| /* |
| * 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 "webrtc/voice_engine/output_mixer_internal.h" |
| |
| #include "webrtc/common_audio/resampler/include/push_resampler.h" |
| #include "webrtc/modules/interface/module_common_types.h" |
| #include "webrtc/modules/utility/interface/audio_frame_operations.h" |
| #include "webrtc/system_wrappers/interface/logging.h" |
| #include "webrtc/system_wrappers/interface/trace.h" |
| |
| namespace webrtc { |
| namespace voe { |
| |
| int RemixAndResample(const AudioFrame& src_frame, |
| PushResampler* resampler, |
| AudioFrame* dst_frame) { |
| const int16_t* audio_ptr = src_frame.data_; |
| int audio_ptr_num_channels = src_frame.num_channels_; |
| int16_t mono_audio[AudioFrame::kMaxDataSizeSamples]; |
| |
| // Downmix before resampling. |
| if (src_frame.num_channels_ == 2 && dst_frame->num_channels_ == 1) { |
| AudioFrameOperations::StereoToMono(src_frame.data_, |
| src_frame.samples_per_channel_, |
| mono_audio); |
| audio_ptr = mono_audio; |
| audio_ptr_num_channels = 1; |
| } |
| |
| if (resampler->InitializeIfNeeded(src_frame.sample_rate_hz_, |
| dst_frame->sample_rate_hz_, |
| audio_ptr_num_channels) == -1) { |
| dst_frame->CopyFrom(src_frame); |
| LOG_FERR3(LS_ERROR, InitializeIfNeeded, src_frame.sample_rate_hz_, |
| dst_frame->sample_rate_hz_, audio_ptr_num_channels); |
| return -1; |
| } |
| |
| const int src_length = src_frame.samples_per_channel_ * |
| audio_ptr_num_channels; |
| int out_length = resampler->Resample(audio_ptr, src_length, dst_frame->data_, |
| AudioFrame::kMaxDataSizeSamples); |
| if (out_length == -1) { |
| dst_frame->CopyFrom(src_frame); |
| LOG_FERR3(LS_ERROR, Resample, src_length, dst_frame->data_, |
| AudioFrame::kMaxDataSizeSamples); |
| return -1; |
| } |
| dst_frame->samples_per_channel_ = out_length / audio_ptr_num_channels; |
| |
| // Upmix after resampling. |
| if (src_frame.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::MonoToStereo(dst_frame); |
| } |
| return 0; |
| } |
| |
| } // namespace voe |
| } // namespace webrtc |