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Extracted a method for sending padded data.

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BUG=
TEST=Ran vie_auto_test and voe_auto_test standard tests.

Review URL: https://webrtc-codereview.appspot.com/605004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2332 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
phoglund@webrtc.org
2012-05-31 10:47:35 +00:00
parent bb24b14c8a
commit baaf2434a7
2 changed files with 65 additions and 56 deletions
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115
src/modules/rtp_rtcp/source/rtp_sender.cc
View File

@@ -461,80 +461,85 @@ WebRtc_Word32 RTPSender::CheckPayloadType(const WebRtc_Word8 payloadType,
}
WebRtc_Word32
RTPSender::SendOutgoingData(const FrameType frameType,
const WebRtc_Word8 payloadType,
const WebRtc_UWord32 captureTimeStamp,
const WebRtc_UWord8* payloadData,
const WebRtc_UWord32 payloadSize,
RTPSender::SendOutgoingData(const FrameType frame_type,
const WebRtc_Word8 payload_type,
const WebRtc_UWord32 capture_timestamp,
const WebRtc_UWord8* payload_data,
const WebRtc_UWord32 payload_size,
const RTPFragmentationHeader* fragmentation,
VideoCodecInformation* codecInfo,
const RTPVideoTypeHeader* rtpTypeHdr)
VideoCodecInformation* codec_info,
const RTPVideoTypeHeader* rtp_type_hdr)
{
{
// Drop this packet if we're not sending media packets
// Drop this packet if we're not sending media packets.
CriticalSectionScoped cs(_sendCritsect);
if (!_sendingMedia)
{
return 0;
}
}
RtpVideoCodecTypes videoType = kRtpNoVideo;
if(CheckPayloadType(payloadType, videoType) != 0)
RtpVideoCodecTypes video_type = kRtpNoVideo;
if (CheckPayloadType(payload_type, video_type) != 0)
{
WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id, "%s invalid argument failed to find payloadType:%d", __FUNCTION__, payloadType);
WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id,
"%s invalid argument failed to find payloadType:%d",
__FUNCTION__, payload_type);
return -1;
}
if(_audioConfigured)
if (_audioConfigured)
{
// assert video frameTypes
assert(frameType == kAudioFrameSpeech ||
frameType == kAudioFrameCN ||
frameType == kFrameEmpty);
assert(frame_type == kAudioFrameSpeech ||
frame_type == kAudioFrameCN ||
frame_type == kFrameEmpty);
return _audio->SendAudio(frameType, payloadType, captureTimeStamp, payloadData, payloadSize,fragmentation);
} else
{
// Assert on audio frameTypes.
assert(frameType != kAudioFrameSpeech &&
frameType != kAudioFrameCN);
return _audio->SendAudio(frame_type, payload_type, capture_timestamp,
payload_data, payload_size,fragmentation);
} else {
assert(frame_type != kAudioFrameSpeech &&
frame_type != kAudioFrameCN);
// If the encoder generate an empty frame send pading.
if (frameType == kFrameEmpty) {
// 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 = BitrateNow();
int bitrate_diff = _targetSendBitrate * 1000 - current_bitrate;
if (bitrate_diff > 0) {
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 = _targetSendBitrate * 25; // 1000 / 8 / 5
if (bytes > bytes_cap) {
bytes = bytes_cap;
}
}
// Send pading data.
return SendPadData(payloadType, captureTimeStamp, bytes);
}
return 0;
if (frame_type == kFrameEmpty) {
return SendPaddingAccordingToBitrate(payload_type, capture_timestamp);
}
return _video->SendVideo(videoType,
frameType,
payloadType,
captureTimeStamp,
payloadData,
payloadSize,
return _video->SendVideo(video_type,
frame_type,
payload_type,
capture_timestamp,
payload_data,
payload_size,
fragmentation,
codecInfo,
rtpTypeHdr);
codec_info,
rtp_type_hdr);
}
}
WebRtc_Word32 RTPSender::SendPaddingAccordingToBitrate(
WebRtc_Word8 payload_type,
WebRtc_UWord32 capture_timestamp) {
// 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 = BitrateNow();
int bitrate_diff = _targetSendBitrate * 1000 - current_bitrate;
if (bitrate_diff > 0) {
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 = _targetSendBitrate * 25; // 1000 / 8 / 5
if (bytes > bytes_cap) {
bytes = bytes_cap;
}
}
// Send padding data.
return SendPadData(payload_type, capture_timestamp, bytes);
}
return 0;
}
WebRtc_Word32 RTPSender::SendPadData(WebRtc_Word8 payload_type,
WebRtc_UWord32 capture_timestamp,
WebRtc_Word32 bytes) {
@@ -879,7 +884,7 @@ void RTPSender::ProcessSendToNetwork() {
_payloadBytesSent += bytes_sent - rtp_header.header.headerLength;
}
}
}
}
}
WebRtc_Word32
@@ -1186,7 +1191,7 @@ void RTPSender::UpdateTransmissionTimeOffset(
"Failed to update transmission time offset, no id.");
return;
}
// Verify first byte in block.
const WebRtc_UWord8 first_block_byte = (id << 4) + 2;
if (rtp_packet[block_pos] != first_block_byte) {

6
src/modules/rtp_rtcp/source/rtp_sender.h
View File

@@ -288,11 +288,15 @@ private:
void UpdateNACKBitRate(const WebRtc_UWord32 bytes,
const WebRtc_UWord32 now);
WebRtc_Word32 SendPaddingAccordingToBitrate(
WebRtc_Word8 payload_type,
WebRtc_UWord32 capture_timestamp);
WebRtc_Word32 _id;
const bool _audioConfigured;
RTPSenderAudio* _audio;
RTPSenderVideo* _video;
CriticalSectionWrapper* _sendCritsect;
CriticalSectionWrapper* _transportCritsect;