Aegisub/aegisub/audio_provider_lavc.cpp

// 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 < (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("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) {
					/* do the actual resampling */
					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, but first make noise if we got an overflow */
					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