[FFmpeg-devel] [PATCH 2/3] ffplay: fix indentation after last commit
Marton Balint
cus at passwd.hu
Thu Oct 30 00:31:26 CET 2014
Signed-off-by: Marton Balint <cus at passwd.hu>
---
ffplay.c | 182 +++++++++++++++++++++++++++++++--------------------------------
1 file changed, 89 insertions(+), 93 deletions(-)
diff --git a/ffplay.c b/ffplay.c
index 24bcae2..a3b34fd 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 */
--
1.8.4.5
More information about the ffmpeg-devel
mailing list