iqbalrifai
/
Aegisub
mirror of https://github.com/odrling/Aegisub
1
0
Fork 0
Code Issues Releases Wiki Activity
Aegisub/aegisub/audio_provider_lavc.cpp

223 lines
7.7 KiB
C++
Raw Blame History

// Copyright (c) 2005-2006, Rodrigo Braz Monteiro, Fredrik Mellbin
// All rights reserved.
//
// Redistribution and use in source and binary forms, with or without
// modification, are permitted provided that the following conditions are met:
//
// * Redistributions of source code must retain the above copyright notice,
// this list of conditions and the following disclaimer.
// * Redistributions in binary form must reproduce the above copyright notice,
// this list of conditions and the following disclaimer in the documentation
// and/or other materials provided with the distribution.
// * Neither the name of the Aegisub Group nor the names of its contributors
// may be used to endorse or promote products derived from this software
// without specific prior written permission.
//
// THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS"
// AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
// IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
// ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
// LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
// CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
// SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
// INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
// CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
// ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
// POSSIBILITY OF SUCH DAMAGE.
//
// -----------------------------------------------------------------------------
//
// AEGISUB
//
// Website: http://aegisub.cellosoft.com
// Contact: mailto:zeratul@cellosoft.com
//
///////////
// Headers
#ifdef WITH_FFMPEG
#ifdef WIN32
#define EMULATE_INTTYPES
#endif
#include <wx/wxprec.h>
/* avcodec.h uses INT64_C in a *single* place. This prolly breaks on Win32,
* but, well. Let's at least fix it for Linux.
*
#define __STDC_CONSTANT_MACROS 1
#include <stdint.h>
* - done in posix/defines.h
*/
extern "C" {
#include <ffmpeg/avcodec.h>
#include <ffmpeg/avformat.h>
}
#include "mkv_wrap.h"
#include "lavc_file.h"
#include "audio_provider_lavc.h"
#include "lavc_file.h"
#include "utils.h"
#include "options.h"
///////////////
// Constructor
LAVCAudioProvider::LAVCAudioProvider(Aegisub::String _filename)
: lavcfile(NULL), codecContext(NULL), rsct(NULL), buffer(NULL)
{
try {
#if 0
/* since seeking currently is likely to be horribly broken with two
* providers accessing the same stream, this is disabled for now.
*/
LAVCVideoProvider *vpro_lavc = dynamic_cast<LAVCVideoProvider *>(vpro);
if (vpro_lavc) {
lavcfile = vpro->lavcfile->AddRef();
filename = vpro_lavc->GetFilename();
} else {
#endif
lavcfile = LAVCFile::Create(_filename);
filename = _filename;
#if 0
}
#endif
audStream = -1;
for (int i = 0; i < lavcfile->fctx->nb_streams; i++) {
codecContext = lavcfile->fctx->streams[i]->codec;
if (codecContext->codec_type == CODEC_TYPE_AUDIO) {
stream = lavcfile->fctx->streams[i];
audStream = i;
break;
}
}
if (audStream == -1) {
codecContext = NULL;
throw _T("Could not find an audio stream");
}
AVCodec *codec = avcodec_find_decoder(codecContext->codec_id);
if (!codec) {
codecContext = NULL;
throw _T("Could not find a suitable audio decoder");
}
if (avcodec_open(codecContext, codec) < 0)
throw _T("Failed to open audio decoder");
sample_rate = Options.AsInt(_T("Audio Sample Rate"));
if (!sample_rate)
sample_rate = codecContext->sample_rate;
channels = 1;
/* FIXME: this entire provider always assumes 16-bit audio. Currently that isn't a problem since
ffmpeg always converts everything to 16-bit, but in the future it might become one. */
bytes_per_sample = 2;
/* aegisub currently supports mono only, so always resample unless it's mono with the desired samplerate */
if ((sample_rate != codecContext->sample_rate) || (codecContext->channels > 1)) {
rsct = audio_resample_init(1, codecContext->channels, sample_rate, codecContext->sample_rate);
if (!rsct)
throw _T("Failed to initialize resampling");
resample_ratio = (float)sample_rate / (float)codecContext->sample_rate;
}
double length = (double)stream->duration * av_q2d(stream->time_base);
num_samples = (int64_t)(length * sample_rate);
buffer = (int16_t *)malloc(AVCODEC_MAX_AUDIO_FRAME_SIZE);
if (!buffer)
throw _T("Out of memory");
} catch (...) {
Destroy();
throw;
}
}
LAVCAudioProvider::~LAVCAudioProvider()
{
Destroy();
}
void LAVCAudioProvider::Destroy()
{
if (buffer)
free(buffer);
if (rsct)
audio_resample_close(rsct);
if (codecContext)
avcodec_close(codecContext);
if (lavcfile)
lavcfile->Release();
}
void LAVCAudioProvider::GetAudio(void *buf, int64_t start, int64_t count)
{
int16_t *_buf = (int16_t *)buf;
int64_t samples_to_decode = num_samples - start; /* samples left to the end of the stream */
if (count < samples_to_decode) /* haven't reached the end yet, so just decode the requested number of samples */
samples_to_decode = count;
if (samples_to_decode < 0) /* requested beyond the end of the stream */
samples_to_decode = 0;
/* if we got asked for more samples than there are left in the stream, add zeros to the decoding buffer until
we have enough to fill the request */
memset(_buf + samples_to_decode, 0, (count - samples_to_decode) * 2);
AVPacket packet;
while (samples_to_decode > 0 && av_read_frame(lavcfile->fctx, &packet) >= 0) {
/* we're not dealing with video packets in this here provider */
if (packet.stream_index == audStream) {
int size = packet.size;
uint8_t *data = packet.data;
while (size > 0) {
int temp_output_buffer_size = AVCODEC_MAX_AUDIO_FRAME_SIZE; /* see constructor, it malloc()'s buffer to this */
int retval, decoded_samples;
retval = avcodec_decode_audio2(codecContext, buffer, &temp_output_buffer_size, data, size);
if (retval <= 0)
throw _T("Failed to decode audio");
if (temp_output_buffer_size <= 0) /* sanity checking, shouldn't ever happen */
throw _T("Audio decoder lied about output size! This can't happen and you didn't see this error message. Move along.");
decoded_samples = temp_output_buffer_size / 2; /* 2 bytes per sample */
size -= retval;
data += retval;
/* do we need to resample? */
if (rsct) {
/* if ((int64_t)(decoded_samples * resample_ratio / codecContext->channels) > samples_to_decode)
decoded_samples = (int64_t)(samples_to_decode / resample_ratio * codecContext->channels); */
/* what is the point of the above? if we ended up with more samples than we wanted,
we should do something about it, not pretend that everything's OK. -Fluff */
decoded_samples = audio_resample(rsct, _buf, buffer, decoded_samples / codecContext->channels);
/* make some noise if we somehow ended up with more samples than we wanted (will cause audio skew) */
if (decoded_samples > samples_to_decode)
wxLogMessage(wxString::Format(_T("Warning: decoder output more samples than requested, audio skew highly likely! (Wanted %d, got %d)"), (int)samples_to_decode, decoded_samples));
} else {
/* no resampling needed, just copy to the buffer */
/* if (decoded_samples > samples_to_decode)
decoded_samples = samples_to_decode; */
/* I do not understand the point of the above -Fluff */
if (decoded_samples > samples_to_decode)
wxLogMessage(wxString::Format(_T("Warning: decoder output more samples than requested, audio skew highly likely! (Wanted %d, got %d)"), (int)samples_to_decode, decoded_samples));
memcpy(_buf, buffer, temp_output_buffer_size);
}
_buf += decoded_samples;
samples_to_decode -= decoded_samples;
}
}
av_free_packet(&packet);
}
}
#endif
Reference in New Issue View Git Blame Copy Permalink