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Update a ton of audio code to use size_t more correctly and in general reduce

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use of int16_t/uint16_t.

This is the upshot of a recommendation by henrik.lundin and kwiberg on an original small change ( https://webrtc-codereview.appspot.com/42569004/#ps1 ) to stop using int16_t just because values could fit in it, and is similar in nature to a previous "mass change to use size_t more" ( https://webrtc-codereview.appspot.com/23129004/ ) which also needed to be split up for review but to land all at once, since, like adding "const", such changes tend to cause a lot of transitive effects.

This was be reviewed and approved in pieces:
https://codereview.webrtc.org/1224093003
https://codereview.webrtc.org/1224123002
https://codereview.webrtc.org/1224163002
https://codereview.webrtc.org/1225133003
https://codereview.webrtc.org/1225173002
https://codereview.webrtc.org/1227163003
https://codereview.webrtc.org/1227203003
https://codereview.webrtc.org/1227213002
https://codereview.webrtc.org/1227893002
https://codereview.webrtc.org/1228793004
https://codereview.webrtc.org/1228803003
https://codereview.webrtc.org/1228823002
https://codereview.webrtc.org/1228823003
https://codereview.webrtc.org/1228843002
https://codereview.webrtc.org/1230693002
https://codereview.webrtc.org/1231713002

The change is being landed as TBR to all the folks who reviewed the above.

BUG=chromium:81439
TEST=none
R=andrew@webrtc.org, pbos@webrtc.org
TBR=aluebs, andrew, asapersson, henrika, hlundin, jan.skoglund, kwiberg, minyue, pbos, pthatcher

Review URL: https://codereview.webrtc.org/1230503003 .

Cr-Commit-Position: refs/heads/master@{#9768}
This commit is contained in:
Peter Kasting
2015-08-24 14:52:23 -07:00
parent b594041ec8
commit dce40cf804
471 changed files with 3716 additions and 3499 deletions
Download Patch File Download Diff File Expand all files Collapse all files

16
webrtc/modules/audio_coding/neteq/packet_buffer.cc
View File

@ -181,7 +181,7 @@ const RTPHeader* PacketBuffer::NextRtpHeader() const {
return const_cast<const RTPHeader*>(&(buffer_.front()->header));
}
Packet* PacketBuffer::GetNextPacket(int* discard_count) {
Packet* PacketBuffer::GetNextPacket(size_t* discard_count) {
if (Empty()) {
// Buffer is empty.
return NULL;
@ -194,7 +194,7 @@ Packet* PacketBuffer::GetNextPacket(int* discard_count) {
// Discard other packets with the same timestamp. These are duplicates or
// redundant payloads that should not be used.
int discards = 0;
size_t discards = 0;
while (!Empty() &&
buffer_.front()->header.timestamp == packet->header.timestamp) {
@ -240,15 +240,15 @@ int PacketBuffer::DiscardAllOldPackets(uint32_t timestamp_limit) {
return DiscardOldPackets(timestamp_limit, 0);
}
int PacketBuffer::NumPacketsInBuffer() const {
return static_cast<int>(buffer_.size());
size_t PacketBuffer::NumPacketsInBuffer() const {
return buffer_.size();
}
int PacketBuffer::NumSamplesInBuffer(DecoderDatabase* decoder_database,
int last_decoded_length) const {
size_t PacketBuffer::NumSamplesInBuffer(DecoderDatabase* decoder_database,
size_t last_decoded_length) const {
PacketList::const_iterator it;
int num_samples = 0;
int last_duration = last_decoded_length;
size_t num_samples = 0;
size_t last_duration = last_decoded_length;
for (it = buffer_.begin(); it != buffer_.end(); ++it) {
Packet* packet = (*it);
AudioDecoder* decoder =