Index: webrtc/modules/audio_coding/neteq/decision_logic_normal.cc |
diff --git a/webrtc/modules/audio_coding/neteq/decision_logic_normal.cc b/webrtc/modules/audio_coding/neteq/decision_logic_normal.cc |
index e985ee0aa394c274ffda9ffc5bcf21429ac676de..d3f6fa6dd45c43da7ad8dec4255f43c01f769b5d 100644 |
--- a/webrtc/modules/audio_coding/neteq/decision_logic_normal.cc |
+++ b/webrtc/modules/audio_coding/neteq/decision_logic_normal.cc |
@@ -27,7 +27,7 @@ namespace webrtc { |
Operations DecisionLogicNormal::GetDecisionSpecialized( |
const SyncBuffer& sync_buffer, |
const Expand& expand, |
- int decoder_frame_length, |
+ size_t decoder_frame_length, |
const RTPHeader* packet_header, |
Modes prev_mode, |
bool play_dtmf, |
@@ -149,7 +149,7 @@ Operations DecisionLogicNormal::ExpectedPacketAvailable(Modes prev_mode, |
Operations DecisionLogicNormal::FuturePacketAvailable( |
const SyncBuffer& sync_buffer, |
const Expand& expand, |
- int decoder_frame_length, |
+ size_t decoder_frame_length, |
Modes prev_mode, |
uint32_t target_timestamp, |
uint32_t available_timestamp, |
@@ -172,9 +172,9 @@ Operations DecisionLogicNormal::FuturePacketAvailable( |
} |
} |
- const int samples_left = static_cast<int>(sync_buffer.FutureLength() - |
- expand.overlap_length()); |
- const int cur_size_samples = samples_left + |
+ const size_t samples_left = |
+ sync_buffer.FutureLength() - expand.overlap_length(); |
+ const size_t cur_size_samples = samples_left + |
packet_buffer_.NumPacketsInBuffer() * decoder_frame_length; |
// If previous was comfort noise, then no merge is needed. |
@@ -205,7 +205,8 @@ Operations DecisionLogicNormal::FuturePacketAvailable( |
// fs_mult_ * 8 = fs / 1000.) |
if (prev_mode == kModeExpand || |
(decoder_frame_length < output_size_samples_ && |
- cur_size_samples > kAllowMergeWithoutExpandMs * fs_mult_ * 8)) { |
+ cur_size_samples > |
+ static_cast<size_t>(kAllowMergeWithoutExpandMs * fs_mult_ * 8))) { |
return kMerge; |
} else if (play_dtmf) { |
// Play DTMF instead of expand. |