Index: webrtc/modules/audio_coding/neteq/tools/neteq_replacement_input.cc |
diff --git a/webrtc/modules/audio_coding/neteq/tools/neteq_replacement_input.cc b/webrtc/modules/audio_coding/neteq/tools/neteq_replacement_input.cc |
index 9b550cbb2bf015ae58510fe59e52488cb2068f7d..00814f1785a34887bfa1bccb570801058876f877 100644 |
--- a/webrtc/modules/audio_coding/neteq/tools/neteq_replacement_input.cc |
+++ b/webrtc/modules/audio_coding/neteq/tools/neteq_replacement_input.cc |
@@ -86,12 +86,16 @@ void NetEqReplacementInput::ReplacePacket() { |
RTC_DCHECK(next_hdr); |
uint8_t payload[12]; |
uint32_t input_frame_size_timestamps = last_frame_size_timestamps_; |
- if (next_hdr->sequenceNumber == packet_->header.sequenceNumber + 1) { |
- // Packets are in order. |
- input_frame_size_timestamps = |
- next_hdr->timestamp - packet_->header.timestamp; |
+ const uint32_t timestamp_diff = |
+ next_hdr->timestamp - packet_->header.timestamp; |
+ if (next_hdr->sequenceNumber == packet_->header.sequenceNumber + 1 && |
+ timestamp_diff <= 120 * 48) { |
+ // Packets are in order and the timestamp diff is less than 5760 samples. |
+ // Accept the timestamp diff as a valid frame size. |
+ input_frame_size_timestamps = timestamp_diff; |
last_frame_size_timestamps_ = input_frame_size_timestamps; |
} |
+ RTC_DCHECK_LE(input_frame_size_timestamps, 120 * 48); |
AleBzk
2017/04/26 14:26:48
I'd rather do this before line 88 and on last_fram
hlundin-webrtc
2017/04/26 14:32:32
I see your point. But I think that we are not as m
|
FakeDecodeFromFile::PrepareEncoded(packet_->header.timestamp, |
input_frame_size_timestamps, |
packet_->payload.size(), payload); |