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rtpenc: Change rtp_send_samples to handle sample sizes other than even bytes

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Signed-off-by: Martin Storsjö <martin@martin.st>
This commit is contained in:
Martin Storsjö 2011-11-30 23:10:54 +02:00
parent 2d31d890bf
commit 77e0c7584b
1 changed files with 12 additions and 9 deletions
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21
libavformat/rtpenc.c
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@ -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: