSTANT2(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"),
+                         "Audio::SendTelephoneEvent", "timestamp",
+                         dtmf_timestamp, "seqnum", packet->SequenceNumber());
+    result = rtp_sender_->SendToNetwork(std::move(packet), kAllowRetransmission,
                                         RtpPacketSender::kHighPriority);
     send_count--;
   } while (send_count > 0 && result);
 
   return result;
 }
 }  // namespace webrtc