[2/3] rtpenc: Change rtp_send_samples to handle sample sizes other than even bytes

Message ID 1322688143-57029-2-git-send-email-martin@martin.st
State Committed
Commit 77e0c7584b595edcec7bf393c0e77dbcfe2a8cb4
Headers show

Commit Message

Martin Storsjö Nov. 30, 2011, 9:22 p.m.
---
 libavformat/rtpenc.c |   21 ++++++++++++---------
 1 files changed, 12 insertions(+), 9 deletions(-)

Comments

Justin Ruggles Dec. 1, 2011, 3:33 p.m. | #1
On 11/30/2011 04:22 PM, Martin Storsjö wrote:

> ---
>  libavformat/rtpenc.c |   21 ++++++++++++---------
>  1 files changed, 12 insertions(+), 9 deletions(-)
> 
> diff --git a/libavformat/rtpenc.c b/libavformat/rtpenc.c
> index 88b85b9..7434837 100644
> --- a/libavformat/rtpenc.c
> +++ b/libavformat/rtpenc.c
> @@ -248,14 +248,16 @@ void ff_rtp_send_data(AVFormatContext *s1, const uint8_t *buf1, int len, int m)
>  /* send an integer number of samples and compute time stamp and fill
>     the rtp send buffer before sending. */
>  static void rtp_send_samples(AVFormatContext *s1,
> -                             const uint8_t *buf1, int size, int sample_size)
> +                             const uint8_t *buf1, int size, int sample_size_bits)
>  {
>      RTPMuxContext *s = s1->priv_data;
>      int len, max_packet_size, n;
> +    /* Calculate the number of bytes to get samples aligned on a byte border */
> +    int aligned_samples_size = sample_size_bits/av_gcd(sample_size_bits, 8);
>  
> -    max_packet_size = (s->max_payload_size / sample_size) * sample_size;
> -    /* not needed, but who nows */
> -    if ((size % sample_size) != 0)
> +    max_packet_size = (s->max_payload_size / aligned_samples_size) * aligned_samples_size;
> +    /* Not needed, but who knows. Don't check if samples aren't an even number of bytes. */
> +    if ((sample_size_bits % 8) == 0 && ((8 * size) % sample_size_bits) != 0)
>          av_abort();
>      n = 0;
>      while (size > 0) {
> @@ -267,7 +269,7 @@ static void rtp_send_samples(AVFormatContext *s1,
>          s->buf_ptr += len;
>          buf1 += len;
>          size -= len;
> -        s->timestamp = s->cur_timestamp + n / sample_size;
> +        s->timestamp = s->cur_timestamp + n * 8 / sample_size_bits;
>          ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0);
>          n += (s->buf_ptr - s->buf);
>      }
> @@ -394,19 +396,20 @@ static int rtp_write_packet(AVFormatContext *s1, AVPacket *pkt)
>      case CODEC_ID_PCM_ALAW:
>      case CODEC_ID_PCM_U8:
>      case CODEC_ID_PCM_S8:
> -        rtp_send_samples(s1, pkt->data, size, 1 * st->codec->channels);
> +        rtp_send_samples(s1, pkt->data, size, 8 * st->codec->channels);
>          break;
>      case CODEC_ID_PCM_U16BE:
>      case CODEC_ID_PCM_U16LE:
>      case CODEC_ID_PCM_S16BE:
>      case CODEC_ID_PCM_S16LE:
> -        rtp_send_samples(s1, pkt->data, size, 2 * st->codec->channels);
> +        rtp_send_samples(s1, pkt->data, size, 16 * st->codec->channels);
>          break;
>      case CODEC_ID_ADPCM_G722:
>          /* The actual sample size is half a byte per sample, but since the
>           * stream clock rate is 8000 Hz while the sample rate is 16000 Hz,
> -         * the correct parameter for send_samples is 1 byte per stream clock. */
> -        rtp_send_samples(s1, pkt->data, size, 1 * st->codec->channels);
> +         * the correct parameter for send_samples_bits is 8 bits per stream
> +         * clock. */
> +        rtp_send_samples(s1, pkt->data, size, 8 * st->codec->channels);
>          break;
>      case CODEC_ID_MP2:
>      case CODEC_ID_MP3:


patch looks fine.

-Justin

Patch

diff --git a/libavformat/rtpenc.c b/libavformat/rtpenc.c
index 88b85b9..7434837 100644
--- a/libavformat/rtpenc.c
+++ b/libavformat/rtpenc.c
@@ -248,14 +248,16 @@  void ff_rtp_send_data(AVFormatContext *s1, const uint8_t *buf1, int len, int m)
 /* send an integer number of samples and compute time stamp and fill
    the rtp send buffer before sending. */
 static void rtp_send_samples(AVFormatContext *s1,
-                             const uint8_t *buf1, int size, int sample_size)
+                             const uint8_t *buf1, int size, int sample_size_bits)
 {
     RTPMuxContext *s = s1->priv_data;
     int len, max_packet_size, n;
+    /* Calculate the number of bytes to get samples aligned on a byte border */
+    int aligned_samples_size = sample_size_bits/av_gcd(sample_size_bits, 8);
 
-    max_packet_size = (s->max_payload_size / sample_size) * sample_size;
-    /* not needed, but who nows */
-    if ((size % sample_size) != 0)
+    max_packet_size = (s->max_payload_size / aligned_samples_size) * aligned_samples_size;
+    /* Not needed, but who knows. Don't check if samples aren't an even number of bytes. */
+    if ((sample_size_bits % 8) == 0 && ((8 * size) % sample_size_bits) != 0)
         av_abort();
     n = 0;
     while (size > 0) {
@@ -267,7 +269,7 @@  static void rtp_send_samples(AVFormatContext *s1,
         s->buf_ptr += len;
         buf1 += len;
         size -= len;
-        s->timestamp = s->cur_timestamp + n / sample_size;
+        s->timestamp = s->cur_timestamp + n * 8 / sample_size_bits;
         ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0);
         n += (s->buf_ptr - s->buf);
     }
@@ -394,19 +396,20 @@  static int rtp_write_packet(AVFormatContext *s1, AVPacket *pkt)
     case CODEC_ID_PCM_ALAW:
     case CODEC_ID_PCM_U8:
     case CODEC_ID_PCM_S8:
-        rtp_send_samples(s1, pkt->data, size, 1 * st->codec->channels);
+        rtp_send_samples(s1, pkt->data, size, 8 * st->codec->channels);
         break;
     case CODEC_ID_PCM_U16BE:
     case CODEC_ID_PCM_U16LE:
     case CODEC_ID_PCM_S16BE:
     case CODEC_ID_PCM_S16LE:
-        rtp_send_samples(s1, pkt->data, size, 2 * st->codec->channels);
+        rtp_send_samples(s1, pkt->data, size, 16 * st->codec->channels);
         break;
     case CODEC_ID_ADPCM_G722:
         /* The actual sample size is half a byte per sample, but since the
          * stream clock rate is 8000 Hz while the sample rate is 16000 Hz,
-         * the correct parameter for send_samples is 1 byte per stream clock. */
-        rtp_send_samples(s1, pkt->data, size, 1 * st->codec->channels);
+         * the correct parameter for send_samples_bits is 8 bits per stream
+         * clock. */
+        rtp_send_samples(s1, pkt->data, size, 8 * st->codec->channels);
         break;
     case CODEC_ID_MP2:
     case CODEC_ID_MP3: