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/webrtc/modules/audio_coding/main/acm2/acm_receiver.cc b/webrtc/modules/audio_coding/main/acm2/acm_receiver.cc
index 4c11197..1cefeb6 100644
--- a/webrtc/modules/audio_coding/main/acm2/acm_receiver.cc
+++ b/webrtc/modules/audio_coding/main/acm2/acm_receiver.cc
@@ -344,7 +344,7 @@
int AcmReceiver::GetAudio(int desired_freq_hz, AudioFrame* audio_frame) {
enum NetEqOutputType type;
- int samples_per_channel;
+ size_t samples_per_channel;
int num_channels;
bool return_silence = false;
@@ -394,7 +394,7 @@
}
// NetEq always returns 10 ms of audio.
- current_sample_rate_hz_ = samples_per_channel * 100;
+ current_sample_rate_hz_ = static_cast<int>(samples_per_channel * 100);
// Update if resampling is required.
bool need_resampling = (desired_freq_hz != -1) &&
@@ -403,18 +403,19 @@
if (need_resampling && !resampled_last_output_frame_) {
// Prime the resampler with the last frame.
int16_t temp_output[AudioFrame::kMaxDataSizeSamples];
- samples_per_channel =
+ int samples_per_channel_int =
resampler_.Resample10Msec(last_audio_buffer_.get(),
current_sample_rate_hz_,
desired_freq_hz,
num_channels,
AudioFrame::kMaxDataSizeSamples,
temp_output);
- if (samples_per_channel < 0) {
+ if (samples_per_channel_int < 0) {
LOG(LERROR) << "AcmReceiver::GetAudio - "
"Resampling last_audio_buffer_ failed.";
return -1;
}
+ samples_per_channel = static_cast<size_t>(samples_per_channel_int);
}
// The audio in |audio_buffer_| is tansferred to |audio_frame_| below, either
@@ -422,17 +423,18 @@
// TODO(henrik.lundin) Glitches in the output may appear if the output rate
// from NetEq changes. See WebRTC issue 3923.
if (need_resampling) {
- samples_per_channel =
+ int samples_per_channel_int =
resampler_.Resample10Msec(audio_buffer_.get(),
current_sample_rate_hz_,
desired_freq_hz,
num_channels,
AudioFrame::kMaxDataSizeSamples,
audio_frame->data_);
- if (samples_per_channel < 0) {
+ if (samples_per_channel_int < 0) {
LOG(LERROR) << "AcmReceiver::GetAudio - Resampling audio_buffer_ failed.";
return -1;
}
+ samples_per_channel = static_cast<size_t>(samples_per_channel_int);
resampled_last_output_frame_ = true;
} else {
resampled_last_output_frame_ = false;
@@ -448,7 +450,7 @@
audio_frame->num_channels_ = num_channels;
audio_frame->samples_per_channel_ = samples_per_channel;
- audio_frame->sample_rate_hz_ = samples_per_channel * 100;
+ audio_frame->sample_rate_hz_ = static_cast<int>(samples_per_channel * 100);
// Should set |vad_activity| before calling SetAudioFrameActivityAndType().
audio_frame->vad_activity_ = previous_audio_activity_;
@@ -787,10 +789,11 @@
frame->sample_rate_hz_ = current_sample_rate_hz_;
}
- frame->samples_per_channel_ = frame->sample_rate_hz_ / 100; // Always 10 ms.
+ frame->samples_per_channel_ =
+ static_cast<size_t>(frame->sample_rate_hz_ / 100); // Always 10 ms.
frame->speech_type_ = AudioFrame::kCNG;
frame->vad_activity_ = AudioFrame::kVadPassive;
- int samples = frame->samples_per_channel_ * frame->num_channels_;
+ size_t samples = frame->samples_per_channel_ * frame->num_channels_;
memset(frame->data_, 0, samples * sizeof(int16_t));
return true;
}
