libavcodec/libvo-aacenc.c - third_party/ffmpeg - Git at Google

 /*
  * AAC encoder wrapper
  * Copyright (c) 2010 Martin Storsjo
  *
  * This file is part of FFmpeg.
  *
  * FFmpeg is free software; you can redistribute it and/or
  * modify it under the terms of the GNU Lesser General Public
  * License as published by the Free Software Foundation; either
  * version 2.1 of the License, or (at your option) any later version.
  *
  * FFmpeg is distributed in the hope that it will be useful,
  * but WITHOUT ANY WARRANTY; without even the implied warranty of
  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
  * Lesser General Public License for more details.
  *
  * You should have received a copy of the GNU Lesser General Public
  * License along with FFmpeg; if not, write to the Free Software
  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  */

 #include <vo-aacenc/voAAC.h>
 #include <vo-aacenc/cmnMemory.h>

 #include "avcodec.h"
 #include "audio_frame_queue.h"
 #include "internal.h"
 #include "mpeg4audio.h"

 #define FRAME_SIZE 1024
 #define ENC_DELAY  1600

 typedef struct AACContext {
     VO_AUDIO_CODECAPI codec_api;
     VO_HANDLE handle;
     VO_MEM_OPERATOR mem_operator;
     VO_CODEC_INIT_USERDATA user_data;
     VO_PBYTE end_buffer;
     AudioFrameQueue afq;
     int last_frame;
     int last_samples;
 } AACContext;


 static int aac_encode_close(AVCodecContext *avctx)
 {
     AACContext *s = avctx->priv_data;

     s->codec_api.Uninit(s->handle);
     av_freep(&avctx->extradata);
     ff_af_queue_close(&s->afq);
     av_freep(&s->end_buffer);

     return 0;
 }

 static av_cold int aac_encode_init(AVCodecContext *avctx)
 {
     AACContext *s = avctx->priv_data;
     AACENC_PARAM params = { 0 };
     int index, ret;

     avctx->frame_size = FRAME_SIZE;
     avctx->initial_padding = ENC_DELAY;
     s->last_frame     = 2;
     ff_af_queue_init(avctx, &s->afq);

     s->end_buffer = av_mallocz_array(avctx->channels, avctx->frame_size * 2);
     if (!s->end_buffer) {
         ret = AVERROR(ENOMEM);
         goto error;
     }

     voGetAACEncAPI(&s->codec_api);

     s->mem_operator.Alloc = cmnMemAlloc;
     s->mem_operator.Copy = cmnMemCopy;
     s->mem_operator.Free = cmnMemFree;
     s->mem_operator.Set = cmnMemSet;
     s->mem_operator.Check = cmnMemCheck;
     s->user_data.memflag = VO_IMF_USERMEMOPERATOR;
     s->user_data.memData = &s->mem_operator;
     s->codec_api.Init(&s->handle, VO_AUDIO_CodingAAC, &s->user_data);

     params.sampleRate = avctx->sample_rate;
     params.bitRate    = avctx->bit_rate;
     params.nChannels  = avctx->channels;
     params.adtsUsed   = !(avctx->flags & AV_CODEC_FLAG_GLOBAL_HEADER);
     if (s->codec_api.SetParam(s->handle, VO_PID_AAC_ENCPARAM, &params)
         != VO_ERR_NONE) {
         av_log(avctx, AV_LOG_ERROR, "Unable to set encoding parameters\n");
         ret = AVERROR(EINVAL);
         goto error;
     }

     for (index = 0; index < 16; index++)
         if (avctx->sample_rate == avpriv_mpeg4audio_sample_rates[index])
             break;
     if (index == 16) {
         av_log(avctx, AV_LOG_ERROR, "Unsupported sample rate %d\n",
                                     avctx->sample_rate);
         ret = AVERROR(ENOSYS);
         goto error;
     }
     if (avctx->flags & AV_CODEC_FLAG_GLOBAL_HEADER) {
         avctx->extradata_size = 2;
         avctx->extradata      = av_mallocz(avctx->extradata_size +
                                            AV_INPUT_BUFFER_PADDING_SIZE);
         if (!avctx->extradata) {
             ret = AVERROR(ENOMEM);
             goto error;
         }

         avctx->extradata[0] = 0x02 << 3 | index >> 1;
         avctx->extradata[1] = (index & 0x01) << 7 | avctx->channels << 3;
     }
     return 0;
 error:
     aac_encode_close(avctx);
     return ret;
 }

 static int aac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
                             const AVFrame *frame, int *got_packet_ptr)
 {
     AACContext *s = avctx->priv_data;
     VO_CODECBUFFER input = { 0 }, output = { 0 };
     VO_AUDIO_OUTPUTINFO output_info = { { 0 } };
     VO_PBYTE samples;
     int ret;

     /* handle end-of-stream small frame and flushing */
     if (!frame) {
         if (s->last_frame <= 0)
             return 0;
         if (s->last_samples > 0 && s->last_samples < ENC_DELAY - FRAME_SIZE) {
             s->last_samples = 0;
             s->last_frame--;
         }
         s->last_frame--;
         memset(s->end_buffer, 0, 2 * avctx->channels * avctx->frame_size);
         samples = s->end_buffer;
     } else {
         if (frame->nb_samples < avctx->frame_size) {
             s->last_samples = frame->nb_samples;
             memcpy(s->end_buffer, frame->data[0], 2 * avctx->channels * frame->nb_samples);
             samples = s->end_buffer;
         } else {
             samples = (VO_PBYTE)frame->data[0];
         }
         /* add current frame to the queue */
         if ((ret = ff_af_queue_add(&s->afq, frame)) < 0)
             return ret;
     }

     if ((ret = ff_alloc_packet2(avctx, avpkt, FFMAX(8192, 768 * avctx->channels), 0)) < 0)
         return ret;

     input.Buffer  = samples;
     input.Length  = 2 * avctx->channels * avctx->frame_size;
     output.Buffer = avpkt->data;
     output.Length = avpkt->size;

     s->codec_api.SetInputData(s->handle, &input);
     if (s->codec_api.GetOutputData(s->handle, &output, &output_info)
         != VO_ERR_NONE) {
         av_log(avctx, AV_LOG_ERROR, "Unable to encode frame\n");
         return AVERROR(EINVAL);
     }

     /* Get the next frame pts/duration */
     ff_af_queue_remove(&s->afq, avctx->frame_size, &avpkt->pts,
                        &avpkt->duration);

     avpkt->size = output.Length;
     *got_packet_ptr = 1;
     return 0;
 }

 /* duplicated from avpriv_mpeg4audio_sample_rates to avoid shared build
  * failures */
 static const int mpeg4audio_sample_rates[16] = {
     96000, 88200, 64000, 48000, 44100, 32000,
     24000, 22050, 16000, 12000, 11025, 8000, 7350
 };

 AVCodec ff_libvo_aacenc_encoder = {
     .name           = "libvo_aacenc",
     .long_name      = NULL_IF_CONFIG_SMALL("Android VisualOn AAC (Advanced Audio Coding)"),
     .type           = AVMEDIA_TYPE_AUDIO,
     .id             = AV_CODEC_ID_AAC,
     .priv_data_size = sizeof(AACContext),
     .init           = aac_encode_init,
     .encode2        = aac_encode_frame,
     .close          = aac_encode_close,
     .supported_samplerates = mpeg4audio_sample_rates,
     .capabilities   = AV_CODEC_CAP_SMALL_LAST_FRAME | AV_CODEC_CAP_DELAY,
     .sample_fmts    = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16,
                                                      AV_SAMPLE_FMT_NONE },
 };
	/*
	* AAC encoder wrapper
	* Copyright (c) 2010 Martin Storsjo
	*
	* This file is part of FFmpeg.
	*
	* FFmpeg is free software; you can redistribute it and/or
	* modify it under the terms of the GNU Lesser General Public
	* License as published by the Free Software Foundation; either
	* version 2.1 of the License, or (at your option) any later version.
	*
	* FFmpeg is distributed in the hope that it will be useful,
	* but WITHOUT ANY WARRANTY; without even the implied warranty of
	* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
	* Lesser General Public License for more details.
	*
	* You should have received a copy of the GNU Lesser General Public
	* License along with FFmpeg; if not, write to the Free Software
	* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
	*/

	#include <vo-aacenc/voAAC.h>
	#include <vo-aacenc/cmnMemory.h>

	#include "avcodec.h"
	#include "audio_frame_queue.h"
	#include "internal.h"
	#include "mpeg4audio.h"

	#define FRAME_SIZE 1024
	#define ENC_DELAY 1600

	typedef struct AACContext {
	VO_AUDIO_CODECAPI codec_api;
	VO_HANDLE handle;
	VO_MEM_OPERATOR mem_operator;
	VO_CODEC_INIT_USERDATA user_data;
	VO_PBYTE end_buffer;
	AudioFrameQueue afq;
	int last_frame;
	int last_samples;
	} AACContext;


	static int aac_encode_close(AVCodecContext *avctx)
	{
	AACContext *s = avctx->priv_data;

	s->codec_api.Uninit(s->handle);
	av_freep(&avctx->extradata);
	ff_af_queue_close(&s->afq);
	av_freep(&s->end_buffer);

	return 0;
	}

	static av_cold int aac_encode_init(AVCodecContext *avctx)
	{
	AACContext *s = avctx->priv_data;
	AACENC_PARAM params = { 0 };
	int index, ret;

	avctx->frame_size = FRAME_SIZE;
	avctx->initial_padding = ENC_DELAY;
	s->last_frame = 2;
	ff_af_queue_init(avctx, &s->afq);

	s->end_buffer = av_mallocz_array(avctx->channels, avctx->frame_size * 2);
	if (!s->end_buffer) {
	ret = AVERROR(ENOMEM);
	goto error;
	}

	voGetAACEncAPI(&s->codec_api);

	s->mem_operator.Alloc = cmnMemAlloc;
	s->mem_operator.Copy = cmnMemCopy;
	s->mem_operator.Free = cmnMemFree;
	s->mem_operator.Set = cmnMemSet;
	s->mem_operator.Check = cmnMemCheck;
	s->user_data.memflag = VO_IMF_USERMEMOPERATOR;
	s->user_data.memData = &s->mem_operator;
	s->codec_api.Init(&s->handle, VO_AUDIO_CodingAAC, &s->user_data);

	params.sampleRate = avctx->sample_rate;
	params.bitRate = avctx->bit_rate;
	params.nChannels = avctx->channels;
	params.adtsUsed = !(avctx->flags & AV_CODEC_FLAG_GLOBAL_HEADER);
	if (s->codec_api.SetParam(s->handle, VO_PID_AAC_ENCPARAM, &params)
	!= VO_ERR_NONE) {
	av_log(avctx, AV_LOG_ERROR, "Unable to set encoding parameters\n");
	ret = AVERROR(EINVAL);
	goto error;
	}

	for (index = 0; index < 16; index++)
	if (avctx->sample_rate == avpriv_mpeg4audio_sample_rates[index])
	break;
	if (index == 16) {
	av_log(avctx, AV_LOG_ERROR, "Unsupported sample rate %d\n",
	avctx->sample_rate);
	ret = AVERROR(ENOSYS);
	goto error;
	}
	if (avctx->flags & AV_CODEC_FLAG_GLOBAL_HEADER) {
	avctx->extradata_size = 2;
	avctx->extradata = av_mallocz(avctx->extradata_size +
	AV_INPUT_BUFFER_PADDING_SIZE);
	if (!avctx->extradata) {
	ret = AVERROR(ENOMEM);
	goto error;
	}

	avctx->extradata[0] = 0x02 << 3 \| index >> 1;
	avctx->extradata[1] = (index & 0x01) << 7 \| avctx->channels << 3;
	}
	return 0;
	error:
	aac_encode_close(avctx);
	return ret;
	}

	static int aac_encode_frame(AVCodecContext avctx, AVPacket avpkt,
	const AVFrame frame, int got_packet_ptr)
	{
	AACContext *s = avctx->priv_data;
	VO_CODECBUFFER input = { 0 }, output = { 0 };
	VO_AUDIO_OUTPUTINFO output_info = { { 0 } };
	VO_PBYTE samples;
	int ret;

	/* handle end-of-stream small frame and flushing */
	if (!frame) {
	if (s->last_frame <= 0)
	return 0;
	if (s->last_samples > 0 && s->last_samples < ENC_DELAY - FRAME_SIZE) {
	s->last_samples = 0;
	s->last_frame--;
	}
	s->last_frame--;
	memset(s->end_buffer, 0, 2 * avctx->channels * avctx->frame_size);
	samples = s->end_buffer;
	} else {
	if (frame->nb_samples < avctx->frame_size) {
	s->last_samples = frame->nb_samples;
	memcpy(s->end_buffer, frame->data[0], 2 * avctx->channels * frame->nb_samples);
	samples = s->end_buffer;
	} else {
	samples = (VO_PBYTE)frame->data[0];
	}
	/* add current frame to the queue */
	if ((ret = ff_af_queue_add(&s->afq, frame)) < 0)
	return ret;
	}

	if ((ret = ff_alloc_packet2(avctx, avpkt, FFMAX(8192, 768 * avctx->channels), 0)) < 0)
	return ret;

	input.Buffer = samples;
	input.Length = 2 * avctx->channels * avctx->frame_size;
	output.Buffer = avpkt->data;
	output.Length = avpkt->size;

	s->codec_api.SetInputData(s->handle, &input);
	if (s->codec_api.GetOutputData(s->handle, &output, &output_info)
	!= VO_ERR_NONE) {
	av_log(avctx, AV_LOG_ERROR, "Unable to encode frame\n");
	return AVERROR(EINVAL);
	}

	/* Get the next frame pts/duration */
	ff_af_queue_remove(&s->afq, avctx->frame_size, &avpkt->pts,
	&avpkt->duration);

	avpkt->size = output.Length;
	*got_packet_ptr = 1;
	return 0;
	}

	/* duplicated from avpriv_mpeg4audio_sample_rates to avoid shared build
	* failures */
	static const int mpeg4audio_sample_rates[16] = {
	96000, 88200, 64000, 48000, 44100, 32000,
	24000, 22050, 16000, 12000, 11025, 8000, 7350
	};

	AVCodec ff_libvo_aacenc_encoder = {
	.name = "libvo_aacenc",
	.long_name = NULL_IF_CONFIG_SMALL("Android VisualOn AAC (Advanced Audio Coding)"),
	.type = AVMEDIA_TYPE_AUDIO,
	.id = AV_CODEC_ID_AAC,
	.priv_data_size = sizeof(AACContext),
	.init = aac_encode_init,
	.encode2 = aac_encode_frame,
	.close = aac_encode_close,
	.supported_samplerates = mpeg4audio_sample_rates,
	.capabilities = AV_CODEC_CAP_SMALL_LAST_FRAME \| AV_CODEC_CAP_DELAY,
	.sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16,
	AV_SAMPLE_FMT_NONE },
	};