xenia-project/FFmpeg
1
0
Fork 0
mirror of https://github.com/xenia-project/FFmpeg.git synced 2024-11-24 12:09:55 +00:00
Code Issues Actions Packages Projects Releases Wiki Activity

lavfi: consistently use int for sample_rate in AVFilterLink and AVFilterBufferRefAudioProps

Browse Source 
Also consistent with AVCodecContext.sample_rate. Simplify/avoid
pointless type checks and conversions.

Breaks audio API/ABI.
This commit is contained in:
Stefano Sabatini 2011-08-21 19:17:24 +02:00
parent 2c0317419b
commit 4381bddc9f
5 changed files with 12 additions and 8 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

4
libavfilter/asrc_abuffer.c
View File

@ -125,9 +125,9 @@ static inline void log_input_change(void *ctx, AVFilterLink *link, AVFilterBuffe
-1, ref->audio->channel_layout);
av_log(ctx, AV_LOG_INFO,
"Audio input format changed: "
"%s:%s:%"PRId64" -> %s:%s:%u, normalizing\n",
"%s:%s:%d -> %s:%s:%d, normalizing\n",
av_get_sample_fmt_name(link->format),
old_layout_str, link->sample_rate,
old_layout_str, (int)link->sample_rate,
av_get_sample_fmt_name(ref->format),
new_layout_str, ref->audio->sample_rate);
}

4
libavfilter/avfilter.c
View File

@ -374,8 +374,8 @@ static void ff_dlog_link(void *ctx, AVFilterLink *link, int end)
av_get_channel_layout_string(buf, sizeof(buf), -1, link->channel_layout);
av_dlog(ctx,
"link[%p r:%"PRId64" cl:%s fmt:%-16s %-16s->%-16s]%s",
link, link->sample_rate, buf,
"link[%p r:%d cl:%s fmt:%-16s %-16s->%-16s]%s",
link, (int)link->sample_rate, buf,
av_get_sample_fmt_name(link->format),
link->src ? link->src->filter->name : "",
link->dst ? link->dst->filter->name : "",

6
libavfilter/avfilter.h
View File

@ -106,7 +106,7 @@ typedef struct AVFilterBuffer {
typedef struct AVFilterBufferRefAudioProps {
int64_t channel_layout; ///< channel layout of audio buffer
int nb_samples; ///< number of audio samples per channel
uint32_t sample_rate; ///< audio buffer sample rate
int sample_rate; ///< audio buffer sample rate
int planar; ///< audio buffer - planar or packed
} AVFilterBufferRefAudioProps;
@ -623,7 +623,11 @@ struct AVFilterLink {
AVRational sample_aspect_ratio; ///< agreed upon sample aspect ratio
/* These parameters apply only to audio */
int64_t channel_layout; ///< channel layout of current buffer (see libavutil/audioconvert.h)
#if LIBAVFILTER_VERSION_MAJOR < 3
int64_t sample_rate; ///< samples per second
#else
int sample_rate; ///< samples per second
#endif
int planar; ///< agreed upon packing mode of audio buffers. true if planar.
int format; ///< agreed upon media format

4
libavfilter/formats.c
View File

@ -266,11 +266,11 @@ int ff_parse_sample_format(int *ret, const char *arg, void *log_ctx)
return 0;
}
int ff_parse_sample_rate(unsigned *ret, const char *arg, void *log_ctx)
int ff_parse_sample_rate(int *ret, const char *arg, void *log_ctx)
{
char *tail;
double srate = av_strtod(arg, &tail);
if (*tail || srate < 1 || (int)srate != srate) {
if (*tail || srate < 1 || (int)srate != srate || srate > INT_MAX) {
av_log(log_ctx, AV_LOG_ERROR, "Invalid sample rate '%s'\n", arg);
return AVERROR(EINVAL);
}

2
libavfilter/internal.h
View File

@ -88,7 +88,7 @@ int ff_parse_pixel_format(enum PixelFormat *ret, const char *arg, void *log_ctx)
* @param log_ctx log context
* @return 0 in case of success, a negative AVERROR code on error
*/
int ff_parse_sample_rate(unsigned *ret, const char *arg, void *log_ctx);
int ff_parse_sample_rate(int *ret, const char *arg, void *log_ctx);
/**
* Parse a sample format name or a corresponding integer representation.