diff options
author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 17:32:43 +0000 |
---|---|---|
committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 17:32:43 +0000 |
commit | 6bf0a5cb5034a7e684dcc3500e841785237ce2dd (patch) | |
tree | a68f146d7fa01f0134297619fbe7e33db084e0aa /third_party/libwebrtc/audio/remix_resample.cc | |
parent | Initial commit. (diff) | |
download | thunderbird-6bf0a5cb5034a7e684dcc3500e841785237ce2dd.tar.xz thunderbird-6bf0a5cb5034a7e684dcc3500e841785237ce2dd.zip |
Adding upstream version 1:115.7.0.upstream/1%115.7.0upstream
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'third_party/libwebrtc/audio/remix_resample.cc')
-rw-r--r-- | third_party/libwebrtc/audio/remix_resample.cc | 91 |
1 files changed, 91 insertions, 0 deletions
diff --git a/third_party/libwebrtc/audio/remix_resample.cc b/third_party/libwebrtc/audio/remix_resample.cc new file mode 100644 index 0000000000..178af622a1 --- /dev/null +++ b/third_party/libwebrtc/audio/remix_resample.cc @@ -0,0 +1,91 @@ +/* + * 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 "audio/remix_resample.h" + +#include "api/audio/audio_frame.h" +#include "audio/utility/audio_frame_operations.h" +#include "common_audio/resampler/include/push_resampler.h" +#include "rtc_base/checks.h" + +namespace webrtc { +namespace voe { + +void RemixAndResample(const AudioFrame& src_frame, + PushResampler<int16_t>* resampler, + AudioFrame* dst_frame) { + RemixAndResample(src_frame.data(), src_frame.samples_per_channel_, + src_frame.num_channels_, src_frame.sample_rate_hz_, + resampler, dst_frame); + dst_frame->timestamp_ = src_frame.timestamp_; + dst_frame->elapsed_time_ms_ = src_frame.elapsed_time_ms_; + dst_frame->ntp_time_ms_ = src_frame.ntp_time_ms_; + dst_frame->packet_infos_ = src_frame.packet_infos_; +} + +void RemixAndResample(const int16_t* src_data, + size_t samples_per_channel, + size_t num_channels, + int sample_rate_hz, + PushResampler<int16_t>* resampler, + AudioFrame* dst_frame) { + const int16_t* audio_ptr = src_data; + size_t audio_ptr_num_channels = num_channels; + int16_t downmixed_audio[AudioFrame::kMaxDataSizeSamples]; + + // Downmix before resampling. + if (num_channels > dst_frame->num_channels_) { + RTC_DCHECK(num_channels == 2 || num_channels == 4) + << "num_channels: " << num_channels; + RTC_DCHECK(dst_frame->num_channels_ == 1 || dst_frame->num_channels_ == 2) + << "dst_frame->num_channels_: " << dst_frame->num_channels_; + + AudioFrameOperations::DownmixChannels( + src_data, num_channels, samples_per_channel, dst_frame->num_channels_, + downmixed_audio); + audio_ptr = downmixed_audio; + audio_ptr_num_channels = dst_frame->num_channels_; + } + + if (resampler->InitializeIfNeeded(sample_rate_hz, dst_frame->sample_rate_hz_, + audio_ptr_num_channels) == -1) { + RTC_FATAL() << "InitializeIfNeeded failed: sample_rate_hz = " + << sample_rate_hz << ", dst_frame->sample_rate_hz_ = " + << dst_frame->sample_rate_hz_ + << ", audio_ptr_num_channels = " << audio_ptr_num_channels; + } + + // TODO(yujo): for muted input frames, don't resample. Either 1) allow + // resampler to return output length without doing the resample, so we know + // how much to zero here; or 2) make resampler accept a hint that the input is + // zeroed. + const size_t src_length = samples_per_channel * audio_ptr_num_channels; + int out_length = + resampler->Resample(audio_ptr, src_length, dst_frame->mutable_data(), + AudioFrame::kMaxDataSizeSamples); + if (out_length == -1) { + RTC_FATAL() << "Resample failed: audio_ptr = " << audio_ptr + << ", src_length = " << src_length + << ", dst_frame->mutable_data() = " + << dst_frame->mutable_data(); + } + dst_frame->samples_per_channel_ = out_length / audio_ptr_num_channels; + + // Upmix after resampling. + if (num_channels == 1 && dst_frame->num_channels_ == 2) { + // The audio in dst_frame really is mono at this point; MonoToStereo will + // set this back to stereo. + dst_frame->num_channels_ = 1; + AudioFrameOperations::UpmixChannels(2, dst_frame); + } +} + +} // namespace voe +} // namespace webrtc |