diff --git a/ffplay.c b/ffplay.c index d346bc5cc9..f56c2d70ab 100644 --- a/ffplay.c +++ b/ffplay.c @@ -38,6 +38,7 @@ #include "libavcodec/audioconvert.h" #include "libavutil/opt.h" #include "libavcodec/avfft.h" +#include "libswresample/swresample.h" #if CONFIG_AVFILTER # include "libavfilter/avcodec.h" @@ -152,9 +153,9 @@ typedef struct VideoState { PacketQueue audioq; int audio_hw_buf_size; /* samples output by the codec. we reserve more space for avsync - compensation */ - 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]; + compensation, resampling and format conversion */ + DECLARE_ALIGNED(16,uint8_t,audio_buf1)[AVCODEC_MAX_AUDIO_FRAME_SIZE * 4]; + DECLARE_ALIGNED(16,uint8_t,audio_buf2)[AVCODEC_MAX_AUDIO_FRAME_SIZE * 4]; uint8_t *audio_buf; unsigned int audio_buf_size; /* in bytes */ int audio_buf_index; /* in bytes */ @@ -162,7 +163,14 @@ typedef struct VideoState { AVPacket audio_pkt_temp; AVPacket audio_pkt; enum AVSampleFormat audio_src_fmt; - AVAudioConvert *reformat_ctx; + enum AVSampleFormat audio_tgt_fmt; + int audio_src_channels; + int audio_tgt_channels; + int64_t audio_src_channel_layout; + int64_t audio_tgt_channel_layout; + int audio_src_freq; + int audio_tgt_freq; + struct SwrContext *swr_ctx; double audio_current_pts; double audio_current_pts_drift; @@ -732,7 +740,7 @@ static void video_audio_display(VideoState *s) nb_freq= 1<<(rdft_bits-1); /* compute display index : center on currently output samples */ - channels = s->audio_st->codec->channels; + channels = s->audio_tgt_channels; nb_display_channels = channels; if (!s->paused) { int data_used= s->show_mode == SHOW_MODE_WAVES ? s->width : (2*nb_freq); @@ -744,7 +752,7 @@ static void video_audio_display(VideoState *s) the last buffer computation */ if (audio_callback_time) { time_diff = av_gettime() - audio_callback_time; - delay -= (time_diff * s->audio_st->codec->sample_rate) / 1000000; + delay -= (time_diff * s->audio_tgt_freq) / 1000000; } delay += 2*data_used; @@ -1922,7 +1930,7 @@ static int synchronize_audio(VideoState *is, short *samples, int n, samples_size; double ref_clock; - n = 2 * is->audio_st->codec->channels; + n = av_get_bytes_per_sample(is->audio_tgt_fmt) * is->audio_tgt_channels; samples_size = samples_size1; /* if not master, then we try to remove or add samples to correct the clock */ @@ -1944,15 +1952,15 @@ static int synchronize_audio(VideoState *is, short *samples, avg_diff = is->audio_diff_cum * (1.0 - is->audio_diff_avg_coef); if (fabs(avg_diff) >= is->audio_diff_threshold) { - wanted_size = samples_size + ((int)(diff * is->audio_st->codec->sample_rate) * n); + wanted_size = samples_size + ((int)(diff * is->audio_tgt_freq) * n); nb_samples = samples_size / n; min_size = ((nb_samples * (100 - SAMPLE_CORRECTION_PERCENT_MAX)) / 100) * n; max_size = ((nb_samples * (100 + SAMPLE_CORRECTION_PERCENT_MAX)) / 100) * n; if (wanted_size < min_size) wanted_size = min_size; - else if (wanted_size > max_size) - wanted_size = max_size; + else if (wanted_size > FFMIN3(max_size, sizeof(is->audio_buf1), sizeof(is->audio_buf2))) + wanted_size = FFMIN3(max_size, sizeof(is->audio_buf1), sizeof(is->audio_buf2)); /* add or remove samples to correction the synchro */ if (wanted_size < samples_size) { @@ -1995,7 +2003,8 @@ static int audio_decode_frame(VideoState *is, double *pts_ptr) AVPacket *pkt_temp = &is->audio_pkt_temp; AVPacket *pkt = &is->audio_pkt; AVCodecContext *dec= is->audio_st->codec; - int n, len1, data_size; + int len1, len2, data_size, resampled_data_size; + int64_t dec_channel_layout; double pts; int new_packet = 0; int flush_complete = 0; @@ -2026,44 +2035,54 @@ static int audio_decode_frame(VideoState *is, double *pts_ptr) 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(AV_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", - av_get_sample_fmt_name(dec->sample_fmt), - av_get_sample_fmt_name(AV_SAMPLE_FMT_S16)); - break; - } - is->audio_src_fmt= dec->sample_fmt; - } + dec_channel_layout = (dec->channel_layout && dec->channels == av_get_channel_layout_nb_channels(dec->channel_layout)) ? dec->channel_layout : av_get_default_channel_layout(dec->channels); - if (is->reformat_ctx) { - const void *ibuf[6]= {is->audio_buf1}; - void *obuf[6]= {is->audio_buf2}; - int istride[6]= {av_get_bytes_per_sample(dec->sample_fmt)}; - 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"); + if (dec->sample_fmt != is->audio_src_fmt || dec_channel_layout != is->audio_src_channel_layout || dec->sample_rate != is->audio_src_freq) { + if (is->swr_ctx) + swr_free(&is->swr_ctx); + is->swr_ctx = swr_alloc2(NULL, is->audio_tgt_channel_layout, is->audio_tgt_fmt, is->audio_tgt_freq, + dec_channel_layout, dec->sample_fmt, dec->sample_rate, + 0, NULL); + if (!is->swr_ctx || swr_init(is->swr_ctx) < 0) { + fprintf(stderr, "Cannot create sample rate converter for conversion of %d Hz %s %d channels to %d Hz %s %d channels!\n", + dec->sample_rate, + av_get_sample_fmt_name(dec->sample_fmt), + dec->channels, + is->audio_tgt_freq, + av_get_sample_fmt_name(is->audio_tgt_fmt), + is->audio_tgt_channels); 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_src_channel_layout = dec_channel_layout; + is->audio_src_channels = dec->channels; + is->audio_src_freq = dec->sample_rate; + is->audio_src_fmt = dec->sample_fmt; + } + + resampled_data_size = data_size; + if (is->swr_ctx) { + const uint8_t *in[] = {is->audio_buf1}; + uint8_t *out[] = {is->audio_buf2}; + len2 = swr_convert(is->swr_ctx, out, sizeof(is->audio_buf2) / is->audio_tgt_channels / av_get_bytes_per_sample(is->audio_tgt_fmt), + in, data_size / dec->channels / av_get_bytes_per_sample(dec->sample_fmt)); + if (len2 < 0) { + fprintf(stderr, "audio_resample() failed\n"); + break; + } + if (len2 == sizeof(is->audio_buf2) / is->audio_tgt_channels / av_get_bytes_per_sample(is->audio_tgt_fmt)) { + fprintf(stderr, "warning: audio buffer is probably too small\n"); + swr_init(is->swr_ctx); + } + is->audio_buf = is->audio_buf2; + resampled_data_size = len2 * is->audio_tgt_channels * av_get_bytes_per_sample(is->audio_tgt_fmt); + } else { is->audio_buf= is->audio_buf1; } /* if no pts, then compute it */ pts = is->audio_clock; *pts_ptr = pts; - n = 2 * dec->channels; - is->audio_clock += (double)data_size / - (double)(n * dec->sample_rate); + is->audio_clock += (double)data_size / (dec->channels * dec->sample_rate * av_get_bytes_per_sample(dec->sample_fmt)); #ifdef DEBUG { static double last_clock; @@ -2073,7 +2092,7 @@ static int audio_decode_frame(VideoState *is, double *pts_ptr) last_clock = is->audio_clock; } #endif - return data_size; + return resampled_data_size; } /* free the current packet */ @@ -2117,7 +2136,7 @@ static void sdl_audio_callback(void *opaque, Uint8 *stream, int len) if (audio_size < 0) { /* if error, just output silence */ is->audio_buf = is->audio_buf1; - is->audio_buf_size = 1024; + is->audio_buf_size = 256 * is->audio_tgt_channels * av_get_bytes_per_sample(is->audio_tgt_fmt); memset(is->audio_buf, 0, is->audio_buf_size); } else { if (is->show_mode != SHOW_MODE_VIDEO) @@ -2136,8 +2155,7 @@ static void sdl_audio_callback(void *opaque, Uint8 *stream, int len) stream += len1; is->audio_buf_index += len1; } - bytes_per_sec = is->audio_st->codec->sample_rate * - 2 * is->audio_st->codec->channels; + bytes_per_sec = is->audio_tgt_freq * is->audio_tgt_channels * av_get_bytes_per_sample(is->audio_tgt_fmt); is->audio_write_buf_size = is->audio_buf_size - is->audio_buf_index; /* Let's assume the audio driver that is used by SDL has two periods. */ is->audio_current_pts = is->audio_clock - (double)(2 * is->audio_hw_buf_size + is->audio_write_buf_size) / bytes_per_sec; @@ -2153,6 +2171,7 @@ static int stream_component_open(VideoState *is, int stream_index) SDL_AudioSpec wanted_spec, spec; AVDictionary *opts; AVDictionaryEntry *t = NULL; + int64_t wanted_channel_layout = 0; if (stream_index < 0 || stream_index >= ic->nb_streams) return -1; @@ -2184,12 +2203,14 @@ static int stream_component_open(VideoState *is, int stream_index) avctx->flags |= CODEC_FLAG_EMU_EDGE; if (avctx->codec_type == AVMEDIA_TYPE_AUDIO) { - if(avctx->sample_rate <= 0 || avctx->channels <= 0){ - fprintf(stderr, "Invalid sample rate or channel count\n"); + wanted_channel_layout = (avctx->channel_layout && avctx->channels == av_get_channel_layout_nb_channels(avctx->channels)) ? avctx->channel_layout : av_get_default_channel_layout(avctx->channels); + wanted_channel_layout &= ~AV_CH_LAYOUT_STEREO_DOWNMIX; + wanted_spec.channels = av_get_channel_layout_nb_channels(wanted_channel_layout); + wanted_spec.freq = avctx->sample_rate; + if (wanted_spec.freq <= 0 || wanted_spec.channels <= 0) { + fprintf(stderr, "Invalid sample rate or channel count!\n"); return -1; } - wanted_spec.freq = avctx->sample_rate; - wanted_spec.channels = avctx->channels; } if (!codec || @@ -2212,7 +2233,21 @@ static int stream_component_open(VideoState *is, int stream_index) return -1; } is->audio_hw_buf_size = spec.size; - is->audio_src_fmt= AV_SAMPLE_FMT_S16; + if (spec.format != AUDIO_S16SYS) { + fprintf(stderr, "SDL advised audio format %d is not supported!\n", spec.format); + return -1; + } + if (spec.channels != wanted_spec.channels) { + wanted_channel_layout = av_get_default_channel_layout(spec.channels); + if (!wanted_channel_layout) { + fprintf(stderr, "SDL advised channel count %d is not supported!\n", spec.channels); + return -1; + } + } + is->audio_src_fmt = is->audio_tgt_fmt = AV_SAMPLE_FMT_S16; + is->audio_src_freq = is->audio_tgt_freq = spec.freq; + is->audio_src_channel_layout = is->audio_tgt_channel_layout = wanted_channel_layout; + is->audio_src_channels = is->audio_tgt_channels = spec.channels; } ic->streams[stream_index]->discard = AVDISCARD_DEFAULT; @@ -2270,9 +2305,8 @@ 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); - is->reformat_ctx = NULL; + if (is->swr_ctx) + swr_free(&is->swr_ctx); break; case AVMEDIA_TYPE_VIDEO: packet_queue_abort(&is->videoq);