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authorDaniel Baumann <daniel.baumann@progress-linux.org>2024-04-19 00:47:55 +0000
committerDaniel Baumann <daniel.baumann@progress-linux.org>2024-04-19 00:47:55 +0000
commit26a029d407be480d791972afb5975cf62c9360a6 (patch)
treef435a8308119effd964b339f76abb83a57c29483 /third_party/libwebrtc/modules/audio_coding/acm2/acm_remixing.cc
parentInitial commit. (diff)
downloadfirefox-26a029d407be480d791972afb5975cf62c9360a6.tar.xz
firefox-26a029d407be480d791972afb5975cf62c9360a6.zip
Adding upstream version 124.0.1.upstream/124.0.1
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'third_party/libwebrtc/modules/audio_coding/acm2/acm_remixing.cc')
-rw-r--r--third_party/libwebrtc/modules/audio_coding/acm2/acm_remixing.cc114
1 files changed, 114 insertions, 0 deletions
diff --git a/third_party/libwebrtc/modules/audio_coding/acm2/acm_remixing.cc b/third_party/libwebrtc/modules/audio_coding/acm2/acm_remixing.cc
new file mode 100644
index 0000000000..13709dbbee
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/acm2/acm_remixing.cc
@@ -0,0 +1,114 @@
+/*
+ * Copyright (c) 2019 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_coding/acm2/acm_remixing.h"
+
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+void DownMixFrame(const AudioFrame& input, rtc::ArrayView<int16_t> output) {
+ RTC_DCHECK_EQ(input.num_channels_, 2);
+ RTC_DCHECK_EQ(output.size(), input.samples_per_channel_);
+
+ if (input.muted()) {
+ std::fill(output.begin(), output.begin() + input.samples_per_channel_, 0);
+ } else {
+ const int16_t* const input_data = input.data();
+ for (size_t n = 0; n < input.samples_per_channel_; ++n) {
+ output[n] = rtc::dchecked_cast<int16_t>(
+ (int32_t{input_data[2 * n]} + int32_t{input_data[2 * n + 1]}) >> 1);
+ }
+ }
+}
+
+void ReMixFrame(const AudioFrame& input,
+ size_t num_output_channels,
+ std::vector<int16_t>* output) {
+ const size_t output_size = num_output_channels * input.samples_per_channel_;
+ RTC_DCHECK(!(input.num_channels_ == 0 && num_output_channels > 0 &&
+ input.samples_per_channel_ > 0));
+
+ if (output->size() != output_size) {
+ output->resize(output_size);
+ }
+
+ // For muted frames, fill the frame with zeros.
+ if (input.muted()) {
+ std::fill(output->begin(), output->end(), 0);
+ return;
+ }
+
+ // Ensure that the special case of zero input channels is handled correctly
+ // (zero samples per channel is already handled correctly in the code below).
+ if (input.num_channels_ == 0) {
+ return;
+ }
+
+ const int16_t* const input_data = input.data();
+ size_t out_index = 0;
+
+ // When upmixing is needed and the input is mono copy the left channel
+ // into the left and right channels, and set any remaining channels to zero.
+ if (input.num_channels_ == 1 && input.num_channels_ < num_output_channels) {
+ for (size_t k = 0; k < input.samples_per_channel_; ++k) {
+ (*output)[out_index++] = input_data[k];
+ (*output)[out_index++] = input_data[k];
+ for (size_t j = 2; j < num_output_channels; ++j) {
+ (*output)[out_index++] = 0;
+ }
+ RTC_DCHECK_EQ(out_index, (k + 1) * num_output_channels);
+ }
+ RTC_DCHECK_EQ(out_index, input.samples_per_channel_ * num_output_channels);
+ return;
+ }
+
+ size_t in_index = 0;
+
+ // When upmixing is needed and the output is surround, copy the available
+ // channels directly, and set the remaining channels to zero.
+ if (input.num_channels_ < num_output_channels) {
+ for (size_t k = 0; k < input.samples_per_channel_; ++k) {
+ for (size_t j = 0; j < input.num_channels_; ++j) {
+ (*output)[out_index++] = input_data[in_index++];
+ }
+ for (size_t j = input.num_channels_; j < num_output_channels; ++j) {
+ (*output)[out_index++] = 0;
+ }
+ RTC_DCHECK_EQ(in_index, (k + 1) * input.num_channels_);
+ RTC_DCHECK_EQ(out_index, (k + 1) * num_output_channels);
+ }
+ RTC_DCHECK_EQ(in_index, input.samples_per_channel_ * input.num_channels_);
+ RTC_DCHECK_EQ(out_index, input.samples_per_channel_ * num_output_channels);
+
+ return;
+ }
+
+ // When downmixing is needed, and the input is stereo, average the channels.
+ if (input.num_channels_ == 2) {
+ for (size_t n = 0; n < input.samples_per_channel_; ++n) {
+ (*output)[n] = rtc::dchecked_cast<int16_t>(
+ (int32_t{input_data[2 * n]} + int32_t{input_data[2 * n + 1]}) >> 1);
+ }
+ return;
+ }
+
+ // When downmixing is needed, and the input is multichannel, drop the surplus
+ // channels.
+ const size_t num_channels_to_drop = input.num_channels_ - num_output_channels;
+ for (size_t k = 0; k < input.samples_per_channel_; ++k) {
+ for (size_t j = 0; j < num_output_channels; ++j) {
+ (*output)[out_index++] = input_data[in_index++];
+ }
+ in_index += num_channels_to_drop;
+ }
+}
+
+} // namespace webrtc