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NetEq: Limit payload size for replacement audio input

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With this fix, the size of the fake encoded payload is limited to 120
ms at 48000 samples/second.

BUG=webrtc:7467

Review-Url: https://codereview.webrtc.org/2838353002
Cr-Commit-Position: refs/heads/master@{#17891}
This commit is contained in:
henrik.lundin
2017-04-26 08:23:35 -07:00
committed by Commit bot
parent afcf7f5591
commit 65881de6c8
1 changed files with 9 additions and 4 deletions
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13
webrtc/modules/audio_coding/neteq/tools/neteq_replacement_input.cc
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@ -85,13 +85,18 @@ void NetEqReplacementInput::ReplacePacket() {
rtc::Optional<RTPHeader> next_hdr = source_->NextHeader();
RTC_DCHECK(next_hdr);
uint8_t payload[12];
RTC_DCHECK_LE(last_frame_size_timestamps_, 120 * 48);
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);
FakeDecodeFromFile::PrepareEncoded(packet_->header.timestamp,
input_frame_size_timestamps,
packet_->payload.size(), payload);