mirror of https://git.ffmpeg.org/ffmpeg.git
avplay: use avcodec_decode_audio4()
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parent
e2a2c49fcd
commit
f199f38573
43
avplay.c
43
avplay.c
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@ -153,18 +153,16 @@ typedef struct VideoState {
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AVStream *audio_st;
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PacketQueue audioq;
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int audio_hw_buf_size;
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/* samples output by the codec. we reserve more space for avsync
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compensation */
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DECLARE_ALIGNED(16,uint8_t,audio_buf1)[(AVCODEC_MAX_AUDIO_FRAME_SIZE * 3) / 2];
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DECLARE_ALIGNED(16,uint8_t,audio_buf2)[(AVCODEC_MAX_AUDIO_FRAME_SIZE * 3) / 2];
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uint8_t silence_buf[SDL_AUDIO_BUFFER_SIZE];
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uint8_t *audio_buf;
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uint8_t *audio_buf1;
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unsigned int audio_buf_size; /* in bytes */
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int audio_buf_index; /* in bytes */
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AVPacket audio_pkt_temp;
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AVPacket audio_pkt;
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enum AVSampleFormat audio_src_fmt;
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AVAudioConvert *reformat_ctx;
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AVFrame *frame;
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int show_audio; /* if true, display audio samples */
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int16_t sample_array[SAMPLE_ARRAY_SIZE];
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@ -2010,7 +2008,7 @@ static int audio_decode_frame(VideoState *is, double *pts_ptr)
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AVPacket *pkt_temp = &is->audio_pkt_temp;
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AVPacket *pkt = &is->audio_pkt;
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AVCodecContext *dec= is->audio_st->codec;
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int n, len1, data_size;
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int n, len1, data_size, got_frame;
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double pts;
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int new_packet = 0;
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int flush_complete = 0;
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@ -2018,13 +2016,16 @@ static int audio_decode_frame(VideoState *is, double *pts_ptr)
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for(;;) {
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/* NOTE: the audio packet can contain several frames */
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while (pkt_temp->size > 0 || (!pkt_temp->data && new_packet)) {
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if (!is->frame) {
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if (!(is->frame = avcodec_alloc_frame()))
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return AVERROR(ENOMEM);
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} else
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avcodec_get_frame_defaults(is->frame);
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if (flush_complete)
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break;
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new_packet = 0;
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data_size = sizeof(is->audio_buf1);
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len1 = avcodec_decode_audio3(dec,
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(int16_t *)is->audio_buf1, &data_size,
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pkt_temp);
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len1 = avcodec_decode_audio4(dec, is->frame, &got_frame, pkt_temp);
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if (len1 < 0) {
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/* if error, we skip the frame */
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pkt_temp->size = 0;
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@ -2034,12 +2035,15 @@ static int audio_decode_frame(VideoState *is, double *pts_ptr)
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pkt_temp->data += len1;
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pkt_temp->size -= len1;
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if (data_size <= 0) {
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if (!got_frame) {
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/* stop sending empty packets if the decoder is finished */
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if (!pkt_temp->data && dec->codec->capabilities & CODEC_CAP_DELAY)
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flush_complete = 1;
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continue;
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}
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data_size = av_samples_get_buffer_size(NULL, dec->channels,
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is->frame->nb_samples,
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dec->sample_fmt, 1);
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if (dec->sample_fmt != is->audio_src_fmt) {
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if (is->reformat_ctx)
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@ -2056,21 +2060,26 @@ static int audio_decode_frame(VideoState *is, double *pts_ptr)
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}
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if (is->reformat_ctx) {
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const void *ibuf[6]= {is->audio_buf1};
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void *obuf[6]= {is->audio_buf2};
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const void *ibuf[6]= { is->frame->data[0] };
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void *obuf[6];
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int istride[6]= {av_get_bytes_per_sample(dec->sample_fmt)};
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int ostride[6]= {2};
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int len= data_size/istride[0];
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obuf[0] = av_realloc(is->audio_buf1, FFALIGN(len * ostride[0], 32));
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if (!obuf[0]) {
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return AVERROR(ENOMEM);
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}
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is->audio_buf1 = obuf[0];
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if (av_audio_convert(is->reformat_ctx, obuf, ostride, ibuf, istride, len)<0) {
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printf("av_audio_convert() failed\n");
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break;
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}
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is->audio_buf= is->audio_buf2;
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is->audio_buf = is->audio_buf1;
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/* FIXME: existing code assume that data_size equals framesize*channels*2
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remove this legacy cruft */
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data_size= len*2;
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}else{
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is->audio_buf= is->audio_buf1;
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is->audio_buf = is->frame->data[0];
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}
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/* if no pts, then compute it */
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@ -2106,8 +2115,7 @@ static int audio_decode_frame(VideoState *is, double *pts_ptr)
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if (pkt->data == flush_pkt.data)
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avcodec_flush_buffers(dec);
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pkt_temp->data = pkt->data;
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pkt_temp->size = pkt->size;
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*pkt_temp = *pkt;
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/* if update the audio clock with the pts */
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if (pkt->pts != AV_NOPTS_VALUE) {
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@ -2275,6 +2283,9 @@ static void stream_component_close(VideoState *is, int stream_index)
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if (is->reformat_ctx)
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av_audio_convert_free(is->reformat_ctx);
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is->reformat_ctx = NULL;
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av_freep(&is->audio_buf1);
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is->audio_buf = NULL;
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av_freep(&is->frame);
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if (is->rdft) {
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av_rdft_end(is->rdft);
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