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Remove old padding path in RTPSender.

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Removing RTPSender::SendPaddingAccordingToBitrate() as well as a couple
of arguments from SendPadData().

BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14989004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6703 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
pbos@webrtc.org 2014-07-16 09:37:29 +00:00
parent efb81d8d1f
commit 63c60ed224
2 changed files with 16 additions and 77 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

86
webrtc/modules/rtp_rtcp/source/rtp_sender.cc
View File

@ -430,14 +430,9 @@ int32_t RTPSender::SendOutgoingData(
"Send", "type", FrameTypeToString(frame_type));
assert(frame_type != kAudioFrameSpeech && frame_type != kAudioFrameCN);
if (frame_type == kFrameEmpty) {
if (paced_sender_->Enabled()) {
// Padding is driven by the pacer and not by the encoder.
return 0;
}
return SendPaddingAccordingToBitrate(payload_type, capture_timestamp,
capture_time_ms) ? 0 : -1;
}
if (frame_type == kFrameEmpty)
return 0;
ret_val = video_->SendVideo(video_type, frame_type, payload_type,
capture_timestamp, capture_time_ms,
payload_data, payload_size,
@ -475,45 +470,6 @@ int RTPSender::SendRedundantPayloads(int payload_type, int bytes_to_send) {
return bytes_to_send - bytes_left;
}
bool RTPSender::SendPaddingAccordingToBitrate(
int8_t payload_type, uint32_t capture_timestamp,
int64_t capture_time_ms) {
// Current bitrate since last estimate(1 second) averaged with the
// estimate since then, to get the most up to date bitrate.
uint32_t current_bitrate = bitrate_sent_.BitrateNow();
uint32_t target_bitrate = GetTargetBitrate();
int bitrate_diff = target_bitrate - current_bitrate;
if (bitrate_diff <= 0) {
return true;
}
int bytes = 0;
if (current_bitrate == 0) {
// Start up phase. Send one 33.3 ms batch to start with.
bytes = (bitrate_diff / 8) / 30;
} else {
bytes = (bitrate_diff / 8);
// Cap at 200 ms of target send data.
int bytes_cap = target_bitrate / 1000 * 25; // 1000 / 8 / 5.
if (bytes > bytes_cap) {
bytes = bytes_cap;
}
}
uint32_t timestamp;
{
CriticalSectionScoped cs(send_critsect_);
// Add the random RTP timestamp offset and store the capture time for
// later calculation of the send time offset.
timestamp = start_timestamp_ + capture_timestamp;
timestamp_ = timestamp;
capture_time_ms_ = capture_time_ms;
last_timestamp_time_ms_ = clock_->TimeInMilliseconds();
}
int bytes_sent = SendPadData(payload_type, timestamp, capture_time_ms,
bytes, false, false);
// We did not manage to send all bytes. Comparing with 31 due to modulus 32.
return bytes - bytes_sent < 31;
}
int RTPSender::BuildPaddingPacket(uint8_t* packet, int header_length,
int32_t bytes) {
int padding_bytes_in_packet = kMaxPaddingLength;
@ -536,9 +492,7 @@ int RTPSender::BuildPaddingPacket(uint8_t* packet, int header_length,
int RTPSender::SendPadData(int payload_type,
uint32_t timestamp,
int64_t capture_time_ms,
int32_t bytes,
bool force_full_size_packets,
bool over_rtx) {
int32_t bytes) {
// Drop this packet if we're not sending media packets.
if (!SendingMedia()) {
return bytes;
@ -547,36 +501,29 @@ int RTPSender::SendPadData(int payload_type,
int bytes_sent = 0;
for (; bytes > 0; bytes -= padding_bytes_in_packet) {
// Always send full padding packets.
if (force_full_size_packets && bytes < kMaxPaddingLength)
if (bytes < kMaxPaddingLength)
bytes = kMaxPaddingLength;
if (bytes < kMaxPaddingLength) {
if (force_full_size_packets) {
bytes = kMaxPaddingLength;
} else {
// Round to the nearest multiple of 32.
bytes = (bytes + 16) & 0xffe0;
}
}
if (bytes < 32) {
// Sanity don't send empty packets.
break;
}
uint32_t ssrc;
uint16_t sequence_number;
bool over_rtx;
{
CriticalSectionScoped cs(send_critsect_);
// Only send padding packets following the last packet of a frame,
// indicated by the marker bit.
if (!over_rtx && !last_packet_marker_bit_)
return bytes_sent;
if (rtx_ == kRtxOff) {
// Without RTX we can't send padding in the middle of frames.
if (!last_packet_marker_bit_)
return bytes_sent;
ssrc = ssrc_;
sequence_number = sequence_number_;
++sequence_number_;
over_rtx = false;
} else {
ssrc = ssrc_rtx_;
sequence_number = sequence_number_rtx_;
++sequence_number_rtx_;
over_rtx = true;
}
}
@ -907,6 +854,7 @@ bool RTPSender::IsFecPacket(const uint8_t* buffer,
}
int RTPSender::TimeToSendPadding(int bytes) {
assert(bytes > 0);
int payload_type;
int64_t capture_time_ms;
uint32_t timestamp;
@ -933,12 +881,8 @@ int RTPSender::TimeToSendPadding(int bytes) {
bytes_sent = SendRedundantPayloads(payload_type, bytes);
bytes -= bytes_sent;
if (bytes > 0) {
int padding_sent = SendPadData(payload_type,
timestamp,
capture_time_ms,
bytes,
true,
rtx != kRtxOff);
int padding_sent =
SendPadData(payload_type, timestamp, capture_time_ms, bytes);
bytes_sent += padding_sent;
}
return bytes_sent;

7
webrtc/modules/rtp_rtcp/source/rtp_sender.h
View File

@ -267,9 +267,7 @@ class RTPSender : public RTPSenderInterface, public Bitrate::Observer {
int SendPadData(int payload_type,
uint32_t timestamp,
int64_t capture_time_ms,
int32_t bytes,
bool force_full_size_packets,
bool only_pad_after_markerbit);
int32_t bytes);
// Called on update of RTP statistics.
void RegisterRtpStatisticsCallback(StreamDataCountersCallback* callback);
@ -309,9 +307,6 @@ class RTPSender : public RTPSenderInterface, public Bitrate::Observer {
int SendRedundantPayloads(int payload_type, int bytes);
bool SendPaddingAccordingToBitrate(int8_t payload_type,
uint32_t capture_timestamp,
int64_t capture_time_ms);
int BuildPaddingPacket(uint8_t* packet, int header_length, int32_t bytes);
void BuildRtxPacket(uint8_t* buffer, uint16_t* length,