14b43beb7c
TBR=niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/915006 git-svn-id: http://webrtc.googlecode.com/svn/trunk@2963 4adac7df-926f-26a2-2b94-8c16560cd09d
74 lines
2.5 KiB
C++
74 lines
2.5 KiB
C++
/*
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* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "output_mixer_internal.h"
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#include "audio_frame_operations.h"
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#include "common_audio/resampler/include/resampler.h"
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#include "module_common_types.h"
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#include "trace.h"
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namespace webrtc {
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namespace voe {
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int RemixAndResample(const AudioFrame& src_frame,
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Resampler* resampler,
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AudioFrame* dst_frame) {
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const int16_t* audio_ptr = src_frame.data_;
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int audio_ptr_num_channels = src_frame.num_channels_;
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int16_t mono_audio[AudioFrame::kMaxDataSizeSamples];
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// Downmix before resampling.
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if (src_frame.num_channels_ == 2 && dst_frame->num_channels_ == 1) {
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AudioFrameOperations::StereoToMono(src_frame.data_,
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src_frame.samples_per_channel_,
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mono_audio);
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audio_ptr = mono_audio;
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audio_ptr_num_channels = 1;
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}
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const ResamplerType resampler_type = audio_ptr_num_channels == 1 ?
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kResamplerSynchronous : kResamplerSynchronousStereo;
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if (resampler->ResetIfNeeded(src_frame.sample_rate_hz_,
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dst_frame->sample_rate_hz_,
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resampler_type) == -1) {
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*dst_frame = src_frame;
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WEBRTC_TRACE(kTraceError, kTraceVoice, -1,
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"%s ResetIfNeeded failed", __FUNCTION__);
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return -1;
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}
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int out_length = 0;
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if (resampler->Push(audio_ptr,
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src_frame.samples_per_channel_* audio_ptr_num_channels,
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dst_frame->data_,
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AudioFrame::kMaxDataSizeSamples,
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out_length) == 0) {
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dst_frame->samples_per_channel_ = out_length / audio_ptr_num_channels;
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} else {
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*dst_frame = src_frame;
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WEBRTC_TRACE(kTraceError, kTraceVoice, -1,
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"%s resampling failed", __FUNCTION__);
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return -1;
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}
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// Upmix after resampling.
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if (src_frame.num_channels_ == 1 && dst_frame->num_channels_ == 2) {
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// The audio in dst_frame really is mono at this point; MonoToStereo will
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// set this back to stereo.
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dst_frame->num_channels_ = 1;
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AudioFrameOperations::MonoToStereo(dst_frame);
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}
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return 0;
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}
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} // namespace voe
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} // namespace webrtc
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