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

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

View File

@ -106,7 +106,7 @@ class NetEqImpl : public webrtc::NetEq {
// Returns kOK on success, or kFail in case of an error.
int GetAudio(size_t max_length,
int16_t* output_audio,
int* samples_per_channel,
size_t* samples_per_channel,
int* num_channels,
NetEqOutputType* type) override;
@ -203,9 +203,9 @@ class NetEqImpl : public webrtc::NetEq {
protected:
static const int kOutputSizeMs = 10;
static const int kMaxFrameSize = 2880; // 60 ms @ 48 kHz.
static const size_t kMaxFrameSize = 2880; // 60 ms @ 48 kHz.
// TODO(hlundin): Provide a better value for kSyncBufferSize.
static const int kSyncBufferSize = 2 * kMaxFrameSize;
static const size_t kSyncBufferSize = 2 * kMaxFrameSize;
// Inserts a new packet into NetEq. This is used by the InsertPacket method
// above. Returns 0 on success, otherwise an error code.
@ -225,7 +225,7 @@ class NetEqImpl : public webrtc::NetEq {
// Returns 0 on success, otherwise an error code.
int GetAudioInternal(size_t max_length,
int16_t* output,
int* samples_per_channel,
size_t* samples_per_channel,
int* num_channels) EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
// Provides a decision to the GetAudioInternal method. The decision what to
@ -318,7 +318,7 @@ class NetEqImpl : public webrtc::NetEq {
// |required_samples| samples. The packets are inserted into |packet_list|.
// Returns the number of samples that the packets in the list will produce, or
// -1 in case of an error.
int ExtractPackets(int required_samples, PacketList* packet_list)
int ExtractPackets(size_t required_samples, PacketList* packet_list)
EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
// Resets various variables and objects to new values based on the sample rate
@ -375,8 +375,8 @@ class NetEqImpl : public webrtc::NetEq {
StatisticsCalculator stats_ GUARDED_BY(crit_sect_);
int fs_hz_ GUARDED_BY(crit_sect_);
int fs_mult_ GUARDED_BY(crit_sect_);
int output_size_samples_ GUARDED_BY(crit_sect_);
int decoder_frame_length_ GUARDED_BY(crit_sect_);
size_t output_size_samples_ GUARDED_BY(crit_sect_);
size_t decoder_frame_length_ GUARDED_BY(crit_sect_);
Modes last_mode_ GUARDED_BY(crit_sect_);
rtc::scoped_ptr<int16_t[]> mute_factor_array_ GUARDED_BY(crit_sect_);
size_t decoded_buffer_length_ GUARDED_BY(crit_sect_);