Index: webrtc/modules/rtp_rtcp/source/rtp_sender.h |
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender.h b/webrtc/modules/rtp_rtcp/source/rtp_sender.h |
index f068ae35701f7f07b49612169d0459626afe5ef5..14ec3c1f68411f3846721bb110b5ffbc84fc8362 100644 |
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender.h |
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender.h |
@@ -35,9 +35,10 @@ |
namespace webrtc { |
class RateLimiter; |
+class RtcEventLog; |
+class RtpPacketToSend; |
class RTPSenderAudio; |
class RTPSenderVideo; |
-class RtcEventLog; |
class RTPSenderInterface { |
public: |
@@ -277,12 +278,16 @@ class RTPSender : public RTPSenderInterface { |
uint32_t Timestamp() const override; |
uint32_t SSRC() const override; |
+ // Deprecated. Create RtpPacketToSend instead and use next function. |
bool SendToNetwork(uint8_t* data_buffer, |
size_t payload_length, |
size_t rtp_header_length, |
int64_t capture_time_ms, |
StorageType storage, |
RtpPacketSender::Priority priority) override; |
+ bool SendToNetwork(std::unique_ptr<RtpPacketToSend> packet, |
+ StorageType storage, |
+ RtpPacketSender::Priority priority); |
// Audio. |
@@ -359,9 +364,7 @@ class RTPSender : public RTPSenderInterface { |
const std::vector<uint32_t>& csrcs) const |
EXCLUSIVE_LOCKS_REQUIRED(send_critsect_); |
- bool PrepareAndSendPacket(uint8_t* buffer, |
- size_t length, |
- int64_t capture_time_ms, |
+ bool PrepareAndSendPacket(std::unique_ptr<RtpPacketToSend> packet, |
bool send_over_rtx, |
bool is_retransmit, |
int probe_cluster_id); |
@@ -370,14 +373,10 @@ class RTPSender : public RTPSenderInterface { |
// return a larger value that their argument. |
size_t TrySendRedundantPayloads(size_t bytes, int probe_cluster_id); |
- void BuildPaddingPacket(uint8_t* packet, |
- size_t header_length, |
- size_t padding_length); |
+ std::unique_ptr<RtpPacketToSend> BuildRtxPacket( |
+ const RtpPacketToSend& packet); |
- bool BuildRtxPacket(uint8_t* buffer, size_t* length, uint8_t* buffer_rtx); |
- |
- bool SendPacketToNetwork(const uint8_t* packet, |
- size_t size, |
+ bool SendPacketToNetwork(const RtpPacketToSend& packet, |
const PacketOptions& options); |
void UpdateDelayStatistics(int64_t capture_time_ms, int64_t now_ms); |
@@ -394,19 +393,8 @@ class RTPSender : public RTPSenderInterface { |
size_t* position) const |
EXCLUSIVE_LOCKS_REQUIRED(send_critsect_); |
- void UpdateTransmissionTimeOffset(uint8_t* rtp_packet, |
- size_t rtp_packet_length, |
- const RTPHeader& rtp_header, |
- int64_t time_diff_ms) const; |
- void UpdateAbsoluteSendTime(uint8_t* rtp_packet, |
- size_t rtp_packet_length, |
- const RTPHeader& rtp_header, |
- int64_t now_ms) const; |
- |
- bool UpdateTransportSequenceNumber(uint8_t* rtp_packet, |
- size_t rtp_packet_length, |
- const RTPHeader& rtp_header, |
- int* sequence_number) const; |
+ bool UpdateTransportSequenceNumber(RtpPacketToSend* packet, |
+ int* packet_id) const; |
void UpdatePlayoutDelayLimits(uint8_t* rtp_packet, |
size_t rtp_packet_length, |
@@ -414,14 +402,10 @@ class RTPSender : public RTPSenderInterface { |
uint16_t min_playout_delay, |
uint16_t max_playout_delay) const; |
- bool AllocateTransportSequenceNumber(int* packet_id) const; |
- |
- void UpdateRtpStats(const uint8_t* buffer, |
- size_t packet_length, |
- const RTPHeader& header, |
+ void UpdateRtpStats(const RtpPacketToSend& packet, |
bool is_rtx, |
bool is_retransmit); |
- bool IsFecPacket(const uint8_t* buffer, const RTPHeader& header) const; |
+ bool IsFecPacket(const RtpPacketToSend& packet) const; |
Clock* const clock_; |
const int64_t clock_delta_ms_; |
@@ -458,7 +442,7 @@ class RTPSender : public RTPSenderInterface { |
PlayoutDelayOracle playout_delay_oracle_; |
bool playout_delay_active_ GUARDED_BY(send_critsect_); |
- RTPPacketHistory packet_history_; |
+ RtpPacketHistory packet_history_; |
// Statistics |
rtc::CriticalSection statistics_crit_; |