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author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 09:22:09 +0000 |
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committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 09:22:09 +0000 |
commit | 43a97878ce14b72f0981164f87f2e35e14151312 (patch) | |
tree | 620249daf56c0258faa40cbdcf9cfba06de2a846 /third_party/libwebrtc/modules/audio_processing/audio_buffer.cc | |
parent | Initial commit. (diff) | |
download | firefox-43a97878ce14b72f0981164f87f2e35e14151312.tar.xz firefox-43a97878ce14b72f0981164f87f2e35e14151312.zip |
Adding upstream version 110.0.1.upstream/110.0.1upstream
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'third_party/libwebrtc/modules/audio_processing/audio_buffer.cc')
-rw-r--r-- | third_party/libwebrtc/modules/audio_processing/audio_buffer.cc | 396 |
1 files changed, 396 insertions, 0 deletions
diff --git a/third_party/libwebrtc/modules/audio_processing/audio_buffer.cc b/third_party/libwebrtc/modules/audio_processing/audio_buffer.cc new file mode 100644 index 0000000000..3dbe1fe072 --- /dev/null +++ b/third_party/libwebrtc/modules/audio_processing/audio_buffer.cc @@ -0,0 +1,396 @@ +/* + * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#include "modules/audio_processing/audio_buffer.h" + +#include <string.h> + +#include <cstdint> + +#include "common_audio/channel_buffer.h" +#include "common_audio/include/audio_util.h" +#include "common_audio/resampler/push_sinc_resampler.h" +#include "modules/audio_processing/splitting_filter.h" +#include "rtc_base/checks.h" + +namespace webrtc { +namespace { + +constexpr size_t kSamplesPer32kHzChannel = 320; +constexpr size_t kSamplesPer48kHzChannel = 480; +constexpr size_t kMaxSamplesPerChannel = AudioBuffer::kMaxSampleRate / 100; + +size_t NumBandsFromFramesPerChannel(size_t num_frames) { + if (num_frames == kSamplesPer32kHzChannel) { + return 2; + } + if (num_frames == kSamplesPer48kHzChannel) { + return 3; + } + return 1; +} + +} // namespace + +AudioBuffer::AudioBuffer(size_t input_rate, + size_t input_num_channels, + size_t buffer_rate, + size_t buffer_num_channels, + size_t output_rate, + size_t output_num_channels) + : input_num_frames_(static_cast<int>(input_rate) / 100), + input_num_channels_(input_num_channels), + buffer_num_frames_(static_cast<int>(buffer_rate) / 100), + buffer_num_channels_(buffer_num_channels), + output_num_frames_(static_cast<int>(output_rate) / 100), + output_num_channels_(0), + num_channels_(buffer_num_channels), + num_bands_(NumBandsFromFramesPerChannel(buffer_num_frames_)), + num_split_frames_(rtc::CheckedDivExact(buffer_num_frames_, num_bands_)), + data_( + new ChannelBuffer<float>(buffer_num_frames_, buffer_num_channels_)) { + RTC_DCHECK_GT(input_num_frames_, 0); + RTC_DCHECK_GT(buffer_num_frames_, 0); + RTC_DCHECK_GT(output_num_frames_, 0); + RTC_DCHECK_GT(input_num_channels_, 0); + RTC_DCHECK_GT(buffer_num_channels_, 0); + RTC_DCHECK_LE(buffer_num_channels_, input_num_channels_); + + const bool input_resampling_needed = input_num_frames_ != buffer_num_frames_; + const bool output_resampling_needed = + output_num_frames_ != buffer_num_frames_; + if (input_resampling_needed) { + for (size_t i = 0; i < buffer_num_channels_; ++i) { + input_resamplers_.push_back(std::unique_ptr<PushSincResampler>( + new PushSincResampler(input_num_frames_, buffer_num_frames_))); + } + } + + if (output_resampling_needed) { + for (size_t i = 0; i < buffer_num_channels_; ++i) { + output_resamplers_.push_back(std::unique_ptr<PushSincResampler>( + new PushSincResampler(buffer_num_frames_, output_num_frames_))); + } + } + + if (num_bands_ > 1) { + split_data_.reset(new ChannelBuffer<float>( + buffer_num_frames_, buffer_num_channels_, num_bands_)); + splitting_filter_.reset(new SplittingFilter( + buffer_num_channels_, num_bands_, buffer_num_frames_)); + } +} + +AudioBuffer::~AudioBuffer() {} + +void AudioBuffer::set_downmixing_to_specific_channel(size_t channel) { + downmix_by_averaging_ = false; + RTC_DCHECK_GT(input_num_channels_, channel); + channel_for_downmixing_ = std::min(channel, input_num_channels_ - 1); +} + +void AudioBuffer::set_downmixing_by_averaging() { + downmix_by_averaging_ = true; +} + +void AudioBuffer::CopyFrom(const float* const* stacked_data, + const StreamConfig& stream_config) { + RTC_DCHECK_EQ(stream_config.num_frames(), input_num_frames_); + RTC_DCHECK_EQ(stream_config.num_channels(), input_num_channels_); + RestoreNumChannels(); + const bool downmix_needed = input_num_channels_ > 1 && num_channels_ == 1; + + const bool resampling_needed = input_num_frames_ != buffer_num_frames_; + + if (downmix_needed) { + RTC_DCHECK_GE(kMaxSamplesPerChannel, input_num_frames_); + + std::array<float, kMaxSamplesPerChannel> downmix; + if (downmix_by_averaging_) { + const float kOneByNumChannels = 1.f / input_num_channels_; + for (size_t i = 0; i < input_num_frames_; ++i) { + float value = stacked_data[0][i]; + for (size_t j = 1; j < input_num_channels_; ++j) { + value += stacked_data[j][i]; + } + downmix[i] = value * kOneByNumChannels; + } + } + const float* downmixed_data = downmix_by_averaging_ + ? downmix.data() + : stacked_data[channel_for_downmixing_]; + + if (resampling_needed) { + input_resamplers_[0]->Resample(downmixed_data, input_num_frames_, + data_->channels()[0], buffer_num_frames_); + } + const float* data_to_convert = + resampling_needed ? data_->channels()[0] : downmixed_data; + FloatToFloatS16(data_to_convert, buffer_num_frames_, data_->channels()[0]); + } else { + if (resampling_needed) { + for (size_t i = 0; i < num_channels_; ++i) { + input_resamplers_[i]->Resample(stacked_data[i], input_num_frames_, + data_->channels()[i], + buffer_num_frames_); + FloatToFloatS16(data_->channels()[i], buffer_num_frames_, + data_->channels()[i]); + } + } else { + for (size_t i = 0; i < num_channels_; ++i) { + FloatToFloatS16(stacked_data[i], buffer_num_frames_, + data_->channels()[i]); + } + } + } +} + +void AudioBuffer::CopyTo(const StreamConfig& stream_config, + float* const* stacked_data) { + RTC_DCHECK_EQ(stream_config.num_frames(), output_num_frames_); + + const bool resampling_needed = output_num_frames_ != buffer_num_frames_; + if (resampling_needed) { + for (size_t i = 0; i < num_channels_; ++i) { + FloatS16ToFloat(data_->channels()[i], buffer_num_frames_, + data_->channels()[i]); + output_resamplers_[i]->Resample(data_->channels()[i], buffer_num_frames_, + stacked_data[i], output_num_frames_); + } + } else { + for (size_t i = 0; i < num_channels_; ++i) { + FloatS16ToFloat(data_->channels()[i], buffer_num_frames_, + stacked_data[i]); + } + } + + for (size_t i = num_channels_; i < stream_config.num_channels(); ++i) { + memcpy(stacked_data[i], stacked_data[0], + output_num_frames_ * sizeof(**stacked_data)); + } +} + +void AudioBuffer::CopyTo(AudioBuffer* buffer) const { + RTC_DCHECK_EQ(buffer->num_frames(), output_num_frames_); + + const bool resampling_needed = output_num_frames_ != buffer_num_frames_; + if (resampling_needed) { + for (size_t i = 0; i < num_channels_; ++i) { + output_resamplers_[i]->Resample(data_->channels()[i], buffer_num_frames_, + buffer->channels()[i], + buffer->num_frames()); + } + } else { + for (size_t i = 0; i < num_channels_; ++i) { + memcpy(buffer->channels()[i], data_->channels()[i], + buffer_num_frames_ * sizeof(**buffer->channels())); + } + } + + for (size_t i = num_channels_; i < buffer->num_channels(); ++i) { + memcpy(buffer->channels()[i], buffer->channels()[0], + output_num_frames_ * sizeof(**buffer->channels())); + } +} + +void AudioBuffer::RestoreNumChannels() { + num_channels_ = buffer_num_channels_; + data_->set_num_channels(buffer_num_channels_); + if (split_data_.get()) { + split_data_->set_num_channels(buffer_num_channels_); + } +} + +void AudioBuffer::set_num_channels(size_t num_channels) { + RTC_DCHECK_GE(buffer_num_channels_, num_channels); + num_channels_ = num_channels; + data_->set_num_channels(num_channels); + if (split_data_.get()) { + split_data_->set_num_channels(num_channels); + } +} + +// The resampler is only for supporting 48kHz to 16kHz in the reverse stream. +void AudioBuffer::CopyFrom(const int16_t* const interleaved_data, + const StreamConfig& stream_config) { + RTC_DCHECK_EQ(stream_config.num_channels(), input_num_channels_); + RTC_DCHECK_EQ(stream_config.num_frames(), input_num_frames_); + RestoreNumChannels(); + + const bool resampling_required = input_num_frames_ != buffer_num_frames_; + + const int16_t* interleaved = interleaved_data; + if (num_channels_ == 1) { + if (input_num_channels_ == 1) { + if (resampling_required) { + std::array<float, kMaxSamplesPerChannel> float_buffer; + S16ToFloatS16(interleaved, input_num_frames_, float_buffer.data()); + input_resamplers_[0]->Resample(float_buffer.data(), input_num_frames_, + data_->channels()[0], + buffer_num_frames_); + } else { + S16ToFloatS16(interleaved, input_num_frames_, data_->channels()[0]); + } + } else { + std::array<float, kMaxSamplesPerChannel> float_buffer; + float* downmixed_data = + resampling_required ? float_buffer.data() : data_->channels()[0]; + if (downmix_by_averaging_) { + for (size_t j = 0, k = 0; j < input_num_frames_; ++j) { + int32_t sum = 0; + for (size_t i = 0; i < input_num_channels_; ++i, ++k) { + sum += interleaved[k]; + } + downmixed_data[j] = sum / static_cast<int16_t>(input_num_channels_); + } + } else { + for (size_t j = 0, k = channel_for_downmixing_; j < input_num_frames_; + ++j, k += input_num_channels_) { + downmixed_data[j] = interleaved[k]; + } + } + + if (resampling_required) { + input_resamplers_[0]->Resample(downmixed_data, input_num_frames_, + data_->channels()[0], + buffer_num_frames_); + } + } + } else { + auto deinterleave_channel = [](size_t channel, size_t num_channels, + size_t samples_per_channel, const int16_t* x, + float* y) { + for (size_t j = 0, k = channel; j < samples_per_channel; + ++j, k += num_channels) { + y[j] = x[k]; + } + }; + + if (resampling_required) { + std::array<float, kMaxSamplesPerChannel> float_buffer; + for (size_t i = 0; i < num_channels_; ++i) { + deinterleave_channel(i, num_channels_, input_num_frames_, interleaved, + float_buffer.data()); + input_resamplers_[i]->Resample(float_buffer.data(), input_num_frames_, + data_->channels()[i], + buffer_num_frames_); + } + } else { + for (size_t i = 0; i < num_channels_; ++i) { + deinterleave_channel(i, num_channels_, input_num_frames_, interleaved, + data_->channels()[i]); + } + } + } +} + +void AudioBuffer::CopyTo(const StreamConfig& stream_config, + int16_t* const interleaved_data) { + const size_t config_num_channels = stream_config.num_channels(); + + RTC_DCHECK(config_num_channels == num_channels_ || num_channels_ == 1); + RTC_DCHECK_EQ(stream_config.num_frames(), output_num_frames_); + + const bool resampling_required = buffer_num_frames_ != output_num_frames_; + + int16_t* interleaved = interleaved_data; + if (num_channels_ == 1) { + std::array<float, kMaxSamplesPerChannel> float_buffer; + + if (resampling_required) { + output_resamplers_[0]->Resample(data_->channels()[0], buffer_num_frames_, + float_buffer.data(), output_num_frames_); + } + const float* deinterleaved = + resampling_required ? float_buffer.data() : data_->channels()[0]; + + if (config_num_channels == 1) { + for (size_t j = 0; j < output_num_frames_; ++j) { + interleaved[j] = FloatS16ToS16(deinterleaved[j]); + } + } else { + for (size_t i = 0, k = 0; i < output_num_frames_; ++i) { + float tmp = FloatS16ToS16(deinterleaved[i]); + for (size_t j = 0; j < config_num_channels; ++j, ++k) { + interleaved[k] = tmp; + } + } + } + } else { + auto interleave_channel = [](size_t channel, size_t num_channels, + size_t samples_per_channel, const float* x, + int16_t* y) { + for (size_t k = 0, j = channel; k < samples_per_channel; + ++k, j += num_channels) { + y[j] = FloatS16ToS16(x[k]); + } + }; + + if (resampling_required) { + for (size_t i = 0; i < num_channels_; ++i) { + std::array<float, kMaxSamplesPerChannel> float_buffer; + output_resamplers_[i]->Resample(data_->channels()[i], + buffer_num_frames_, float_buffer.data(), + output_num_frames_); + interleave_channel(i, config_num_channels, output_num_frames_, + float_buffer.data(), interleaved); + } + } else { + for (size_t i = 0; i < num_channels_; ++i) { + interleave_channel(i, config_num_channels, output_num_frames_, + data_->channels()[i], interleaved); + } + } + + for (size_t i = num_channels_; i < config_num_channels; ++i) { + for (size_t j = 0, k = i, n = num_channels_; j < output_num_frames_; + ++j, k += config_num_channels, n += config_num_channels) { + interleaved[k] = interleaved[n]; + } + } + } +} + +void AudioBuffer::SplitIntoFrequencyBands() { + splitting_filter_->Analysis(data_.get(), split_data_.get()); +} + +void AudioBuffer::MergeFrequencyBands() { + splitting_filter_->Synthesis(split_data_.get(), data_.get()); +} + +void AudioBuffer::ExportSplitChannelData( + size_t channel, + int16_t* const* split_band_data) const { + for (size_t k = 0; k < num_bands(); ++k) { + const float* band_data = split_bands_const(channel)[k]; + + RTC_DCHECK(split_band_data[k]); + RTC_DCHECK(band_data); + for (size_t i = 0; i < num_frames_per_band(); ++i) { + split_band_data[k][i] = FloatS16ToS16(band_data[i]); + } + } +} + +void AudioBuffer::ImportSplitChannelData( + size_t channel, + const int16_t* const* split_band_data) { + for (size_t k = 0; k < num_bands(); ++k) { + float* band_data = split_bands(channel)[k]; + RTC_DCHECK(split_band_data[k]); + RTC_DCHECK(band_data); + for (size_t i = 0; i < num_frames_per_band(); ++i) { + band_data[i] = split_band_data[k][i]; + } + } +} + +} // namespace webrtc |