zqz/platform-external-webrtc
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity

Revert of Add EncodedImageCallback::OnEncodedImage(). (patchset #13 id:280001 of https://codereview.webrtc.org/2089773002/ )

Browse Source 
Reason for revert:
broke internal tests

Original issue's description:
> Add EncodedImageCallback::OnEncodedImage().
>
> OnEncodedImage() is going to replace Encoded(), which is deprecated now.
> The new OnEncodedImage() returns Result struct that contains frame_id,
> which tells the encoder RTP timestamp for the frame.
>
> BUG=chromium:621691
> R=niklas.enbom@webrtc.org, sprang@webrtc.org, stefan@webrtc.org
>
> Committed: https://crrev.com/ad34dbe934d47f88011045671b4aea00dbd5a795
> Cr-Commit-Position: refs/heads/master@{#13613}

TBR=pbos@webrtc.org,mflodman@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,niklas.enbom@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=chromium:621691

Review-Url: https://codereview.webrtc.org/2206743002
Cr-Commit-Position: refs/heads/master@{#13614}
This commit is contained in:
sergeyu
2016-08-02 14:33:16 -07:00
committed by Commit bot
parent ad34dbe934
commit ac4dc2cefe
43 changed files with 380 additions and 472 deletions
Download Patch File Download Diff File Expand all files Collapse all files

56
webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc
View File

@ -145,7 +145,7 @@ bool RTPSenderAudio::MarkerBit(FrameType frame_type, int8_t payload_type) {
return marker_bit;
}
bool RTPSenderAudio::SendAudio(FrameType frame_type,
int32_t RTPSenderAudio::SendAudio(FrameType frame_type,
int8_t payload_type,
uint32_t capture_timestamp,
const uint8_t* payload_data,
@ -195,7 +195,7 @@ bool RTPSenderAudio::SendAudio(FrameType frame_type,
if (packet_size_samples >
(capture_timestamp - dtmf_timestamp_last_sent_)) {
// not time to send yet
return true;
return 0;
}
}
dtmf_timestamp_last_sent_ = capture_timestamp;
@ -228,24 +228,24 @@ bool RTPSenderAudio::SendAudio(FrameType frame_type,
ended, dtmf_payload_type, dtmf_timestamp_,
static_cast<uint16_t>(dtmf_duration_samples), false);
} else {
if (!SendTelephoneEventPacket(ended, dtmf_payload_type, dtmf_timestamp_,
dtmf_duration_samples,
!dtmf_event_first_packet_sent_)) {
return false;
if (SendTelephoneEventPacket(ended, dtmf_payload_type, dtmf_timestamp_,
dtmf_duration_samples,
!dtmf_event_first_packet_sent_) != 0) {
return -1;
}
dtmf_event_first_packet_sent_ = true;
return true;
return 0;
}
}
return true;
return 0;
}
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 0;
}
return false;
return -1;
}
uint8_t data_buffer[IP_PACKET_SIZE];
bool marker_bit = MarkerBit(frame_type, payload_type);
@ -269,11 +269,11 @@ bool RTPSenderAudio::SendAudio(FrameType frame_type,
clock_->TimeInMilliseconds());
}
if (rtpHeaderLength <= 0) {
return false;
return -1;
}
if (max_payload_length < (rtpHeaderLength + payload_size)) {
// Too large payload buffer.
return false;
return -1;
}
if (red_payload_type >= 0 && // Have we configured RED?
fragmentation && fragmentation->fragmentationVectorSize > 1 &&
@ -281,7 +281,7 @@ bool RTPSenderAudio::SendAudio(FrameType frame_type,
if (timestampOffset <= 0x3fff) {
if (fragmentation->fragmentationVectorSize != 2) {
// we only support 2 codecs when using RED
return false;
return -1;
}
// only 0x80 if we have multiple blocks
data_buffer[rtpHeaderLength++] =
@ -290,7 +290,7 @@ bool RTPSenderAudio::SendAudio(FrameType frame_type,
// sanity blockLength
if (blockLength > 0x3ff) { // block length 10 bits 1023 bytes
return false;
return -1;
}
uint32_t REDheader = (timestampOffset << 10) + blockLength;
ByteWriter<uint32_t>::WriteBigEndian(data_buffer + rtpHeaderLength,
@ -349,7 +349,7 @@ bool RTPSenderAudio::SendAudio(FrameType frame_type,
TRACE_EVENT_ASYNC_END2("webrtc", "Audio", capture_timestamp, "timestamp",
rtp_sender_->Timestamp(), "seqnum",
rtp_sender_->SequenceNumber());
bool send_result = rtp_sender_->SendToNetwork(
int32_t send_result = rtp_sender_->SendToNetwork(
data_buffer, payload_size, rtpHeaderLength, rtc::TimeMillis(),
kAllowRetransmission, RtpPacketSender::kHighPriority);
if (first_packet_sent_()) {
@ -403,18 +403,18 @@ int32_t RTPSenderAudio::SendTelephoneEvent(uint8_t key,
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) {
int32_t 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 = 0;
uint8_t sendCount = 1;
int32_t retVal = 0;
if (ended) {
// resend last packet in an event 3 times
send_count = 3;
sendCount = 3;
}
do {
// Send DTMF data
@ -422,7 +422,7 @@ bool RTPSenderAudio::SendTelephoneEventPacket(bool ended,
dtmfbuffer, dtmf_payload_type, marker_bit, dtmf_timestamp,
clock_->TimeInMilliseconds());
if (header_length <= 0)
return false;
return -1;
// reset CSRC and X bit
dtmfbuffer[0] &= 0xe0;
@ -451,12 +451,12 @@ bool RTPSenderAudio::SendTelephoneEventPacket(bool ended,
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(),
retVal = rtp_sender_->SendToNetwork(dtmfbuffer, 4, 12, rtc::TimeMillis(),
kAllowRetransmission,
RtpPacketSender::kHighPriority);
send_count--;
} while (send_count > 0 && result == 0);
sendCount--;
} while (sendCount > 0 && retVal == 0);
return result;
return retVal;
}
} // namespace webrtc