| Index: webrtc/modules/audio_coding/neteq/delay_manager.cc
|
| diff --git a/webrtc/modules/audio_coding/neteq/delay_manager.cc b/webrtc/modules/audio_coding/neteq/delay_manager.cc
|
| index 806d02b8deb538184d542e419b49c5fec37b5da9..af49f00f8af0be85d31da95933085e55110241c6 100644
|
| --- a/webrtc/modules/audio_coding/neteq/delay_manager.cc
|
| +++ b/webrtc/modules/audio_coding/neteq/delay_manager.cc
|
| @@ -15,6 +15,7 @@
|
|
|
| #include <algorithm> // max, min
|
|
|
| +#include "webrtc/base/safe_conversions.h"
|
| #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
|
| #include "webrtc/modules/audio_coding/neteq/delay_peak_detector.h"
|
| #include "webrtc/modules/include/module_common_types.h"
|
| @@ -93,10 +94,11 @@ int DelayManager::Update(uint16_t sequence_number,
|
| packet_len_ms = packet_len_ms_;
|
| } else {
|
| // Calculate timestamps per packet and derive packet length in ms.
|
| - int packet_len_samp =
|
| + int64_t packet_len_samp =
|
| static_cast<uint32_t>(timestamp - last_timestamp_) /
|
| static_cast<uint16_t>(sequence_number - last_seq_no_);
|
| - packet_len_ms = (1000 * packet_len_samp) / sample_rate_hz;
|
| + packet_len_ms =
|
| + rtc::checked_cast<int>(1000 * packet_len_samp / sample_rate_hz);
|
| }
|
|
|
| if (packet_len_ms > 0) {
|
|
|