PacketBuffer is now ref counted.
Since all FrameObjects have a reference to its PacketBuffer and since
the PacketBuffer can be thrown away at any moment the PacketBuffer
has to be ref counted in order to avoid FrameObjects dereferencing a potentially
destroyed object.
BUG=webrtc:5514
R=danilchap@webrtc.org, mflodman@webrtc.org, stefan@webrtc.org
Review URL: https://codereview.webrtc.org/2199133004 .
Cr-Commit-Position: refs/heads/master@{#13725}
diff --git a/webrtc/modules/video_coding/packet_buffer.cc b/webrtc/modules/video_coding/packet_buffer.cc
index df17350..0d36b9c 100644
--- a/webrtc/modules/video_coding/packet_buffer.cc
+++ b/webrtc/modules/video_coding/packet_buffer.cc
@@ -12,7 +12,9 @@
#include <algorithm>
#include <limits>
+#include <utility>
+#include "webrtc/base/atomicops.h"
#include "webrtc/base/checks.h"
#include "webrtc/base/logging.h"
#include "webrtc/modules/video_coding/frame_object.h"
@@ -21,10 +23,19 @@
namespace webrtc {
namespace video_coding {
+rtc::scoped_refptr<PacketBuffer> PacketBuffer::Create(
+ Clock* clock,
+ size_t start_buffer_size,
+ size_t max_buffer_size,
+ OnReceivedFrameCallback* received_frame_callback) {
+ return rtc::scoped_refptr<PacketBuffer>(new PacketBuffer(
+ clock, start_buffer_size, max_buffer_size, received_frame_callback));
+}
+
PacketBuffer::PacketBuffer(Clock* clock,
size_t start_buffer_size,
size_t max_buffer_size,
- OnCompleteFrameCallback* frame_callback)
+ OnReceivedFrameCallback* received_frame_callback)
: clock_(clock),
size_(start_buffer_size),
max_size_(max_buffer_size),
@@ -33,13 +44,15 @@
first_packet_received_(false),
data_buffer_(start_buffer_size),
sequence_buffer_(start_buffer_size),
- reference_finder_(frame_callback) {
+ received_frame_callback_(received_frame_callback) {
RTC_DCHECK_LE(start_buffer_size, max_buffer_size);
// Buffer size must always be a power of 2.
RTC_DCHECK((start_buffer_size & (start_buffer_size - 1)) == 0);
RTC_DCHECK((max_buffer_size & (max_buffer_size - 1)) == 0);
}
+PacketBuffer::~PacketBuffer() {}
+
bool PacketBuffer::InsertPacket(const VCMPacket& packet) {
rtc::CritScope lock(&crit_);
uint16_t seq_num = packet.seqNum;
@@ -69,12 +82,6 @@
if (AheadOf(seq_num, last_seq_num_))
last_seq_num_ = seq_num;
- // If this is a padding or FEC packet, don't insert it.
- if (packet.sizeBytes == 0) {
- reference_finder_.PaddingReceived(packet.seqNum);
- return true;
- }
-
sequence_buffer_[index].frame_begin = packet.isFirstPacket;
sequence_buffer_[index].frame_end = packet.markerBit;
sequence_buffer_[index].seq_num = packet.seqNum;
@@ -169,7 +176,8 @@
std::unique_ptr<RtpFrameObject> frame(
new RtpFrameObject(this, start_seq_num, seq_num, frame_size,
max_nack_count, clock_->TimeInMilliseconds()));
- reference_finder_.ManageFrame(std::move(frame));
+
+ received_frame_callback_->OnReceivedFrame(std::move(frame));
}
index = (index + 1) % size_;
@@ -239,5 +247,17 @@
first_packet_received_ = false;
}
+int PacketBuffer::AddRef() const {
+ return rtc::AtomicOps::Increment(&ref_count_);
+}
+
+int PacketBuffer::Release() const {
+ int count = rtc::AtomicOps::Decrement(&ref_count_);
+ if (!count) {
+ delete this;
+ }
+ return count;
+}
+
} // namespace video_coding
} // namespace webrtc