ffplay: fix indentation after last commit
Signed-off-by: Marton Balint <cus@passwd.hu>
This commit is contained in:
Marton Balint
parent
631ac655c0
commit
cc4741888d
182
ffplay.c
182
ffplay.c
@ -2424,105 +2424,101 @@ static int audio_decode_frame(VideoState *is)
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int wanted_nb_samples;
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Frame *af;
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{
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if (is->paused)
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if (is->paused)
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return -1;
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do {
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if (!(af = frame_queue_peek_readable(&is->sampq)))
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return -1;
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frame_queue_next(&is->sampq);
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} while (af->serial != is->audioq.serial);
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do {
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if (!(af = frame_queue_peek_readable(&is->sampq)))
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return -1;
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frame_queue_next(&is->sampq);
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} while (af->serial != is->audioq.serial);
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data_size = av_samples_get_buffer_size(NULL, av_frame_get_channels(af->frame),
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af->frame->nb_samples,
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af->frame->format, 1);
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{
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data_size = av_samples_get_buffer_size(NULL, av_frame_get_channels(af->frame),
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af->frame->nb_samples,
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af->frame->format, 1);
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dec_channel_layout =
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(af->frame->channel_layout && av_frame_get_channels(af->frame) == av_get_channel_layout_nb_channels(af->frame->channel_layout)) ?
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af->frame->channel_layout : av_get_default_channel_layout(av_frame_get_channels(af->frame));
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wanted_nb_samples = synchronize_audio(is, af->frame->nb_samples);
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dec_channel_layout =
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(af->frame->channel_layout && av_frame_get_channels(af->frame) == av_get_channel_layout_nb_channels(af->frame->channel_layout)) ?
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af->frame->channel_layout : av_get_default_channel_layout(av_frame_get_channels(af->frame));
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wanted_nb_samples = synchronize_audio(is, af->frame->nb_samples);
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if (af->frame->format != is->audio_src.fmt ||
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dec_channel_layout != is->audio_src.channel_layout ||
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af->frame->sample_rate != is->audio_src.freq ||
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(wanted_nb_samples != af->frame->nb_samples && !is->swr_ctx)) {
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swr_free(&is->swr_ctx);
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is->swr_ctx = swr_alloc_set_opts(NULL,
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is->audio_tgt.channel_layout, is->audio_tgt.fmt, is->audio_tgt.freq,
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dec_channel_layout, af->frame->format, af->frame->sample_rate,
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0, NULL);
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if (!is->swr_ctx || swr_init(is->swr_ctx) < 0) {
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av_log(NULL, AV_LOG_ERROR,
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"Cannot create sample rate converter for conversion of %d Hz %s %d channels to %d Hz %s %d channels!\n",
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af->frame->sample_rate, av_get_sample_fmt_name(af->frame->format), av_frame_get_channels(af->frame),
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is->audio_tgt.freq, av_get_sample_fmt_name(is->audio_tgt.fmt), is->audio_tgt.channels);
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swr_free(&is->swr_ctx);
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return -1;
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}
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is->audio_src.channel_layout = dec_channel_layout;
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is->audio_src.channels = av_frame_get_channels(af->frame);
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is->audio_src.freq = af->frame->sample_rate;
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is->audio_src.fmt = af->frame->format;
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}
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if (is->swr_ctx) {
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const uint8_t **in = (const uint8_t **)af->frame->extended_data;
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uint8_t **out = &is->audio_buf1;
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int out_count = (int64_t)wanted_nb_samples * is->audio_tgt.freq / af->frame->sample_rate + 256;
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int out_size = av_samples_get_buffer_size(NULL, is->audio_tgt.channels, out_count, is->audio_tgt.fmt, 0);
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int len2;
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if (out_size < 0) {
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av_log(NULL, AV_LOG_ERROR, "av_samples_get_buffer_size() failed\n");
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return -1;
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}
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if (wanted_nb_samples != af->frame->nb_samples) {
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if (swr_set_compensation(is->swr_ctx, (wanted_nb_samples - af->frame->nb_samples) * is->audio_tgt.freq / af->frame->sample_rate,
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wanted_nb_samples * is->audio_tgt.freq / af->frame->sample_rate) < 0) {
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av_log(NULL, AV_LOG_ERROR, "swr_set_compensation() failed\n");
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return -1;
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}
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}
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av_fast_malloc(&is->audio_buf1, &is->audio_buf1_size, out_size);
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if (!is->audio_buf1)
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return AVERROR(ENOMEM);
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len2 = swr_convert(is->swr_ctx, out, out_count, in, af->frame->nb_samples);
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if (len2 < 0) {
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av_log(NULL, AV_LOG_ERROR, "swr_convert() failed\n");
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return -1;
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}
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if (len2 == out_count) {
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av_log(NULL, AV_LOG_WARNING, "audio buffer is probably too small\n");
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if (swr_init(is->swr_ctx) < 0)
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swr_free(&is->swr_ctx);
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}
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is->audio_buf = is->audio_buf1;
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resampled_data_size = len2 * is->audio_tgt.channels * av_get_bytes_per_sample(is->audio_tgt.fmt);
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} else {
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is->audio_buf = af->frame->data[0];
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resampled_data_size = data_size;
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}
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audio_clock0 = is->audio_clock;
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/* update the audio clock with the pts */
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if (!isnan(af->pts))
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is->audio_clock = af->pts + (double) af->frame->nb_samples / af->frame->sample_rate;
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else
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is->audio_clock = NAN;
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is->audio_clock_serial = af->serial;
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#ifdef DEBUG
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{
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static double last_clock;
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printf("audio: delay=%0.3f clock=%0.3f clock0=%0.3f\n",
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is->audio_clock - last_clock,
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is->audio_clock, audio_clock0);
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last_clock = is->audio_clock;
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}
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#endif
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return resampled_data_size;
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if (af->frame->format != is->audio_src.fmt ||
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dec_channel_layout != is->audio_src.channel_layout ||
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af->frame->sample_rate != is->audio_src.freq ||
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(wanted_nb_samples != af->frame->nb_samples && !is->swr_ctx)) {
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swr_free(&is->swr_ctx);
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is->swr_ctx = swr_alloc_set_opts(NULL,
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is->audio_tgt.channel_layout, is->audio_tgt.fmt, is->audio_tgt.freq,
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dec_channel_layout, af->frame->format, af->frame->sample_rate,
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0, NULL);
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if (!is->swr_ctx || swr_init(is->swr_ctx) < 0) {
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av_log(NULL, AV_LOG_ERROR,
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"Cannot create sample rate converter for conversion of %d Hz %s %d channels to %d Hz %s %d channels!\n",
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af->frame->sample_rate, av_get_sample_fmt_name(af->frame->format), av_frame_get_channels(af->frame),
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is->audio_tgt.freq, av_get_sample_fmt_name(is->audio_tgt.fmt), is->audio_tgt.channels);
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swr_free(&is->swr_ctx);
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return -1;
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}
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is->audio_src.channel_layout = dec_channel_layout;
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is->audio_src.channels = av_frame_get_channels(af->frame);
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is->audio_src.freq = af->frame->sample_rate;
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is->audio_src.fmt = af->frame->format;
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}
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if (is->swr_ctx) {
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const uint8_t **in = (const uint8_t **)af->frame->extended_data;
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uint8_t **out = &is->audio_buf1;
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int out_count = (int64_t)wanted_nb_samples * is->audio_tgt.freq / af->frame->sample_rate + 256;
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int out_size = av_samples_get_buffer_size(NULL, is->audio_tgt.channels, out_count, is->audio_tgt.fmt, 0);
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int len2;
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if (out_size < 0) {
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av_log(NULL, AV_LOG_ERROR, "av_samples_get_buffer_size() failed\n");
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return -1;
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}
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if (wanted_nb_samples != af->frame->nb_samples) {
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if (swr_set_compensation(is->swr_ctx, (wanted_nb_samples - af->frame->nb_samples) * is->audio_tgt.freq / af->frame->sample_rate,
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wanted_nb_samples * is->audio_tgt.freq / af->frame->sample_rate) < 0) {
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av_log(NULL, AV_LOG_ERROR, "swr_set_compensation() failed\n");
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return -1;
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}
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}
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av_fast_malloc(&is->audio_buf1, &is->audio_buf1_size, out_size);
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if (!is->audio_buf1)
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return AVERROR(ENOMEM);
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len2 = swr_convert(is->swr_ctx, out, out_count, in, af->frame->nb_samples);
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if (len2 < 0) {
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av_log(NULL, AV_LOG_ERROR, "swr_convert() failed\n");
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return -1;
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}
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if (len2 == out_count) {
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av_log(NULL, AV_LOG_WARNING, "audio buffer is probably too small\n");
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if (swr_init(is->swr_ctx) < 0)
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swr_free(&is->swr_ctx);
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}
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is->audio_buf = is->audio_buf1;
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resampled_data_size = len2 * is->audio_tgt.channels * av_get_bytes_per_sample(is->audio_tgt.fmt);
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} else {
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is->audio_buf = af->frame->data[0];
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resampled_data_size = data_size;
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}
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audio_clock0 = is->audio_clock;
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/* update the audio clock with the pts */
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if (!isnan(af->pts))
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is->audio_clock = af->pts + (double) af->frame->nb_samples / af->frame->sample_rate;
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else
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is->audio_clock = NAN;
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is->audio_clock_serial = af->serial;
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#ifdef DEBUG
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{
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static double last_clock;
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printf("audio: delay=%0.3f clock=%0.3f clock0=%0.3f\n",
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is->audio_clock - last_clock,
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is->audio_clock, audio_clock0);
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last_clock = is->audio_clock;
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}
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#endif
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return resampled_data_size;
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}
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/* prepare a new audio buffer */
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