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Remove potential real-time reallocation in PushResampler

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This CL removes the use of absl::InlineVector in the PushResampler which
causes real-time reallocations for setups with more than 8 channels.

As part of the CL, it also removes one dependency on absl for the
common_audio module.

Bug: webrtc:11197
Change-Id: I0788ee9a0f3d08b91bb18caa65f660fb52368a97
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161729
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30059}
This commit is contained in:
Per Åhgren
2019-12-11 10:57:13 +01:00
committed by Commit Bot
parent 375eff4f04
commit 3fdb3cbc6a
3 changed files with 13 additions and 10 deletions
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1
common_audio/BUILD.gn
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@ -58,7 +58,6 @@ rtc_library("common_audio") {
"../system_wrappers", "../system_wrappers",
"../system_wrappers:cpu_features_api", "../system_wrappers:cpu_features_api",
"third_party/fft4g", "third_party/fft4g",
"//third_party/abseil-cpp/absl/container:inlined_vector",
"//third_party/abseil-cpp/absl/types:optional", "//third_party/abseil-cpp/absl/types:optional",
] ]

5
common_audio/resampler/include/push_resampler.h
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@ -40,6 +40,11 @@ class PushResampler {
int src_sample_rate_hz_; int src_sample_rate_hz_;
int dst_sample_rate_hz_; int dst_sample_rate_hz_;
size_t num_channels_; size_t num_channels_;
// Vector that is needed to provide the proper inputs and outputs to the
// interleave/de-interleave methods used in Resample. This needs to be
// heap-allocated on the state to support an arbitrary number of channels
// without doing run-time heap-allocations in the Resample method.
std::vector<T*> channel_data_array_;
struct ChannelResampler { struct ChannelResampler {
std::unique_ptr<PushSincResampler> resampler; std::unique_ptr<PushSincResampler> resampler;

17
common_audio/resampler/push_resampler.cc
View File

@ -15,7 +15,6 @@
#include <memory> #include <memory>
#include "absl/container/inlined_vector.h"
#include "common_audio/include/audio_util.h" #include "common_audio/include/audio_util.h"
#include "common_audio/resampler/push_sinc_resampler.h" #include "common_audio/resampler/push_sinc_resampler.h"
#include "rtc_base/checks.h" #include "rtc_base/checks.h"
@ -100,6 +99,8 @@ int PushResampler<T>::InitializeIfNeeded(int src_sample_rate_hz,
channel_resampler->destination.resize(dst_size_10ms_mono); channel_resampler->destination.resize(dst_size_10ms_mono);
} }
channel_data_array_.resize(num_channels_);
return 0; return 0;
} }
@ -121,12 +122,11 @@ int PushResampler<T>::Resample(const T* src,
const size_t src_length_mono = src_length / num_channels_; const size_t src_length_mono = src_length / num_channels_;
const size_t dst_capacity_mono = dst_capacity / num_channels_; const size_t dst_capacity_mono = dst_capacity / num_channels_;
absl::InlinedVector<T*, 8> source_pointers; for (size_t ch = 0; ch < num_channels_; ++ch) {
for (auto& resampler : channel_resamplers_) { channel_data_array_[ch] = channel_resamplers_[ch].source.data();
source_pointers.push_back(resampler.source.data());
} }
Deinterleave(src, src_length_mono, num_channels_, source_pointers.data()); Deinterleave(src, src_length_mono, num_channels_, channel_data_array_.data());
size_t dst_length_mono = 0; size_t dst_length_mono = 0;
@ -136,12 +136,11 @@ int PushResampler<T>::Resample(const T* src,
dst_capacity_mono); dst_capacity_mono);
} }
absl::InlinedVector<T*, 8> destination_pointers; for (size_t ch = 0; ch < num_channels_; ++ch) {
for (auto& resampler : channel_resamplers_) { channel_data_array_[ch] = channel_resamplers_[ch].destination.data();
destination_pointers.push_back(resampler.destination.data());
} }
Interleave(destination_pointers.data(), dst_length_mono, num_channels_, dst); Interleave(channel_data_array_.data(), dst_length_mono, num_channels_, dst);
return static_cast<int>(dst_length_mono * num_channels_); return static_cast<int>(dst_length_mono * num_channels_);
} }