AnotherFoxGuy/webrtc-audio-processing
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Bump to WebRTC M120 release

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Some API deprecation -- ExperimentalAgc and ExperimentalNs are gone.
We're continuing to carry iSAC even though it's gone upstream, but maybe
we'll want to drop that soon.
This commit is contained in:
Arun Raghavan
2023-12-12 10:42:58 -05:00
parent 9a202fb8c2
commit c6abf6cd3f
479 changed files with 20900 additions and 11996 deletions
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20
webrtc/modules/audio_processing/aec3/block_buffer.cc
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@ -14,25 +14,9 @@
namespace webrtc {
BlockBuffer::BlockBuffer(size_t size,
size_t num_bands,
size_t num_channels,
size_t frame_length)
BlockBuffer::BlockBuffer(size_t size, size_t num_bands, size_t num_channels)
: size(static_cast<int>(size)),
buffer(size,
std::vector<std::vector<std::vector<float>>>(
num_bands,
std::vector<std::vector<float>>(
num_channels,
std::vector<float>(frame_length, 0.f)))) {
for (auto& block : buffer) {
for (auto& band : block) {
for (auto& channel : band) {
std::fill(channel.begin(), channel.end(), 0.f);
}
}
}
}
buffer(size, Block(num_bands, num_channels)) {}
BlockBuffer::~BlockBuffer() = default;