X-Git-Url: https://git.sesse.net/?a=blobdiff_plain;f=libavcodec%2Froqaudioenc.c;h=cebc53cce506269f4c9711786f11add583bbb981;hb=c5063e0348db97626aecc17c42fd41718fd62f13;hp=f6bd726c4ff7fc50552f6885857e56c0830afc86;hpb=d36beb3f6902b1217beda576aa18abf7eb72b03c;p=ffmpeg diff --git a/libavcodec/roqaudioenc.c b/libavcodec/roqaudioenc.c index f6bd726c4ff..cebc53cce50 100644 --- a/libavcodec/roqaudioenc.c +++ b/libavcodec/roqaudioenc.c @@ -4,20 +4,20 @@ * Copyright (c) 2005 Eric Lasota * Based on RoQ specs (c)2001 Tim Ferguson * - * This file is part of FFmpeg. + * This file is part of Libav. * - * FFmpeg is free software; you can redistribute it and/or + * Libav 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, + * Libav 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 + * License along with Libav; if not, write to the Free Software * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ @@ -25,9 +25,8 @@ #include "avcodec.h" #include "bytestream.h" -#define ROQ_FIRST_FRAME_SIZE (735*8) #define ROQ_FRAME_SIZE 735 - +#define ROQ_HEADER_SIZE 8 #define MAX_DPCM (127*127) @@ -35,33 +34,59 @@ typedef struct { short lastSample[2]; + int input_frames; + int buffered_samples; + int16_t *frame_buffer; } ROQDPCMContext; + +static av_cold int roq_dpcm_encode_close(AVCodecContext *avctx) +{ + ROQDPCMContext *context = avctx->priv_data; + + av_freep(&avctx->coded_frame); + av_freep(&context->frame_buffer); + + return 0; +} + static av_cold int roq_dpcm_encode_init(AVCodecContext *avctx) { ROQDPCMContext *context = avctx->priv_data; + int ret; if (avctx->channels > 2) { av_log(avctx, AV_LOG_ERROR, "Audio must be mono or stereo\n"); - return -1; + return AVERROR(EINVAL); } if (avctx->sample_rate != 22050) { av_log(avctx, AV_LOG_ERROR, "Audio must be 22050 Hz\n"); - return -1; - } - if (avctx->sample_fmt != AV_SAMPLE_FMT_S16) { - av_log(avctx, AV_LOG_ERROR, "Audio must be signed 16-bit\n"); - return -1; + return AVERROR(EINVAL); } - avctx->frame_size = ROQ_FIRST_FRAME_SIZE; + avctx->frame_size = ROQ_FRAME_SIZE; + avctx->bit_rate = (ROQ_HEADER_SIZE + ROQ_FRAME_SIZE * avctx->channels) * + (22050 / ROQ_FRAME_SIZE) * 8; + + context->frame_buffer = av_malloc(8 * ROQ_FRAME_SIZE * avctx->channels * + sizeof(*context->frame_buffer)); + if (!context->frame_buffer) { + ret = AVERROR(ENOMEM); + goto error; + } context->lastSample[0] = context->lastSample[1] = 0; avctx->coded_frame= avcodec_alloc_frame(); - avctx->coded_frame->key_frame= 1; + if (!avctx->coded_frame) { + ret = AVERROR(ENOMEM); + goto error; + } return 0; +error: + roq_dpcm_encode_close(avctx); + return ret; } static unsigned char dpcm_predict(short *previous, short current) @@ -107,25 +132,45 @@ static unsigned char dpcm_predict(short *previous, short current) static int roq_dpcm_encode_frame(AVCodecContext *avctx, unsigned char *frame, int buf_size, void *data) { - int i, samples, stereo, ch; - const short *in; - unsigned char *out; - + int i, stereo, data_size; + const int16_t *in = data; + uint8_t *out = frame; ROQDPCMContext *context = avctx->priv_data; stereo = (avctx->channels == 2); + if (!data && context->input_frames >= 8) + return 0; + + if (data && context->input_frames < 8) { + memcpy(&context->frame_buffer[context->buffered_samples * avctx->channels], + in, avctx->frame_size * avctx->channels * sizeof(*in)); + context->buffered_samples += avctx->frame_size; + if (context->input_frames < 7) { + context->input_frames++; + return 0; + } + in = context->frame_buffer; + } + if (stereo) { context->lastSample[0] &= 0xFF00; context->lastSample[1] &= 0xFF00; } - out = frame; - in = data; + if (context->input_frames == 7 || !data) + data_size = avctx->channels * context->buffered_samples; + else + data_size = avctx->channels * avctx->frame_size; + + if (buf_size < ROQ_HEADER_SIZE + data_size) { + av_log(avctx, AV_LOG_ERROR, "output buffer is too small\n"); + return AVERROR(EINVAL); + } bytestream_put_byte(&out, stereo ? 0x21 : 0x20); bytestream_put_byte(&out, 0x10); - bytestream_put_le32(&out, avctx->frame_size*avctx->channels); + bytestream_put_le32(&out, data_size); if (stereo) { bytestream_put_byte(&out, (context->lastSample[1])>>8); @@ -134,34 +179,26 @@ static int roq_dpcm_encode_frame(AVCodecContext *avctx, bytestream_put_le16(&out, context->lastSample[0]); /* Write the actual samples */ - samples = avctx->frame_size; - for (i=0; ichannels; ch++) - *out++ = dpcm_predict(&context->lastSample[ch], *in++); + for (i = 0; i < data_size; i++) + *out++ = dpcm_predict(&context->lastSample[i & 1], *in++); - /* Use smaller frames from now on */ - avctx->frame_size = ROQ_FRAME_SIZE; + context->input_frames++; + if (!data) + context->input_frames = FFMAX(context->input_frames, 8); /* Return the result size */ - return out - frame; -} - -static av_cold int roq_dpcm_encode_close(AVCodecContext *avctx) -{ - av_freep(&avctx->coded_frame); - - return 0; + return ROQ_HEADER_SIZE + data_size; } AVCodec ff_roq_dpcm_encoder = { - "roq_dpcm", - AVMEDIA_TYPE_AUDIO, - CODEC_ID_ROQ_DPCM, - sizeof(ROQDPCMContext), - roq_dpcm_encode_init, - roq_dpcm_encode_frame, - roq_dpcm_encode_close, - NULL, + .name = "roq_dpcm", + .type = AVMEDIA_TYPE_AUDIO, + .id = CODEC_ID_ROQ_DPCM, + .priv_data_size = sizeof(ROQDPCMContext), + .init = roq_dpcm_encode_init, + .encode = roq_dpcm_encode_frame, + .close = roq_dpcm_encode_close, + .capabilities = CODEC_CAP_DELAY, .sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE}, .long_name = NULL_IF_CONFIG_SMALL("id RoQ DPCM"), };