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}
diff --git a/talk/media/base/audiorenderer.h b/talk/media/base/audiorenderer.h
index 1553318..5c03576 100644
--- a/talk/media/base/audiorenderer.h
+++ b/talk/media/base/audiorenderer.h
@@ -28,6 +28,8 @@
#ifndef TALK_MEDIA_BASE_AUDIORENDERER_H_
#define TALK_MEDIA_BASE_AUDIORENDERER_H_
+#include <cstddef>
+
namespace cricket {
// Abstract interface for rendering the audio data.
@@ -40,7 +42,7 @@
int bits_per_sample,
int sample_rate,
int number_of_channels,
- int number_of_frames) = 0;
+ size_t number_of_frames) = 0;
// Called when the AudioRenderer is going away.
virtual void OnClose() = 0;
diff --git a/talk/media/base/fakemediaengine.h b/talk/media/base/fakemediaengine.h
index 086f831..2c579e2 100644
--- a/talk/media/base/fakemediaengine.h
+++ b/talk/media/base/fakemediaengine.h
@@ -438,7 +438,7 @@
int bits_per_sample,
int sample_rate,
int number_of_channels,
- int number_of_frames) override {}
+ size_t number_of_frames) override {}
void OnClose() override { renderer_ = NULL; }
AudioRenderer* renderer() const { return renderer_; }
diff --git a/talk/media/webrtc/fakewebrtcvoiceengine.h b/talk/media/webrtc/fakewebrtcvoiceengine.h
index 3ac2f3b..cea9e40 100644
--- a/talk/media/webrtc/fakewebrtcvoiceengine.h
+++ b/talk/media/webrtc/fakewebrtcvoiceengine.h
@@ -132,7 +132,7 @@
WEBRTC_STUB(ProcessStream, (webrtc::AudioFrame* frame));
WEBRTC_STUB(ProcessStream, (
const float* const* src,
- int samples_per_channel,
+ size_t samples_per_channel,
int input_sample_rate_hz,
webrtc::AudioProcessing::ChannelLayout input_layout,
int output_sample_rate_hz,
@@ -147,7 +147,7 @@
WEBRTC_STUB(ProcessReverseStream, (webrtc::AudioFrame * frame));
WEBRTC_STUB(AnalyzeReverseStream, (
const float* const* data,
- int samples_per_channel,
+ size_t samples_per_channel,
int sample_rate_hz,
webrtc::AudioProcessing::ChannelLayout layout));
WEBRTC_STUB(ProcessReverseStream,
diff --git a/talk/media/webrtc/webrtcvoiceengine.cc b/talk/media/webrtc/webrtcvoiceengine.cc
index e288e70..80208ba 100644
--- a/talk/media/webrtc/webrtcvoiceengine.cc
+++ b/talk/media/webrtc/webrtcvoiceengine.cc
@@ -1569,7 +1569,7 @@
void WebRtcVoiceEngine::Process(int channel,
webrtc::ProcessingTypes type,
int16_t audio10ms[],
- int length,
+ size_t length,
int sampling_freq,
bool is_stereo) {
rtc::CritScope cs(&signal_media_critical_);
@@ -1665,7 +1665,7 @@
int bits_per_sample,
int sample_rate,
int number_of_channels,
- int number_of_frames) override {
+ size_t number_of_frames) override {
voe_audio_transport_->OnData(channel_,
audio_data,
bits_per_sample,
diff --git a/talk/media/webrtc/webrtcvoiceengine.h b/talk/media/webrtc/webrtcvoiceengine.h
index c8e7980..21056cd 100644
--- a/talk/media/webrtc/webrtcvoiceengine.h
+++ b/talk/media/webrtc/webrtcvoiceengine.h
@@ -130,7 +130,7 @@
void Process(int channel,
webrtc::ProcessingTypes type,
int16_t audio10ms[],
- int length,
+ size_t length,
int sampling_freq,
bool is_stereo) override;
