forked from mia/Aegisub
222 lines
7.7 KiB
C++
222 lines
7.7 KiB
C++
// Copyright (c) 2005-2006, Rodrigo Braz Monteiro, Fredrik Mellbin
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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:zeratul@cellosoft.com
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//
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///////////
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// Headers
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#ifdef WITH_FFMPEG
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#ifdef WIN32
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#define EMULATE_INTTYPES
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#endif
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#include <wx/wxprec.h>
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/* avcodec.h uses INT64_C in a *single* place. This prolly breaks on Win32,
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* but, well. Let's at least fix it for Linux.
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*
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#define __STDC_CONSTANT_MACROS 1
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#include <stdint.h>
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* - done in posix/defines.h
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*/
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extern "C" {
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#include <ffmpeg/avcodec.h>
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#include <ffmpeg/avformat.h>
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}
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#include "mkv_wrap.h"
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#include "lavc_file.h"
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#include "audio_provider_lavc.h"
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#include "lavc_file.h"
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#include "utils.h"
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#include "options.h"
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///////////////
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// Constructor
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LAVCAudioProvider::LAVCAudioProvider(Aegisub::String _filename)
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: lavcfile(NULL), codecContext(NULL), rsct(NULL), buffer(NULL)
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{
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try {
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#if 0
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/* since seeking currently is likely to be horribly broken with two
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* providers accessing the same stream, this is disabled for now.
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*/
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LAVCVideoProvider *vpro_lavc = dynamic_cast<LAVCVideoProvider *>(vpro);
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if (vpro_lavc) {
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lavcfile = vpro->lavcfile->AddRef();
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filename = vpro_lavc->GetFilename();
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} else {
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#endif
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lavcfile = LAVCFile::Create(_filename);
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filename = _filename;
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#if 0
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}
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#endif
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audStream = -1;
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for (int i = 0; i < lavcfile->fctx->nb_streams; i++) {
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codecContext = lavcfile->fctx->streams[i]->codec;
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if (codecContext->codec_type == CODEC_TYPE_AUDIO) {
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stream = lavcfile->fctx->streams[i];
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audStream = i;
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break;
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}
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}
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if (audStream == -1) {
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codecContext = NULL;
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throw _T("Could not find an audio stream");
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}
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AVCodec *codec = avcodec_find_decoder(codecContext->codec_id);
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if (!codec) {
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codecContext = NULL;
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throw _T("Could not find a suitable audio decoder");
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}
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if (avcodec_open(codecContext, codec) < 0)
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throw _T("Failed to open audio decoder");
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sample_rate = Options.AsInt(_T("Audio Sample Rate"));
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if (!sample_rate)
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sample_rate = codecContext->sample_rate;
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channels = 1;
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/* FIXME: this entire provider always assumes 16-bit audio. Currently that isn't a problem since
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ffmpeg always converts everything to 16-bit, but in the future it might become one. */
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bytes_per_sample = 2;
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/* aegisub currently supports mono only, so always resample unless it's mono with the desired samplerate */
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if ((sample_rate != codecContext->sample_rate) || (codecContext->channels > 1)) {
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rsct = audio_resample_init(1, codecContext->channels, sample_rate, codecContext->sample_rate);
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if (!rsct)
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throw _T("Failed to initialize resampling");
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resample_ratio = (float)sample_rate / (float)codecContext->sample_rate;
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}
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double length = (double)stream->duration * av_q2d(stream->time_base);
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num_samples = (int64_t)(length * sample_rate);
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buffer = (int16_t *)malloc(AVCODEC_MAX_AUDIO_FRAME_SIZE);
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if (!buffer)
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throw _T("Out of memory");
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} catch (...) {
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Destroy();
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throw;
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}
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}
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LAVCAudioProvider::~LAVCAudioProvider()
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{
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Destroy();
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}
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void LAVCAudioProvider::Destroy()
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{
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if (buffer)
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free(buffer);
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if (rsct)
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audio_resample_close(rsct);
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if (codecContext)
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avcodec_close(codecContext);
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if (lavcfile)
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lavcfile->Release();
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}
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void LAVCAudioProvider::GetAudio(void *buf, int64_t start, int64_t count)
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{
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int16_t *_buf = (int16_t *)buf;
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int64_t samples_to_decode = num_samples - start; /* samples left to the end of the stream */
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if (count < samples_to_decode) /* haven't reached the end yet, so just decode the requested number of samples */
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samples_to_decode = count;
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if (samples_to_decode < 0) /* requested beyond the end of the stream */
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samples_to_decode = 0;
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/* if we got asked for more samples than there are left in the stream, add zeros to the decoding buffer until
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we have enough to fill the request */
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memset(_buf + samples_to_decode, 0, (count - samples_to_decode) * 2);
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AVPacket packet;
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while (samples_to_decode > 0 && av_read_frame(lavcfile->fctx, &packet) >= 0) {
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/* we're not dealing with video packets in this here provider */
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if (packet.stream_index == audStream) {
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int size = packet.size;
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uint8_t *data = packet.data;
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while (size > 0) {
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int temp_output_buffer_size = AVCODEC_MAX_AUDIO_FRAME_SIZE; /* see constructor, it malloc()'s buffer to this */
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int retval, decoded_samples;
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retval = avcodec_decode_audio2(codecContext, buffer, &temp_output_buffer_size, data, size);
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if (retval <= 0)
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throw _T("Failed to decode audio");
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if (temp_output_buffer_size <= 0) /* sanity checking, shouldn't ever happen */
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throw _T("Audio decoder lied about output size! This can't happen and you didn't see this error message. Move along.");
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decoded_samples = temp_output_buffer_size / 2; /* 2 bytes per sample */
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size -= retval;
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data += retval;
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/* do we need to resample? */
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if (rsct) {
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/* if ((int64_t)(decoded_samples * resample_ratio / codecContext->channels) > samples_to_decode)
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decoded_samples = (int64_t)(samples_to_decode / resample_ratio * codecContext->channels); */
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/* what is the point of the above? if we ended up with more samples than we wanted,
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we should do something about it, not pretend that everything's OK. -Fluff */
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decoded_samples = audio_resample(rsct, _buf, buffer, decoded_samples / codecContext->channels);
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/* make some noise if we somehow ended up with more samples than we wanted (will cause audio skew) */
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if (decoded_samples > samples_to_decode)
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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));
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} else {
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/* no resampling needed, just copy to the buffer */
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/* if (decoded_samples > samples_to_decode)
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decoded_samples = samples_to_decode; */
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/* I do not understand the point of the above -Fluff */
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if (decoded_samples > samples_to_decode)
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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));
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memcpy(_buf, buffer, temp_output_buffer_size);
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}
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_buf += decoded_samples;
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samples_to_decode -= decoded_samples;
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}
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}
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av_free_packet(&packet);
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}
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}
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#endif
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