mirror of https://github.com/odrling/Aegisub
split out the downmixing audio provider into its own .h/.cpp, in preparation for using it in the ffmpeg audio provider.
Originally committed to SVN as r2257.
This commit is contained in:
Karl Blomster
parent
d120a4758b
commit
6bec0322e4
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@ -0,0 +1,118 @@
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// Copyright (c) 2007-2008, Niels Martin Hansen
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// All rights reserved.
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//
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// Redistribution and use in source and binary forms, with or without
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// modification, are permitted provided that the following conditions are met:
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//
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// * Redistributions of source code must retain the above copyright notice,
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// this list of conditions and the following disclaimer.
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// * Redistributions in binary form must reproduce the above copyright notice,
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// this list of conditions and the following disclaimer in the documentation
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// and/or other materials provided with the distribution.
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// * Neither the name of the Aegisub Group nor the names of its contributors
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// may be used to endorse or promote products derived from this software
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// without specific prior written permission.
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//
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// THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS"
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// AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
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// IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
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// ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
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// LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
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// CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
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// SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
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// INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
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// CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
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// ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
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// POSSIBILITY OF SUCH DAMAGE.
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//
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// -----------------------------------------------------------------------------
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//
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// AEGISUB
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//
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// Website: http://aegisub.cellosoft.com
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// Contact: mailto:jiifurusu@gmail.com
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//////////////////
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// Headers
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#include "audio_provider_downmix.h"
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//////////////////
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// Constructor
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DownmixingAudioProvider::DownmixingAudioProvider(AudioProvider *source) {
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filename = source->GetFilename();
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channels = 1; // target
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src_channels = source->GetChannels();
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num_samples = source->GetNumSamples();
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bytes_per_sample = source->GetBytesPerSample();
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sample_rate = source->GetSampleRate();
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// We now own this
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provider = source;
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if (!(bytes_per_sample == 1 || bytes_per_sample == 2))
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throw _T("Downmixing Audio Provider: Can only downmix 8 and 16 bit audio");
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}
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/////////////////
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// Destructor
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DownmixingAudioProvider::~DownmixingAudioProvider() {
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delete provider;
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}
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////////////////
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// Actual work happens here
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void DownmixingAudioProvider::GetAudio(void *buf, int64_t start, int64_t count) {
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if (count == 0) return;
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// We can do this ourselves
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if (start >= num_samples) {
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if (bytes_per_sample == 1)
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// 8 bit formats are usually unsigned with bias 127
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memset(buf, 127, count);
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else
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// While everything else is signed
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memset(buf, 0, count*bytes_per_sample);
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return;
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}
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// So alloc some temporary memory for this
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// Depending on use, this might be made faster by using
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// a pre-allocced block of memory...?
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char *tmp = new char[count*bytes_per_sample*src_channels];
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provider->GetAudio(tmp, start, count);
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// Now downmix
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// Just average the samples over the channels (really bad if they're out of phase!)
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// XXX: Assuming here that sample data are in machine endian, an upstream provider should ensure that
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if (bytes_per_sample == 1) {
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uint8_t *src = (uint8_t *)tmp;
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uint8_t *dst = (uint8_t *)buf;
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while (count > 0) {
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int sum = 0;
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for (int c = 0; c < src_channels; c++)
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sum += *(src++);
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*(dst++) = (uint8_t)(sum / src_channels);
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count--;
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}
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}
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else if (bytes_per_sample == 2) {
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int16_t *src = (int16_t *)tmp;
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int16_t *dst = (int16_t *)buf;
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while (count > 0) {
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int sum = 0;
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for (int c = 0; c < src_channels; c++)
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sum += *(src++);
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*(dst++) = (int16_t)(sum / src_channels);
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count--;
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}
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}
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// Done downmixing, free the work buffer
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delete[] tmp;
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}
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@ -0,0 +1,49 @@
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// Copyright (c) 2007-2008, Niels Martin Hansen
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// All rights reserved.
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//
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// Redistribution and use in source and binary forms, with or without
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// modification, are permitted provided that the following conditions are met:
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//
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// * Redistributions of source code must retain the above copyright notice,
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// this list of conditions and the following disclaimer.
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// * Redistributions in binary form must reproduce the above copyright notice,
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// this list of conditions and the following disclaimer in the documentation
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// and/or other materials provided with the distribution.
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// * Neither the name of the Aegisub Group nor the names of its contributors
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// may be used to endorse or promote products derived from this software
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// without specific prior written permission.
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//
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// THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS"
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// AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
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// IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
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// ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
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// LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
|
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|
// CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
|
||||||
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// SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
|
||||||
|
// INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
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||||||
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// CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
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// ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
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// POSSIBILITY OF SUCH DAMAGE.
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//
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// -----------------------------------------------------------------------------
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//
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// AEGISUB
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//
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// Website: http://aegisub.cellosoft.com
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// Contact: mailto:jiifurusu@gmail.com
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//
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#include "include/aegisub/audio_provider.h"
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class DownmixingAudioProvider : public AudioProvider {
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private:
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AudioProvider *provider;
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int src_channels;
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public:
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DownmixingAudioProvider(AudioProvider *source);
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~DownmixingAudioProvider();
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void GetAudio(void *buf, int64_t start, int64_t count);
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};
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@ -37,6 +37,7 @@
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#include <wx/filename.h>
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#include <wx/filename.h>
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#include <wx/file.h>
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#include <wx/file.h>
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#include "audio_provider_pcm.h"
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#include "audio_provider_pcm.h"
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#include "audio_provider_downmix.h"
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#include "utils.h"
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#include "utils.h"
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#include "aegisub_endian.h"
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#include "aegisub_endian.h"
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#include <stdint.h>
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#include <stdint.h>
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@ -370,91 +371,6 @@ public:
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};
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};
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// Mix down any number of channels to mono
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class DownmixingAudioProvider : public AudioProvider {
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private:
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AudioProvider *provider;
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int src_channels;
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public:
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DownmixingAudioProvider(AudioProvider *source)
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{
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filename = source->GetFilename();
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channels = 1; // target
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src_channels = source->GetChannels();
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num_samples = source->GetNumSamples();
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bytes_per_sample = source->GetBytesPerSample();
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sample_rate = source->GetSampleRate();
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// We now own this
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provider = source;
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if (!(bytes_per_sample == 1 || bytes_per_sample == 2)) throw _T("Downmixing Audio Provider: Can only downmix 8 and 16 bit audio");
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}
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~DownmixingAudioProvider()
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{
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delete provider;
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}
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void GetAudio(void *buf, int64_t start, int64_t count)
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{
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if (count == 0) return;
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// We can do this ourselves
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if (start >= num_samples) {
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if (bytes_per_sample == 1)
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// 8 bit formats are usually unsigned with bias 127
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memset(buf, 127, count);
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else
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// While everything else is signed
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memset(buf, 0, count*bytes_per_sample);
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return;
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}
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// So alloc some temporary memory for this
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// Depending on use, this might be made faster by using
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// a pre-allocced block of memory...?
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char *tmp = new char[count*bytes_per_sample*src_channels];
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provider->GetAudio(tmp, start, count);
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// Now downmix
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// Just average the samples over the channels (really bad if they're out of phase!)
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// XXX: Assuming here that sample data are in machine endian, an upstream provider should ensure that
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if (bytes_per_sample == 1) {
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uint8_t *src = (uint8_t *)tmp;
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uint8_t *dst = (uint8_t *)buf;
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while (count > 0) {
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int sum = 0;
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for (int c = 0; c < src_channels; c++)
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sum += *(src++);
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*(dst++) = (uint8_t)(sum / src_channels);
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count--;
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}
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}
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else if (bytes_per_sample == 2) {
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int16_t *src = (int16_t *)tmp;
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int16_t *dst = (int16_t *)buf;
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while (count > 0) {
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int sum = 0;
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for (int c = 0; c < src_channels; c++)
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sum += *(src++);
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*(dst++) = (int16_t)(sum / src_channels);
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count--;
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}
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}
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// Done downmixing, free the work buffer
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delete[] tmp;
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}
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};
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AudioProvider *CreatePCMAudioProvider(const wxString &filename)
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AudioProvider *CreatePCMAudioProvider(const wxString &filename)
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{
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{
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AudioProvider *provider = 0;
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AudioProvider *provider = 0;
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