Aegisub/aegisub/audio_provider_lavc.cpp
Karl Blomster bf931df635 fix retarded breakage in previous commit.
Originally committed to SVN as r2304.
2008-08-14 23:49:11 +00:00

297 lines
9.9 KiB
C++

// 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.
*/
/* Update: this used to be commented out but is now needed on Windows.
* Not sure about Linux, so it's wrapped in an ifdef.
*/
#ifdef WIN32
#define __STDC_CONSTANT_MACROS 1
#include <stdint.h>
#endif /* WIN32 */
/* - 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.c_str();
#if 0
}
#endif
audStream = -1;
for (int i = 0; i < (int)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("ffmpeg audio provider: Could not find an audio stream");
}
AVCodec *codec = avcodec_find_decoder(codecContext->codec_id);
if (!codec) {
codecContext = NULL;
throw _T("ffmpeg audio provider: Could not find a suitable audio decoder");
}
if (avcodec_open(codecContext, codec) < 0)
throw _T("ffmpeg audio provider: Failed to open audio decoder");
sample_rate = Options.AsInt(_T("Audio Sample Rate"));
if (!sample_rate) {
/* aegisub wants audio with sample rate higher than 32khz */
if (codecContext->sample_rate < 32000)
sample_rate = 48000;
else
sample_rate = codecContext->sample_rate;
}
/* we rely on the intermediate audio provider to do downmixing for us later if necessary */
channels = codecContext->channels;
/* FIXME: we need support for more audio types than just 16-bit int */
switch (codecContext->sample_fmt) {
case SAMPLE_FMT_S16: bytes_per_sample = 2; break;
default:
throw _T("ffmpeg audio provider: Only 16-bit audio is supported");
}
/* initiate resampling if necessary */
if (sample_rate != codecContext->sample_rate) {
rsct = audio_resample_init(channels, channels, sample_rate, codecContext->sample_rate);
if (!rsct)
throw _T("ffmpeg audio provider: Failed to initialize resampling");
resample_ratio = (float)sample_rate / (float)codecContext->sample_rate;
}
/* libavcodec seems to give back invalid stream length values for Matroska files.
* As a workaround, we can use the overall file length.
*/
double length;
if(stream->duration == AV_NOPTS_VALUE)
length = (double)lavcfile->fctx->duration / AV_TIME_BASE;
else
length = (double)stream->duration * av_q2d(stream->time_base);
num_samples = (int64_t)(length * sample_rate); /* number of samples per channel */
buffer = (int16_t *)malloc(AVCODEC_MAX_AUDIO_FRAME_SIZE);
if (!buffer)
throw _T("ffmpeg audio provider: Failed to allocate audio decoding buffer, out of memory?");
leftover_samples = 0;
last_output_sample = -1;
} 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;
/* this exception disabled for now */
/* if (last_output_sample != start-1)
throw _T("ffmpeg audio provider: nonlinear access attempted, try loading audio to RAM or HD cache"); */
last_output_sample += count;
int64_t samples_to_decode = (num_samples - start) * channels; /* 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 * channels; /* times the number of channels */
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 * channels) - samples_to_decode) * bytes_per_sample);
/* do we have leftover samples from last time we were called? */
/* FIXME: this assumes that requests are always linear! attempts at random access give bogus results! */
if (leftover_samples > 0) {
int length = (samples_to_decode > leftover_samples) ? leftover_samples : samples_to_decode;
samples_to_decode -= length;
leftover_samples -= length;
/* put them in the output buffer */
samples_to_decode -= leftover_samples;
while (length > 0) {
*(_buf++) = *(overshoot_buffer++);
length--;
}
}
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_bytes, decoded_samples;
retval = avcodec_decode_audio2(codecContext, buffer, &temp_output_buffer_size, data, size);
if (retval <= 0)
throw _T("ffmpeg audio provider: failed to decode audio");
/* decoding succeeded but the output buffer is empty, go to next packet */
if (temp_output_buffer_size == 0) {
av_free_packet(&packet);
continue;
}
decoded_bytes = temp_output_buffer_size;
decoded_samples = decoded_bytes / bytes_per_sample; /* FIXME: stop assuming everything is 16-bit! */
size -= retval;
data += retval;
/* do we need to resample? */
if (rsct) {
/* allocate some memory to save the resampled data in */
int16_t *temp_output_buffer = (int16_t *)malloc(AVCODEC_MAX_AUDIO_FRAME_SIZE);
if (!temp_output_buffer)
throw _T("ffmpeg audio provider: Failed to allocate audio resampling buffer, out of memory?");
/* do the actual resampling */
decoded_samples = audio_resample(rsct, temp_output_buffer, buffer, decoded_samples / codecContext->channels);
/* did we end up with more samples than we were asked for? */
if (decoded_samples > samples_to_decode) {
/* in that case, count them */
leftover_samples = decoded_samples - samples_to_decode;
/* and put them aside for later */
memcpy(buffer, &temp_output_buffer[samples_to_decode+1], leftover_samples * bytes_per_sample);
overshoot_buffer = buffer;
/* output the other samples that didn't overflow */
memcpy(_buf, temp_output_buffer, samples_to_decode * bytes_per_sample);
_buf += samples_to_decode;
} else {
memcpy(_buf, temp_output_buffer, decoded_samples * bytes_per_sample);
_buf += decoded_samples;
}
free(temp_output_buffer);
} else { /* no resampling needed */
/* overflow? (as above) */
if (decoded_samples > samples_to_decode) {
/* count sheep^H^H^H^H^Hsamples */
leftover_samples = decoded_samples - samples_to_decode;
/* and put them aside for later (mm, lamb chops) */
overshoot_buffer = &buffer[samples_to_decode+1];
/* output the other samples that didn't overflow */
memcpy(_buf, buffer, samples_to_decode * bytes_per_sample);
_buf += samples_to_decode;
} else {
/* just do a straight copy to buffer */
memcpy(_buf, buffer, decoded_bytes);
_buf += decoded_samples;
}
}
samples_to_decode -= decoded_samples;
}
}
av_free_packet(&packet);
}
}
#endif