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Handle padded audio packets correctly

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RTP packets can be padded with extra data at the end of the payload. The usable
payload length of the packet should then be reduced with the padding length,
since the padding must be discarded. This was not the case; instead, the entire
payload, including padding data, was forwarded to the audio channel and in the
end to the decoder.

A special case of padding is packets which are empty except for the padding.
That is, they carry no usable payload. These packets are sometimes used for
probing the network and were discarded in
RTPReceiverAudio::ParseAudioCodecSpecific. The result is that NetEq never sees
those empty packets, just the holes in the sequence number series; this can
throw off the target buffer calculations.

With this change, the empty (after removing the padding) packets are let through,
all the way down to NetEq, to a new method called NetEq::InsertEmptyPacket. This
method notifies the DelayManager that an empty packet was received.

BUG=webrtc:7610, webrtc:7625

Review-Url: https://codereview.webrtc.org/2870043003
Cr-Commit-Position: refs/heads/master@{#18083}
This commit is contained in:
henrik.lundin
2017-05-10 07:38:01 -07:00
committed by Commit bot
parent 423a288a12
commit b8c55b15a3
11 changed files with 113 additions and 8 deletions
Download Patch File Download Diff File Expand all files Collapse all files

22
webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.cc
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@ -212,9 +212,13 @@ int32_t RTPReceiverAudio::ParseAudioCodecSpecific(
size_t payload_length,
const AudioPayload& audio_specific,
bool is_red) {
if (payload_length == 0) {
return 0;
RTC_DCHECK_GE(payload_length, rtp_header->header.paddingLength);
const size_t payload_data_length =
payload_length - rtp_header->header.paddingLength;
if (payload_data_length == 0) {
rtp_header->type.Audio.isCNG = false;
rtp_header->frameType = kEmptyFrame;
return data_callback_->OnReceivedPayloadData(nullptr, 0, rtp_header);
}
bool telephone_event_packet =
@ -229,16 +233,17 @@ int32_t RTPReceiverAudio::ParseAudioCodecSpecific(
// | event |E|R| volume | duration |
// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
//
if (payload_length % 4 != 0) {
if (payload_data_length % 4 != 0) {
return -1;
}
size_t number_of_events = payload_length / 4;
size_t number_of_events = payload_data_length / 4;
// sanity
if (number_of_events >= MAX_NUMBER_OF_PARALLEL_TELEPHONE_EVENTS) {
number_of_events = MAX_NUMBER_OF_PARALLEL_TELEPHONE_EVENTS;
}
for (size_t n = 0; n < number_of_events; ++n) {
RTC_DCHECK_GE(payload_data_length, (4 * n) + 2);
bool end = (payload_data[(4 * n) + 1] & 0x80) ? true : false;
std::set<uint8_t>::iterator event =
@ -291,17 +296,18 @@ int32_t RTPReceiverAudio::ParseAudioCodecSpecific(
}
}
// TODO(holmer): Break this out to have RED parsing handled generically.
RTC_DCHECK_GT(payload_data_length, 0);
if (is_red && !(payload_data[0] & 0x80)) {
// we recive only one frame packed in a RED packet remove the RED wrapper
rtp_header->header.payloadType = payload_data[0];
// only one frame in the RED strip the one byte to help NetEq
return data_callback_->OnReceivedPayloadData(
payload_data + 1, payload_length - 1, rtp_header);
payload_data + 1, payload_data_length - 1, rtp_header);
}
rtp_header->type.Audio.channel = audio_specific.channels;
return data_callback_->OnReceivedPayloadData(
payload_data, payload_length, rtp_header);
return data_callback_->OnReceivedPayloadData(payload_data,
payload_data_length, rtp_header);
}
} // namespace webrtc