From 5a4476e229748348b16b56a81e79e5c0422be4b9 Mon Sep 17 00:00:00 2001 From: Peter Ross Date: Sat, 2 Aug 2008 05:01:30 +0000 Subject: [PATCH] Add sample format converter to FFplay. Originally committed as revision 14508 to svn://svn.ffmpeg.org/ffmpeg/trunk --- ffplay.c | 51 ++++++++++++++++++++++++++++++++++++++++++++++----- 1 file changed, 46 insertions(+), 5 deletions(-) diff --git a/ffplay.c b/ffplay.c index 31b74ae96a..33ee806009 100644 --- a/ffplay.c +++ b/ffplay.c @@ -26,6 +26,7 @@ #include "libavformat/rtsp.h" #include "libavdevice/avdevice.h" #include "libswscale/swscale.h" +#include "libavcodec/audioconvert.h" #include "cmdutils.h" @@ -127,12 +128,16 @@ typedef struct VideoState { int audio_hw_buf_size; /* samples output by the codec. we reserve more space for avsync compensation */ - DECLARE_ALIGNED(16,uint8_t,audio_buf[(AVCODEC_MAX_AUDIO_FRAME_SIZE * 3) / 2]); + DECLARE_ALIGNED(16,uint8_t,audio_buf1[(AVCODEC_MAX_AUDIO_FRAME_SIZE * 3) / 2]); + DECLARE_ALIGNED(16,uint8_t,audio_buf2[(AVCODEC_MAX_AUDIO_FRAME_SIZE * 3) / 2]); + uint8_t *audio_buf; unsigned int audio_buf_size; /* in bytes */ int audio_buf_index; /* in bytes */ AVPacket audio_pkt; uint8_t *audio_pkt_data; int audio_pkt_size; + enum SampleFormat audio_src_fmt; + AVAudioConvert *reformat_ctx; int show_audio; /* if true, display audio samples */ int16_t sample_array[SAMPLE_ARRAY_SIZE]; @@ -1568,7 +1573,7 @@ static int synchronize_audio(VideoState *is, short *samples, } /* decode one audio frame and returns its uncompressed size */ -static int audio_decode_frame(VideoState *is, uint8_t *audio_buf, int buf_size, double *pts_ptr) +static int audio_decode_frame(VideoState *is, double *pts_ptr) { AVPacket *pkt = &is->audio_pkt; AVCodecContext *dec= is->audio_st->codec; @@ -1578,9 +1583,9 @@ static int audio_decode_frame(VideoState *is, uint8_t *audio_buf, int buf_size, for(;;) { /* NOTE: the audio packet can contain several frames */ while (is->audio_pkt_size > 0) { - data_size = buf_size; + data_size = sizeof(is->audio_buf1); len1 = avcodec_decode_audio2(dec, - (int16_t *)audio_buf, &data_size, + (int16_t *)is->audio_buf1, &data_size, is->audio_pkt_data, is->audio_pkt_size); if (len1 < 0) { /* if error, we skip the frame */ @@ -1592,6 +1597,39 @@ static int audio_decode_frame(VideoState *is, uint8_t *audio_buf, int buf_size, is->audio_pkt_size -= len1; if (data_size <= 0) continue; + + if (dec->sample_fmt != is->audio_src_fmt) { + if (is->reformat_ctx) + av_audio_convert_free(is->reformat_ctx); + is->reformat_ctx= av_audio_convert_alloc(SAMPLE_FMT_S16, 1, + dec->sample_fmt, 1, NULL, 0); + if (!is->reformat_ctx) { + fprintf(stderr, "Cannot convert %s sample format to %s sample format\n", + avcodec_get_sample_fmt_name(dec->sample_fmt), + avcodec_get_sample_fmt_name(SAMPLE_FMT_S16)); + break; + } + is->audio_src_fmt= dec->sample_fmt; + } + + if (is->reformat_ctx) { + const void *ibuf[6]= {is->audio_buf1}; + void *obuf[6]= {is->audio_buf2}; + int istride[6]= {av_get_bits_per_sample_format(dec->sample_fmt)/8}; + int ostride[6]= {2}; + int len= data_size/istride[0]; + if (av_audio_convert(is->reformat_ctx, obuf, ostride, ibuf, istride, len)<0) { + printf("av_audio_convert() failed\n"); + break; + } + is->audio_buf= is->audio_buf2; + /* FIXME: existing code assume that data_size equals framesize*channels*2 + remove this legacy cruft */ + data_size= len*2; + }else{ + is->audio_buf= is->audio_buf1; + } + /* if no pts, then compute it */ pts = is->audio_clock; *pts_ptr = pts; @@ -1655,7 +1693,7 @@ static void sdl_audio_callback(void *opaque, Uint8 *stream, int len) while (len > 0) { if (is->audio_buf_index >= is->audio_buf_size) { - audio_size = audio_decode_frame(is, is->audio_buf, sizeof(is->audio_buf), &pts); + audio_size = audio_decode_frame(is, &pts); if (audio_size < 0) { /* if error, just output silence */ is->audio_buf_size = 1024; @@ -1731,6 +1769,7 @@ static int stream_component_open(VideoState *is, int stream_index) return -1; } is->audio_hw_buf_size = spec.size; + is->audio_src_fmt= SAMPLE_FMT_S16; } if(thread_count>1) @@ -1797,6 +1836,8 @@ static void stream_component_close(VideoState *is, int stream_index) SDL_CloseAudio(); packet_queue_end(&is->audioq); + if (is->reformat_ctx) + av_audio_convert_free(is->reformat_ctx); break; case CODEC_TYPE_VIDEO: packet_queue_abort(&is->videoq);