| Index: media/cast/net/cast_transport_sender_impl.cc
|
| diff --git a/media/cast/net/cast_transport_sender_impl.cc b/media/cast/net/cast_transport_sender_impl.cc
|
| index f4732193e3e7d694b8ef328d8e83b2d54d31368d..10646ee9f74d552aaba38addf1639c9b99b955a7 100644
|
| --- a/media/cast/net/cast_transport_sender_impl.cc
|
| +++ b/media/cast/net/cast_transport_sender_impl.cc
|
| @@ -38,13 +38,14 @@ int LookupOptionWithDefault(const base::DictionaryValue& options,
|
| }
|
| };
|
|
|
| -int32 GetTransportSendBufferSize(const base::DictionaryValue& options) {
|
| +int32_t GetTransportSendBufferSize(const base::DictionaryValue& options) {
|
| // Socket send buffer size needs to be at least greater than one burst
|
| // size.
|
| - int32 max_burst_size =
|
| + int32_t max_burst_size =
|
| LookupOptionWithDefault(options, kOptionPacerMaxBurstSize,
|
| - kMaxBurstSize) * kMaxIpPacketSize;
|
| - int32 min_send_buffer_size =
|
| + kMaxBurstSize) *
|
| + kMaxIpPacketSize;
|
| + int32_t min_send_buffer_size =
|
| LookupOptionWithDefault(options, kOptionSendBufferMinSize, 0);
|
| return std::max(max_burst_size, min_send_buffer_size);
|
| }
|
| @@ -254,7 +255,7 @@ void EncryptAndSendFrame(const EncodedFrame& frame,
|
| }
|
| } // namespace
|
|
|
| -void CastTransportSenderImpl::InsertFrame(uint32 ssrc,
|
| +void CastTransportSenderImpl::InsertFrame(uint32_t ssrc,
|
| const EncodedFrame& frame) {
|
| if (audio_sender_ && ssrc == audio_sender_->ssrc()) {
|
| EncryptAndSendFrame(frame, &audio_encryptor_, audio_sender_.get());
|
| @@ -266,9 +267,9 @@ void CastTransportSenderImpl::InsertFrame(uint32 ssrc,
|
| }
|
|
|
| void CastTransportSenderImpl::SendSenderReport(
|
| - uint32 ssrc,
|
| + uint32_t ssrc,
|
| base::TimeTicks current_time,
|
| - uint32 current_time_as_rtp_timestamp) {
|
| + uint32_t current_time_as_rtp_timestamp) {
|
| if (audio_sender_ && ssrc == audio_sender_->ssrc()) {
|
| audio_rtcp_session_->SendRtcpFromRtpSender(
|
| current_time, current_time_as_rtp_timestamp,
|
| @@ -283,8 +284,8 @@ void CastTransportSenderImpl::SendSenderReport(
|
| }
|
|
|
| void CastTransportSenderImpl::CancelSendingFrames(
|
| - uint32 ssrc,
|
| - const std::vector<uint32>& frame_ids) {
|
| + uint32_t ssrc,
|
| + const std::vector<uint32_t>& frame_ids) {
|
| if (audio_sender_ && ssrc == audio_sender_->ssrc()) {
|
| audio_sender_->CancelSendingFrames(frame_ids);
|
| } else if (video_sender_ && ssrc == video_sender_->ssrc()) {
|
| @@ -294,8 +295,8 @@ void CastTransportSenderImpl::CancelSendingFrames(
|
| }
|
| }
|
|
|
| -void CastTransportSenderImpl::ResendFrameForKickstart(uint32 ssrc,
|
| - uint32 frame_id) {
|
| +void CastTransportSenderImpl::ResendFrameForKickstart(uint32_t ssrc,
|
| + uint32_t frame_id) {
|
| if (audio_sender_ && ssrc == audio_sender_->ssrc()) {
|
| DCHECK(audio_rtcp_session_);
|
| audio_sender_->ResendFrameForKickstart(
|
| @@ -312,7 +313,7 @@ void CastTransportSenderImpl::ResendFrameForKickstart(uint32 ssrc,
|
| }
|
|
|
| void CastTransportSenderImpl::ResendPackets(
|
| - uint32 ssrc,
|
| + uint32_t ssrc,
|
| const MissingFramesAndPacketsMap& missing_packets,
|
| bool cancel_rtx_if_not_in_list,
|
| const DedupInfo& dedup_info) {
|
| @@ -358,7 +359,7 @@ void CastTransportSenderImpl::SendRawEvents() {
|
| bool CastTransportSenderImpl::OnReceivedPacket(scoped_ptr<Packet> packet) {
|
| const uint8_t* const data = &packet->front();
|
| const size_t length = packet->size();
|
| - uint32 ssrc;
|
| + uint32_t ssrc;
|
| if (Rtcp::IsRtcpPacket(data, length)) {
|
| ssrc = Rtcp::GetSsrcOfSender(data, length);
|
| } else if (!RtpParser::ParseSsrc(data, length, &ssrc)) {
|
| @@ -430,7 +431,7 @@ void CastTransportSenderImpl::OnReceivedLogMessage(
|
| }
|
|
|
| void CastTransportSenderImpl::OnReceivedCastMessage(
|
| - uint32 ssrc,
|
| + uint32_t ssrc,
|
| const RtcpCastMessageCallback& cast_message_cb,
|
| const RtcpCastMessage& cast_message) {
|
| if (!cast_message_cb.is_null())
|
| @@ -438,7 +439,7 @@ void CastTransportSenderImpl::OnReceivedCastMessage(
|
|
|
| DedupInfo dedup_info;
|
| if (audio_sender_ && audio_sender_->ssrc() == ssrc) {
|
| - const int64 acked_bytes =
|
| + const int64_t acked_bytes =
|
| audio_sender_->GetLastByteSentForFrame(cast_message.ack_frame_id);
|
| last_byte_acked_for_audio_ =
|
| std::max(acked_bytes, last_byte_acked_for_audio_);
|
| @@ -465,13 +466,13 @@ void CastTransportSenderImpl::OnReceivedCastMessage(
|
| dedup_info);
|
| }
|
|
|
| -void CastTransportSenderImpl::AddValidSsrc(uint32 ssrc) {
|
| +void CastTransportSenderImpl::AddValidSsrc(uint32_t ssrc) {
|
| valid_ssrcs_.insert(ssrc);
|
| }
|
|
|
| void CastTransportSenderImpl::SendRtcpFromRtpReceiver(
|
| - uint32 ssrc,
|
| - uint32 sender_ssrc,
|
| + uint32_t ssrc,
|
| + uint32_t sender_ssrc,
|
| const RtcpTimeData& time_data,
|
| const RtcpCastMessage* cast_message,
|
| base::TimeDelta target_delay,
|
|
|