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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
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18
webrtc/modules/audio_device/audio_device_buffer.h
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@ -19,7 +19,7 @@ namespace webrtc {
class CriticalSectionWrapper;
const uint32_t kPulsePeriodMs = 1000;
const uint32_t kMaxBufferSizeBytes = 3840; // 10ms in stereo @ 96kHz
const size_t kMaxBufferSizeBytes = 3840; // 10ms in stereo @ 96kHz
class AudioDeviceObserver;
@ -50,7 +50,7 @@ public:
AudioDeviceModule::ChannelType& channel) const;
virtual int32_t SetRecordedBuffer(const void* audioBuffer,
uint32_t nSamples);
size_t nSamples);
int32_t SetCurrentMicLevel(uint32_t level);
virtual void SetVQEData(int playDelayMS,
int recDelayMS,
@ -58,7 +58,7 @@ public:
virtual int32_t DeliverRecordedData();
uint32_t NewMicLevel() const;
virtual int32_t RequestPlayoutData(uint32_t nSamples);
virtual int32_t RequestPlayoutData(size_t nSamples);
virtual int32_t GetPlayoutData(void* audioBuffer);
int32_t StartInputFileRecording(
@ -87,22 +87,22 @@ private:
AudioDeviceModule::ChannelType _recChannel;
// 2 or 4 depending on mono or stereo
uint8_t _recBytesPerSample;
uint8_t _playBytesPerSample;
size_t _recBytesPerSample;
size_t _playBytesPerSample;
// 10ms in stereo @ 96kHz
int8_t _recBuffer[kMaxBufferSizeBytes];
// one sample <=> 2 or 4 bytes
uint32_t _recSamples;
uint32_t _recSize; // in bytes
size_t _recSamples;
size_t _recSize; // in bytes
// 10ms in stereo @ 96kHz
int8_t _playBuffer[kMaxBufferSizeBytes];
// one sample <=> 2 or 4 bytes
uint32_t _playSamples;
uint32_t _playSize; // in bytes
size_t _playSamples;
size_t _playSize; // in bytes
FileWrapper& _recFile;
FileWrapper& _playFile;