[FFmpeg-cvslog] libavutil: add utility functions to simplify allocation of audio buffers.

Justin Ruggles git at videolan.org
Thu Nov 24 03:38:34 CET 2011


ffmpeg | branch: master | Justin Ruggles <justin.ruggles at gmail.com> | Sat Nov 12 15:43:43 2011 -0500| [bbb46f3ec7128d8a624f2aa5b4f99ec44c0b9567] | committer: Justin Ruggles

libavutil: add utility functions to simplify allocation of audio buffers.

Based on code by Stefano Sabatini.

> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=bbb46f3ec7128d8a624f2aa5b4f99ec44c0b9567
---

 doc/APIchanges        |    4 +++
 libavutil/avutil.h    |    2 +-
 libavutil/samplefmt.c |   65 +++++++++++++++++++++++++++++++++++++++++++++++++
 libavutil/samplefmt.h |   53 ++++++++++++++++++++++++++++++++++++++++
 4 files changed, 123 insertions(+), 1 deletions(-)

diff --git a/doc/APIchanges b/doc/APIchanges
index 0e5fd68..8efee0c 100644
--- a/doc/APIchanges
+++ b/doc/APIchanges
@@ -13,6 +13,10 @@ libavutil:   2011-04-18
 
 API changes, most recent first:
 
+2011-xx-xx - xxxxxxx - lavu 51.18.0
+  Add av_samples_get_buffer_size(), av_samples_fill_arrays(), and
+  av_samples_alloc(), to samplefmt.h.
+
 2011-xx-xx - xxxxxxx - lavu 51.17.0
   Add planar sample formats and av_sample_fmt_is_planar() to samplefmt.h.
 
diff --git a/libavutil/avutil.h b/libavutil/avutil.h
index 0c256ca..a93a079 100644
--- a/libavutil/avutil.h
+++ b/libavutil/avutil.h
@@ -153,7 +153,7 @@
  */
 
 #define LIBAVUTIL_VERSION_MAJOR 51
-#define LIBAVUTIL_VERSION_MINOR 17
+#define LIBAVUTIL_VERSION_MINOR 18
 #define LIBAVUTIL_VERSION_MICRO  0
 
 #define LIBAVUTIL_VERSION_INT   AV_VERSION_INT(LIBAVUTIL_VERSION_MAJOR, \
diff --git a/libavutil/samplefmt.c b/libavutil/samplefmt.c
index 6fc3b7e..f38d05e 100644
--- a/libavutil/samplefmt.c
+++ b/libavutil/samplefmt.c
@@ -92,3 +92,68 @@ int av_sample_fmt_is_planar(enum AVSampleFormat sample_fmt)
          return 0;
      return sample_fmt_info[sample_fmt].planar;
 }
+
+int av_samples_get_buffer_size(int *linesize, int nb_channels, int nb_samples,
+                               enum AVSampleFormat sample_fmt, int align)
+{
+    int line_size;
+    int sample_size = av_get_bytes_per_sample(sample_fmt);
+    int planar      = av_sample_fmt_is_planar(sample_fmt);
+
+    /* validate parameter ranges */
+    if (!sample_size || nb_samples <= 0 || nb_channels <= 0)
+        return AVERROR(EINVAL);
+
+    /* check for integer overflow */
+    if (nb_channels > INT_MAX / align ||
+        (int64_t)nb_channels * nb_samples > (INT_MAX - (align * nb_channels)) / sample_size)
+        return AVERROR(EINVAL);
+
+    line_size = planar ? FFALIGN(nb_samples * sample_size,               align) :
+                         FFALIGN(nb_samples * sample_size * nb_channels, align);
+    if (linesize)
+        *linesize = line_size;
+
+    return planar ? line_size * nb_channels : line_size;
+}
+
+int av_samples_fill_arrays(uint8_t **audio_data, int *linesize,
+                           uint8_t *buf, int nb_channels, int nb_samples,
+                           enum AVSampleFormat sample_fmt, int align)
+{
+    int ch, planar, buf_size;
+
+    planar   = av_sample_fmt_is_planar(sample_fmt);
+    buf_size = av_samples_get_buffer_size(linesize, nb_channels, nb_samples,
+                                          sample_fmt, align);
+    if (buf_size < 0)
+        return buf_size;
+
+    audio_data[0] = buf;
+    for (ch = 1; planar && ch < nb_channels; ch++)
+        audio_data[ch] = audio_data[ch-1] + *linesize;
+
+    return 0;
+}
+
+int av_samples_alloc(uint8_t **audio_data, int *linesize, int nb_channels,
+                     int nb_samples, enum AVSampleFormat sample_fmt, int align)
+{
+    uint8_t *buf;
+    int size = av_samples_get_buffer_size(NULL, nb_channels, nb_samples,
+                                          sample_fmt, align);
+    if (size < 0)
+        return size;
+
+    buf = av_mallocz(size);
+    if (!buf)
+        return AVERROR(ENOMEM);
+
+    size = av_samples_fill_arrays(audio_data, linesize, buf, nb_channels,
+                                  nb_samples, sample_fmt, align);
+    if (size < 0) {
+        av_free(buf);
+        return size;
+    }
+    return 0;
+}
diff --git a/libavutil/samplefmt.h b/libavutil/samplefmt.h
index ce7ffc7..b671556 100644
--- a/libavutil/samplefmt.h
+++ b/libavutil/samplefmt.h
@@ -92,4 +92,57 @@ int av_get_bytes_per_sample(enum AVSampleFormat sample_fmt);
  */
 int av_sample_fmt_is_planar(enum AVSampleFormat sample_fmt);
 
+/**
+ * Get the required buffer size for the given audio parameters.
+ *
+ * @param[out] linesize calculated linesize, may be NULL
+ * @param nb_channels   the number of channels
+ * @param nb_samples    the number of samples in a single channel
+ * @param sample_fmt    the sample format
+ * @return              required buffer size, or negative error code on failure
+ */
+int av_samples_get_buffer_size(int *linesize, int nb_channels, int nb_samples,
+                               enum AVSampleFormat sample_fmt, int align);
+
+/**
+ * Fill channel data pointers and linesize for samples with sample
+ * format sample_fmt.
+ *
+ * The pointers array is filled with the pointers to the samples data:
+ * for planar, set the start point of each channel's data within the buffer,
+ * for packed, set the start point of the entire buffer only.
+ *
+ * The linesize array is filled with the aligned size of each channel's data
+ * buffer for planar layout, or the aligned size of the buffer for all channels
+ * for packed layout.
+ *
+ * @param[out] audio_data  array to be filled with the pointer for each channel
+ * @param[out] linesize    calculated linesize
+ * @param buf              the pointer to a buffer containing the samples
+ * @param nb_channels      the number of channels
+ * @param nb_samples       the number of samples in a single channel
+ * @param sample_fmt       the sample format
+ * @param align            buffer size alignment (1 = no alignment required)
+ * @return                 0 on success or a negative error code on failure
+ */
+int av_samples_fill_arrays(uint8_t **audio_data, int *linesize, uint8_t *buf,
+                           int nb_channels, int nb_samples,
+                           enum AVSampleFormat sample_fmt, int align);
+
+/**
+ * Allocate a samples buffer for nb_samples samples, and fill data pointers and
+ * linesize accordingly.
+ * The allocated samples buffer can be freed by using av_freep(&audio_data[0])
+ *
+ * @param[out] audio_data  array to be filled with the pointer for each channel
+ * @param[out] linesize    aligned size for audio buffer(s)
+ * @param nb_channels      number of audio channels
+ * @param nb_samples       number of samples per channel
+ * @param align            buffer size alignment (1 = no alignment required)
+ * @return                 0 on success or a negative error code on failure
+ * @see av_samples_fill_arrays()
+ */
+int av_samples_alloc(uint8_t **audio_data, int *linesize, int nb_channels,
+                     int nb_samples, enum AVSampleFormat sample_fmt, int align);
+
 #endif /* AVUTIL_SAMPLEFMT_H */



More information about the ffmpeg-cvslog mailing list