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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

24
webrtc/modules/audio_coding/codecs/g711/include/audio_encoder_pcm.h
View File

@ -39,8 +39,8 @@ class AudioEncoderPcm : public AudioEncoder {
int SampleRateHz() const override;
int NumChannels() const override;
size_t MaxEncodedBytes() const override;
int Num10MsFramesInNextPacket() const override;
int Max10MsFramesInAPacket() const override;
size_t Num10MsFramesInNextPacket() const override;
size_t Max10MsFramesInAPacket() const override;
int GetTargetBitrate() const override;
EncodedInfo EncodeInternal(uint32_t rtp_timestamp,
const int16_t* audio,
@ -50,9 +50,9 @@ class AudioEncoderPcm : public AudioEncoder {
protected:
AudioEncoderPcm(const Config& config, int sample_rate_hz);
virtual int16_t EncodeCall(const int16_t* audio,
size_t input_len,
uint8_t* encoded) = 0;
virtual size_t EncodeCall(const int16_t* audio,
size_t input_len,
uint8_t* encoded) = 0;
virtual int BytesPerSample() const = 0;
@ -60,7 +60,7 @@ class AudioEncoderPcm : public AudioEncoder {
const int sample_rate_hz_;
const int num_channels_;
const int payload_type_;
const int num_10ms_frames_per_packet_;
const size_t num_10ms_frames_per_packet_;
const size_t full_frame_samples_;
std::vector<int16_t> speech_buffer_;
uint32_t first_timestamp_in_buffer_;
@ -76,9 +76,9 @@ class AudioEncoderPcmA final : public AudioEncoderPcm {
: AudioEncoderPcm(config, kSampleRateHz) {}
protected:
int16_t EncodeCall(const int16_t* audio,
size_t input_len,
uint8_t* encoded) override;
size_t EncodeCall(const int16_t* audio,
size_t input_len,
uint8_t* encoded) override;
int BytesPerSample() const override;
@ -96,9 +96,9 @@ class AudioEncoderPcmU final : public AudioEncoderPcm {
: AudioEncoderPcm(config, kSampleRateHz) {}
protected:
int16_t EncodeCall(const int16_t* audio,
size_t input_len,
uint8_t* encoded) override;
size_t EncodeCall(const int16_t* audio,
size_t input_len,
uint8_t* encoded) override;
int BytesPerSample() const override;

28
webrtc/modules/audio_coding/codecs/g711/include/g711_interface.h
View File

@ -38,9 +38,9 @@ extern "C" {
* Always equal to len input parameter.
*/
int16_t WebRtcG711_EncodeA(const int16_t* speechIn,
int16_t len,
uint8_t* encoded);
size_t WebRtcG711_EncodeA(const int16_t* speechIn,
size_t len,
uint8_t* encoded);
/****************************************************************************
* WebRtcG711_EncodeU(...)
@ -59,9 +59,9 @@ int16_t WebRtcG711_EncodeA(const int16_t* speechIn,
* Always equal to len input parameter.
*/
int16_t WebRtcG711_EncodeU(const int16_t* speechIn,
int16_t len,
uint8_t* encoded);
size_t WebRtcG711_EncodeU(const int16_t* speechIn,
size_t len,
uint8_t* encoded);
/****************************************************************************
* WebRtcG711_DecodeA(...)
@ -82,10 +82,10 @@ int16_t WebRtcG711_EncodeU(const int16_t* speechIn,
* -1 - Error
*/
int16_t WebRtcG711_DecodeA(const uint8_t* encoded,
int16_t len,
int16_t* decoded,
int16_t* speechType);
size_t WebRtcG711_DecodeA(const uint8_t* encoded,
size_t len,
int16_t* decoded,
int16_t* speechType);
/****************************************************************************
* WebRtcG711_DecodeU(...)
@ -106,10 +106,10 @@ int16_t WebRtcG711_DecodeA(const uint8_t* encoded,
* -1 - Error
*/
int16_t WebRtcG711_DecodeU(const uint8_t* encoded,
int16_t len,
int16_t* decoded,
int16_t* speechType);
size_t WebRtcG711_DecodeU(const uint8_t* encoded,
size_t len,
int16_t* decoded,
int16_t* speechType);
/**********************************************************************
* WebRtcG711_Version(...)