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rtpenc: Remove an av_abort() that depends on user-supplied data

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Signed-off-by: Martin Storsjö <martin@martin.st>
This commit is contained in:
Martin Storsjö 2012-08-08 23:23:28 +03:00
parent 7ca14c731e
commit bfb82fcddf
1 changed files with 9 additions and 12 deletions
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21
libavformat/rtpenc.c
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@ -281,8 +281,8 @@ 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_bits)
static int rtp_send_samples(AVFormatContext *s1,
const uint8_t *buf1, int size, int sample_size_bits)
{
RTPMuxContext *s = s1->priv_data;
int len, max_packet_size, n;
@ -292,7 +292,7 @@ static void rtp_send_samples(AVFormatContext *s1,
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();
return AVERROR(EINVAL);
n = 0;
while (size > 0) {
s->buf_ptr = s->buf;
@ -307,6 +307,7 @@ static void rtp_send_samples(AVFormatContext *s1,
ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0);
n += (s->buf_ptr - s->buf);
}
return 0;
}
static void rtp_send_mpegaudio(AVFormatContext *s1,
@ -461,25 +462,21 @@ static int rtp_write_packet(AVFormatContext *s1, AVPacket *pkt)
case AV_CODEC_ID_PCM_ALAW:
case AV_CODEC_ID_PCM_U8:
case AV_CODEC_ID_PCM_S8:
rtp_send_samples(s1, pkt->data, size, 8 * st->codec->channels);
break;
return rtp_send_samples(s1, pkt->data, size, 8 * st->codec->channels);
case AV_CODEC_ID_PCM_U16BE:
case AV_CODEC_ID_PCM_U16LE:
case AV_CODEC_ID_PCM_S16BE:
case AV_CODEC_ID_PCM_S16LE:
rtp_send_samples(s1, pkt->data, size, 16 * st->codec->channels);
break;
return rtp_send_samples(s1, pkt->data, size, 16 * st->codec->channels);
case AV_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_bits is 8 bits per stream
* clock. */
rtp_send_samples(s1, pkt->data, size, 8 * st->codec->channels);
break;
return rtp_send_samples(s1, pkt->data, size, 8 * st->codec->channels);
case AV_CODEC_ID_ADPCM_G726:
rtp_send_samples(s1, pkt->data, size,
st->codec->bits_per_coded_sample * st->codec->channels);
break;
return rtp_send_samples(s1, pkt->data, size,
st->codec->bits_per_coded_sample * st->codec->channels);
case AV_CODEC_ID_MP2:
case AV_CODEC_ID_MP3:
rtp_send_mpegaudio(s1, pkt->data, size);