From cc4741888d7fa74074a295158fcbb88475a08a88 Mon Sep 17 00:00:00 2001 From: Marton Balint Date: Fri, 24 Oct 2014 15:16:41 +0200 Subject: ffplay: fix indentation after last commit Signed-off-by: Marton Balint --- ffplay.c | 182 +++++++++++++++++++++++++++++++-------------------------------- 1 file changed, 89 insertions(+), 93 deletions(-) (limited to 'ffplay.c') diff --git a/ffplay.c b/ffplay.c index 24bcae2a6b..a3b34fd966 100644 --- a/ffplay.c +++ b/ffplay.c @@ -2424,105 +2424,101 @@ static int audio_decode_frame(VideoState *is) int wanted_nb_samples; Frame *af; - { - if (is->paused) - return -1; + if (is->paused) + return -1; - do { - if (!(af = frame_queue_peek_readable(&is->sampq))) + do { + if (!(af = frame_queue_peek_readable(&is->sampq))) + return -1; + frame_queue_next(&is->sampq); + } while (af->serial != is->audioq.serial); + + data_size = av_samples_get_buffer_size(NULL, av_frame_get_channels(af->frame), + af->frame->nb_samples, + af->frame->format, 1); + + dec_channel_layout = + (af->frame->channel_layout && av_frame_get_channels(af->frame) == av_get_channel_layout_nb_channels(af->frame->channel_layout)) ? + af->frame->channel_layout : av_get_default_channel_layout(av_frame_get_channels(af->frame)); + wanted_nb_samples = synchronize_audio(is, af->frame->nb_samples); + + if (af->frame->format != is->audio_src.fmt || + dec_channel_layout != is->audio_src.channel_layout || + af->frame->sample_rate != is->audio_src.freq || + (wanted_nb_samples != af->frame->nb_samples && !is->swr_ctx)) { + swr_free(&is->swr_ctx); + is->swr_ctx = swr_alloc_set_opts(NULL, + is->audio_tgt.channel_layout, is->audio_tgt.fmt, is->audio_tgt.freq, + dec_channel_layout, af->frame->format, af->frame->sample_rate, + 0, NULL); + if (!is->swr_ctx || swr_init(is->swr_ctx) < 0) { + av_log(NULL, AV_LOG_ERROR, + "Cannot create sample rate converter for conversion of %d Hz %s %d channels to %d Hz %s %d channels!\n", + af->frame->sample_rate, av_get_sample_fmt_name(af->frame->format), av_frame_get_channels(af->frame), + is->audio_tgt.freq, av_get_sample_fmt_name(is->audio_tgt.fmt), is->audio_tgt.channels); + swr_free(&is->swr_ctx); + return -1; + } + is->audio_src.channel_layout = dec_channel_layout; + is->audio_src.channels = av_frame_get_channels(af->frame); + is->audio_src.freq = af->frame->sample_rate; + is->audio_src.fmt = af->frame->format; + } + + if (is->swr_ctx) { + const uint8_t **in = (const uint8_t **)af->frame->extended_data; + uint8_t **out = &is->audio_buf1; + int out_count = (int64_t)wanted_nb_samples * is->audio_tgt.freq / af->frame->sample_rate + 256; + int out_size = av_samples_get_buffer_size(NULL, is->audio_tgt.channels, out_count, is->audio_tgt.fmt, 0); + int len2; + if (out_size < 0) { + av_log(NULL, AV_LOG_ERROR, "av_samples_get_buffer_size() failed\n"); + return -1; + } + if (wanted_nb_samples != af->frame->nb_samples) { + if (swr_set_compensation(is->swr_ctx, (wanted_nb_samples - af->frame->nb_samples) * is->audio_tgt.freq / af->frame->sample_rate, + wanted_nb_samples * is->audio_tgt.freq / af->frame->sample_rate) < 0) { + av_log(NULL, AV_LOG_ERROR, "swr_set_compensation() failed\n"); return -1; - frame_queue_next(&is->sampq); - } while (af->serial != is->audioq.serial); - - { - data_size = av_samples_get_buffer_size(NULL, av_frame_get_channels(af->frame), - af->frame->nb_samples, - af->frame->format, 1); - - dec_channel_layout = - (af->frame->channel_layout && av_frame_get_channels(af->frame) == av_get_channel_layout_nb_channels(af->frame->channel_layout)) ? - af->frame->channel_layout : av_get_default_channel_layout(av_frame_get_channels(af->frame)); - wanted_nb_samples = synchronize_audio(is, af->frame->nb_samples); - - if (af->frame->format != is->audio_src.fmt || - dec_channel_layout != is->audio_src.channel_layout || - af->frame->sample_rate != is->audio_src.freq || - (wanted_nb_samples != af->frame->nb_samples && !is->swr_ctx)) { - swr_free(&is->swr_ctx); - is->swr_ctx = swr_alloc_set_opts(NULL, - is->audio_tgt.channel_layout, is->audio_tgt.fmt, is->audio_tgt.freq, - dec_channel_layout, af->frame->format, af->frame->sample_rate, - 0, NULL); - if (!is->swr_ctx || swr_init(is->swr_ctx) < 0) { - av_log(NULL, AV_LOG_ERROR, - "Cannot create sample rate converter for conversion of %d Hz %s %d channels to %d Hz %s %d channels!\n", - af->frame->sample_rate, av_get_sample_fmt_name(af->frame->format), av_frame_get_channels(af->frame), - is->audio_tgt.freq, av_get_sample_fmt_name(is->audio_tgt.fmt), is->audio_tgt.channels); - swr_free(&is->swr_ctx); - return -1; - } - is->audio_src.channel_layout = dec_channel_layout; - is->audio_src.channels = av_frame_get_channels(af->frame); - is->audio_src.freq = af->frame->sample_rate; - is->audio_src.fmt = af->frame->format; - } - - if (is->swr_ctx) { - const uint8_t **in = (const uint8_t **)af->frame->extended_data; - uint8_t **out = &is->audio_buf1; - int out_count = (int64_t)wanted_nb_samples * is->audio_tgt.freq / af->frame->sample_rate + 256; - int out_size = av_samples_get_buffer_size(NULL, is->audio_tgt.channels, out_count, is->audio_tgt.fmt, 0); - int len2; - if (out_size < 0) { - av_log(NULL, AV_LOG_ERROR, "av_samples_get_buffer_size() failed\n"); - return -1; - } - if (wanted_nb_samples != af->frame->nb_samples) { - if (swr_set_compensation(is->swr_ctx, (wanted_nb_samples - af->frame->nb_samples) * is->audio_tgt.freq / af->frame->sample_rate, - wanted_nb_samples * is->audio_tgt.freq / af->frame->sample_rate) < 0) { - av_log(NULL, AV_LOG_ERROR, "swr_set_compensation() failed\n"); - return -1; - } - } - av_fast_malloc(&is->audio_buf1, &is->audio_buf1_size, out_size); - if (!is->audio_buf1) - return AVERROR(ENOMEM); - len2 = swr_convert(is->swr_ctx, out, out_count, in, af->frame->nb_samples); - if (len2 < 0) { - av_log(NULL, AV_LOG_ERROR, "swr_convert() failed\n"); - return -1; - } - if (len2 == out_count) { - av_log(NULL, AV_LOG_WARNING, "audio buffer is probably too small\n"); - if (swr_init(is->swr_ctx) < 0) - swr_free(&is->swr_ctx); - } - is->audio_buf = is->audio_buf1; - resampled_data_size = len2 * is->audio_tgt.channels * av_get_bytes_per_sample(is->audio_tgt.fmt); - } else { - is->audio_buf = af->frame->data[0]; - resampled_data_size = data_size; } + } + av_fast_malloc(&is->audio_buf1, &is->audio_buf1_size, out_size); + if (!is->audio_buf1) + return AVERROR(ENOMEM); + len2 = swr_convert(is->swr_ctx, out, out_count, in, af->frame->nb_samples); + if (len2 < 0) { + av_log(NULL, AV_LOG_ERROR, "swr_convert() failed\n"); + return -1; + } + if (len2 == out_count) { + av_log(NULL, AV_LOG_WARNING, "audio buffer is probably too small\n"); + if (swr_init(is->swr_ctx) < 0) + swr_free(&is->swr_ctx); + } + is->audio_buf = is->audio_buf1; + resampled_data_size = len2 * is->audio_tgt.channels * av_get_bytes_per_sample(is->audio_tgt.fmt); + } else { + is->audio_buf = af->frame->data[0]; + resampled_data_size = data_size; + } - audio_clock0 = is->audio_clock; - /* update the audio clock with the pts */ - if (!isnan(af->pts)) - is->audio_clock = af->pts + (double) af->frame->nb_samples / af->frame->sample_rate; - else - is->audio_clock = NAN; - is->audio_clock_serial = af->serial; + audio_clock0 = is->audio_clock; + /* update the audio clock with the pts */ + if (!isnan(af->pts)) + is->audio_clock = af->pts + (double) af->frame->nb_samples / af->frame->sample_rate; + else + is->audio_clock = NAN; + is->audio_clock_serial = af->serial; #ifdef DEBUG - { - static double last_clock; - printf("audio: delay=%0.3f clock=%0.3f clock0=%0.3f\n", - is->audio_clock - last_clock, - is->audio_clock, audio_clock0); - last_clock = is->audio_clock; - } -#endif - return resampled_data_size; - } + { + static double last_clock; + printf("audio: delay=%0.3f clock=%0.3f clock0=%0.3f\n", + is->audio_clock - last_clock, + is->audio_clock, audio_clock0); + last_clock = is->audio_clock; } +#endif + return resampled_data_size; } /* prepare a new audio buffer */ -- cgit v1.2.3