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Remove use of RTPFragmentationHeader from RTPSenderAudio

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The RTPFragmentationHeader was used when sending audio using RED
for loss protection. This feature has been deprecated and
gradually removed. This cl removes remnants of support from
the RTP send path.

Bug: webrtc:6471
Change-Id: Ia1249047b09c16f79498827f74c2ce07aa38b8f7
Reviewed-on: https://webrtc-review.googlesource.com/16427
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20473}
This commit is contained in:
Niels Möller
2017-10-27 10:51:20 +02:00
committed by Commit Bot
parent ce50239558
commit 90397d9aa9
3 changed files with 10 additions and 21 deletions
Download Patch File Download Diff File Expand all files Collapse all files

6
modules/rtp_rtcp/source/rtp_sender.cc
View File

@ -417,9 +417,11 @@ bool RTPSender::SendOutgoingData(FrameType frame_type,
if (audio_configured_) {
TRACE_EVENT_ASYNC_STEP1("webrtc", "Audio", rtp_timestamp, "Send", "type",
FrameTypeToString(frame_type));
// The only known way to produce of RTPFragmentationHeader for audio is
// to use the AudioCodingModule directly.
RTC_DCHECK(fragmentation == nullptr);
result = audio_->SendAudio(frame_type, payload_type, rtp_timestamp,
payload_data, payload_size, fragmentation);
payload_data, payload_size);
} else {
TRACE_EVENT_ASYNC_STEP1("webrtc", "Video", capture_time_ms,
"Send", "type", FrameTypeToString(frame_type));

22
modules/rtp_rtcp/source/rtp_sender_audio.cc
View File

@ -119,8 +119,7 @@ bool RTPSenderAudio::SendAudio(FrameType frame_type,
int8_t payload_type,
uint32_t rtp_timestamp,
const uint8_t* payload_data,
size_t payload_size,
const RTPFragmentationHeader* fragmentation) {
size_t payload_size) {
// From RFC 4733:
// A source has wide latitude as to how often it sends event updates. A
// natural interval is the spacing between non-event audio packets. [...]
@ -223,21 +222,10 @@ bool RTPSenderAudio::SendAudio(FrameType frame_type,
packet->SetExtension<AudioLevel>(frame_type == kAudioFrameSpeech,
audio_level_dbov);
if (fragmentation && fragmentation->fragmentationVectorSize > 0) {
// Use the fragment info if we have one.
uint8_t* payload =
packet->AllocatePayload(1 + fragmentation->fragmentationLength[0]);
if (!payload) // Too large payload buffer.
return false;
payload[0] = fragmentation->fragmentationPlType[0];
memcpy(payload + 1, payload_data + fragmentation->fragmentationOffset[0],
fragmentation->fragmentationLength[0]);
} else {
uint8_t* payload = packet->AllocatePayload(payload_size);
if (!payload) // Too large payload buffer.
return false;
memcpy(payload, payload_data, payload_size);
}
uint8_t* payload = packet->AllocatePayload(payload_size);
if (!payload) // Too large payload buffer.
return false;
memcpy(payload, payload_data, payload_size);
if (!rtp_sender_->AssignSequenceNumber(packet.get()))
return false;

3
modules/rtp_rtcp/source/rtp_sender_audio.h
View File

@ -39,8 +39,7 @@ class RTPSenderAudio {
int8_t payload_type,
uint32_t capture_timestamp,
const uint8_t* payload_data,
size_t payload_size,
const RTPFragmentationHeader* fragmentation);
size_t payload_size);
// Store the audio level in dBov for
// header-extension-for-audio-level-indication.