Index: webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.cc |
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.cc b/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.cc |
index acc5926e5b5829db837678226958890b0f90ebb8..e3237948fb323e3b6a152d82f03b6338e5e7690c 100644 |
--- a/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.cc |
+++ b/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.cc |
@@ -212,9 +212,10 @@ int32_t RTPReceiverAudio::ParseAudioCodecSpecific( |
size_t payload_length, |
const AudioPayload& audio_specific, |
bool is_red) { |
- |
if (payload_length == 0) { |
minyue-webrtc
2017/05/10 13:03:18
I tried it, and I am afraid OnReceivedPayloadData(
|
- return 0; |
+ rtp_header->type.Audio.isCNG = false; |
+ rtp_header->frameType = kEmptyFrame; |
+ return data_callback_->OnReceivedPayloadData(nullptr, 0, rtp_header); |
} |
bool telephone_event_packet = |
@@ -239,6 +240,7 @@ int32_t RTPReceiverAudio::ParseAudioCodecSpecific( |
number_of_events = MAX_NUMBER_OF_PARALLEL_TELEPHONE_EVENTS; |
} |
for (size_t n = 0; n < number_of_events; ++n) { |
+ RTC_DCHECK_GE(payload_length, (4 * n) + 2); |
bool end = (payload_data[(4 * n) + 1] & 0x80) ? true : false; |
std::set<uint8_t>::iterator event = |
@@ -291,6 +293,7 @@ int32_t RTPReceiverAudio::ParseAudioCodecSpecific( |
} |
} |
// TODO(holmer): Break this out to have RED parsing handled generically. |
+ RTC_DCHECK_GT(payload_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]; |