50b2efef6e
The old resampler is used whenever it supports the requested rates. Otherwise the sinc resampler is enabled. Integrated with output_mixer in order to test the change through output_mixer_unittest. The sinc resampler will not yet be used, since we don't feed VoE with any rates that trigger it. BUG=webrtc:1395 R=bjornv@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1355004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3915 4adac7df-926f-26a2-2b94-8c16560cd09d
71 lines
2.7 KiB
C++
71 lines
2.7 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 "webrtc/voice_engine/output_mixer_internal.h"
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#include "webrtc/common_audio/resampler/include/push_resampler.h"
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#include "webrtc/modules/interface/module_common_types.h"
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#include "webrtc/modules/utility/interface/audio_frame_operations.h"
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#include "webrtc/system_wrappers/interface/logging.h"
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#include "webrtc/system_wrappers/interface/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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PushResampler* 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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if (resampler->InitializeIfNeeded(src_frame.sample_rate_hz_,
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dst_frame->sample_rate_hz_,
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audio_ptr_num_channels) == -1) {
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dst_frame->CopyFrom(src_frame);
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LOG_FERR3(LS_ERROR, InitializeIfNeeded, src_frame.sample_rate_hz_,
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dst_frame->sample_rate_hz_, audio_ptr_num_channels);
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return -1;
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}
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const int src_length = src_frame.samples_per_channel_ *
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audio_ptr_num_channels;
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int out_length = resampler->Resample(audio_ptr, src_length, dst_frame->data_,
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AudioFrame::kMaxDataSizeSamples);
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if (out_length == -1) {
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dst_frame->CopyFrom(src_frame);
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LOG_FERR3(LS_ERROR, Resample, src_length, dst_frame->data_,
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AudioFrame::kMaxDataSizeSamples);
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return -1;
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}
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dst_frame->samples_per_channel_ = out_length / audio_ptr_num_channels;
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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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