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NetEq: Don't interpolate longer than the output size

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This can happen in rare and strange cases.

Also taking the opportunity to replace all asserts with DCHECKs in
that file.

BUG=chromium:659225

Review-Url: https://codereview.webrtc.org/2499013002
Cr-Commit-Position: refs/heads/master@{#15070}
This commit is contained in:
henrik.lundin 2016-11-14 08:18:52 -08:00 committed by Commit bot
parent 87d1a78754
commit 80c06fa574
1 changed files with 18 additions and 12 deletions
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30
webrtc/modules/audio_coding/neteq/normal.cc
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@ -14,6 +14,7 @@
#include <algorithm> // min
#include "webrtc/base/checks.h"
#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
#include "webrtc/modules/audio_coding/codecs/audio_decoder.h"
#include "webrtc/modules/audio_coding/neteq/audio_multi_vector.h"
@ -34,7 +35,7 @@ int Normal::Process(const int16_t* input,
return static_cast<int>(length);
}
assert(output->Empty());
RTC_DCHECK(output->Empty());
// Output should be empty at this point.
if (length % output->Channels() != 0) {
// The length does not match the number of channels.
@ -44,7 +45,7 @@ int Normal::Process(const int16_t* input,
output->PushBackInterleaved(input, length);
const int fs_mult = fs_hz_ / 8000;
assert(fs_mult > 0);
RTC_DCHECK_GT(fs_mult, 0);
// fs_shift = log2(fs_mult), rounded down.
// Note that |fs_shift| is not "exact" for 48 kHz.
// TODO(hlundin): Investigate this further.
@ -115,8 +116,8 @@ int Normal::Process(const int16_t* input,
int increment = 64 / fs_mult;
for (size_t i = 0; i < length_per_channel; i++) {
// Scale with mute factor.
assert(channel_ix < output->Channels());
assert(i < output->Size());
RTC_DCHECK_LT(channel_ix, output->Channels());
RTC_DCHECK_LT(i, output->Size());
int32_t scaled_signal = (*output)[channel_ix][i] *
external_mute_factor_array[channel_ix];
// Shift 14 with proper rounding.
@ -129,14 +130,19 @@ int Normal::Process(const int16_t* input,
// Interpolate the expanded data into the new vector.
// (NB/WB/SWB32/SWB48 8/16/32/48 samples.)
assert(fs_shift < 3); // Will always be 0, 1, or, 2.
RTC_DCHECK_LT(fs_shift, 3); // Will always be 0, 1, or, 2.
increment = 4 >> fs_shift;
int fraction = increment;
for (size_t i = 0; i < static_cast<size_t>(8 * fs_mult); i++) {
// Don't interpolate over more samples than what is in output. When this
// cap strikes, the interpolation will likely sound worse, but this is an
// emergency operation in response to unexpected input.
const size_t interp_len_samples =
std::min(static_cast<size_t>(8 * fs_mult), output->Size());
for (size_t i = 0; i < interp_len_samples; ++i) {
// TODO(hlundin): Add 16 instead of 8 for correct rounding. Keeping 8
// now for legacy bit-exactness.
assert(channel_ix < output->Channels());
assert(i < output->Size());
RTC_DCHECK_LT(channel_ix, output->Channels());
RTC_DCHECK_LT(i, output->Size());
(*output)[channel_ix][i] =
static_cast<int16_t>((fraction * (*output)[channel_ix][i] +
(32 - fraction) * expanded[channel_ix][i] + 8) >> 5);
@ -144,7 +150,7 @@ int Normal::Process(const int16_t* input,
}
}
} else if (last_mode == kModeRfc3389Cng) {
assert(output->Channels() == 1); // Not adapted for multi-channel yet.
RTC_DCHECK_EQ(output->Channels(), 1); // Not adapted for multi-channel yet.
static const size_t kCngLength = 48;
RTC_DCHECK_LE(static_cast<size_t>(8 * fs_mult), kCngLength);
int16_t cng_output[kCngLength];
@ -165,7 +171,7 @@ int Normal::Process(const int16_t* input,
}
// Interpolate the CNG into the new vector.
// (NB/WB/SWB32/SWB48 8/16/32/48 samples.)
assert(fs_shift < 3); // Will always be 0, 1, or, 2.
RTC_DCHECK_LT(fs_shift, 3); // Will always be 0, 1, or, 2.
int16_t increment = 4 >> fs_shift;
int16_t fraction = increment;
for (size_t i = 0; i < static_cast<size_t>(8 * fs_mult); i++) {
@ -185,8 +191,8 @@ int Normal::Process(const int16_t* input,
for (size_t channel_ix = 0; channel_ix < output->Channels();
++channel_ix) {
// Scale with mute factor.
assert(channel_ix < output->Channels());
assert(i < output->Size());
RTC_DCHECK_LT(channel_ix, output->Channels());
RTC_DCHECK_LT(i, output->Size());
int32_t scaled_signal = (*output)[channel_ix][i] *
external_mute_factor_array[channel_ix];
// Shift 14 with proper rounding.