Repo created

TMessagesProj/jni/voip/webrtc/audio/remix_resample.cc

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+/*
+ *  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