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Unified Diff: webrtc/modules/audio_coding/neteq/merge.cc

Issue 1901633002: Adding 120 ms frame length support in NetEq. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: fixing two errors Created 4 years, 8 months ago
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Index: webrtc/modules/audio_coding/neteq/merge.cc
diff --git a/webrtc/modules/audio_coding/neteq/merge.cc b/webrtc/modules/audio_coding/neteq/merge.cc
index b62df613672d4ac3c213272a17d952c34a1e1252..94db1129cd9262240c6a1d0c9cf3f8c7f4bbbad4 100644
--- a/webrtc/modules/audio_coding/neteq/merge.cc
+++ b/webrtc/modules/audio_coding/neteq/merge.cc
@@ -39,6 +39,8 @@ Merge::Merge(int fs_hz,
assert(num_channels_ > 0);
}
+Merge::~Merge() = default;
+
size_t Merge::Process(int16_t* input, size_t input_length,
int16_t* external_mute_factor_array,
AudioMultiVector* output) {
@@ -91,9 +93,8 @@ size_t Merge::Process(int16_t* input, size_t input_length,
old_length, input_length_per_channel, expand_period);
}
- static const int kTempDataSize = 3600;
- int16_t temp_data[kTempDataSize]; // TODO(hlundin) Remove this.
- int16_t* decoded_output = temp_data + best_correlation_index;
+ temp_data_.resize(input_length_per_channel + best_correlation_index);
+ int16_t* decoded_output = temp_data_.data() + best_correlation_index;
// Mute the new decoded data if needed (and unmute it linearly).
// This is the overlapping part of expanded_signal.
@@ -127,7 +128,7 @@ size_t Merge::Process(int16_t* input, size_t input_length,
int16_t increment =
static_cast<int16_t>(16384 / (interpolation_length + 1)); // In Q14.
int16_t mute_factor = 16384 - increment;
- memmove(temp_data, expanded_channel,
+ memmove(temp_data_.data(), expanded_channel,
sizeof(int16_t) * best_correlation_index);
DspHelper::CrossFade(&expanded_channel[best_correlation_index],
input_channel, interpolation_length,
@@ -140,8 +141,8 @@ size_t Merge::Process(int16_t* input, size_t input_length,
} else {
assert(output->Size() == output_length);
}
- memcpy(&(*output)[channel][0], temp_data,
- sizeof(temp_data[0]) * output_length);
+ memcpy(&(*output)[channel][0], temp_data_.data(),
+ sizeof(temp_data_[0]) * output_length);
}
// Copy back the first part of the data to |sync_buffer_| and remove it from
@@ -208,22 +209,20 @@ int16_t Merge::SignalScaling(const int16_t* input, size_t input_length,
std::min(static_cast<size_t>(64 * fs_mult_), input_length);
const int16_t expanded_max =
WebRtcSpl_MaxAbsValueW16(expanded_signal, mod_input_length);
- const int16_t input_max = WebRtcSpl_MaxAbsValueW16(input, mod_input_length);
-
- // Calculate energy of expanded signal.
- // |log_fs_mult| is log2(fs_mult_), but is not exact for 48000 Hz.
- int log_fs_mult = 30 - WebRtcSpl_NormW32(fs_mult_);
- int expanded_shift = 6 + log_fs_mult
- - WebRtcSpl_NormW32(expanded_max * expanded_max);
- expanded_shift = std::max(expanded_shift, 0);
+ int32_t factor = (expanded_max * expanded_max) /
minyue-webrtc 2016/05/02 10:44:52 first fix: old code does give enough shifts, sinc
hlundin-webrtc 2016/05/02 11:28:15 Acknowledged.
minyue-webrtc 2016/05/02 11:41:12 ok. but I ought to modify my sentence as "old code
+ (std::numeric_limits<int32_t>::max() /
+ static_cast<int32_t>(mod_input_length));
+ const int expanded_shift = factor == 0 ? 0 : 31 - WebRtcSpl_NormW32(factor);
int32_t energy_expanded = WebRtcSpl_DotProductWithScale(expanded_signal,
expanded_signal,
mod_input_length,
expanded_shift);
// Calculate energy of input signal.
- int input_shift = 6 + log_fs_mult - WebRtcSpl_NormW32(input_max * input_max);
- input_shift = std::max(input_shift, 0);
+ const int16_t input_max = WebRtcSpl_MaxAbsValueW16(input, mod_input_length);
+ factor = (input_max * input_max) / (std::numeric_limits<int32_t>::max() /
+ static_cast<int32_t>(mod_input_length));
+ const int input_shift = factor == 0 ? 0 : 31 - WebRtcSpl_NormW32(factor);
int32_t energy_input = WebRtcSpl_DotProductWithScale(input, input,
mod_input_length,
input_shift);

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