Chromium Code Reviews| 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]; |