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Reland of Adding a some checks and switching out a few assert for RTC_[D]CHECK. (patchset #1 id:1 of https://codereview.webrtc.org/2018553002/ )

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Adding a some checks and switching out a few assert for RTC_[D]CHECK.
These changes are around use of AudioFrame.data_ to help us catch issues earlier since assert() is left out in release builds, including builds with DCHECK enabled.  I've also added a few full-on CHECKs to avoid reading past buffer boundaries or continuing on in a failed state.

TBR=kjellander@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true

Committed: https://crrev.com/60c4e0ae8f124f08372645a95042f4a1246d7aa3
Cr-Commit-Position: refs/heads/master@{#12925}

Committed: https://crrev.com/5771beb265129082d31736259b7dc6ca037cff4d
Cr-Commit-Position: refs/heads/master@{#12926}

Review URL: https://codereview.webrtc.org/2014973002 .

Cr-Commit-Position: refs/heads/master@{#12927}
This commit is contained in:
Tommi 2016-05-26 22:03:05 +02:00
parent 5771beb265
commit 54e1c6a500
1 changed files with 11 additions and 11 deletions
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22
webrtc/voice_engine/utility.cc
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@ -10,6 +10,7 @@
#include "webrtc/voice_engine/utility.h"
#include "webrtc/base/checks.h"
#include "webrtc/base/logging.h"
#include "webrtc/common_audio/resampler/include/push_resampler.h"
#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
@ -52,21 +53,18 @@ void RemixAndResample(const int16_t* src_data,
if (resampler->InitializeIfNeeded(sample_rate_hz, dst_frame->sample_rate_hz_,
audio_ptr_num_channels) == -1) {
LOG(LS_ERROR) << "InitializeIfNeeded failed: sample_rate_hz = "
<< sample_rate_hz << ", dst_frame->sample_rate_hz_ = "
<< dst_frame->sample_rate_hz_
<< ", audio_ptr_num_channels = " << audio_ptr_num_channels;
assert(false);
FATAL() << "InitializeIfNeeded failed: sample_rate_hz = " << sample_rate_hz
<< ", dst_frame->sample_rate_hz_ = " << dst_frame->sample_rate_hz_
<< ", audio_ptr_num_channels = " << audio_ptr_num_channels;
}
const size_t src_length = samples_per_channel * audio_ptr_num_channels;
int out_length = resampler->Resample(audio_ptr, src_length, dst_frame->data_,
AudioFrame::kMaxDataSizeSamples);
if (out_length == -1) {
LOG(LS_ERROR) << "Resample failed: audio_ptr = " << audio_ptr
<< ", src_length = " << src_length
<< ", dst_frame->data_ = " << dst_frame->data_;
assert(false);
FATAL() << "Resample failed: audio_ptr = " << audio_ptr
<< ", src_length = " << src_length
<< ", dst_frame->data_ = " << dst_frame->data_;
}
dst_frame->samples_per_channel_ = out_length / audio_ptr_num_channels;
@ -84,8 +82,10 @@ void MixWithSat(int16_t target[],
const int16_t source[],
size_t source_channel,
size_t source_len) {
assert(target_channel == 1 || target_channel == 2);
assert(source_channel == 1 || source_channel == 2);
RTC_DCHECK_GE(target_channel, 1u);
RTC_DCHECK_LE(target_channel, 2u);
RTC_DCHECK_GE(source_channel, 1u);
RTC_DCHECK_LE(source_channel, 2u);
if (target_channel == 2 && source_channel == 1) {
// Convert source from mono to stereo.