diff options
author | Martin Storsjö <martin@martin.st> | 2012-08-09 00:23:28 +0400 |
---|---|---|
committer | Martin Storsjö <martin@martin.st> | 2012-08-09 01:24:41 +0400 |
commit | bfb82fcddfbebd6f9ae97165e3ccd86c5386be1b (patch) | |
tree | 44d7ba7302696f8601e2e721b3dd613435b400b6 /libavformat | |
parent | 7ca14c731ed1fca7072c335095c2d96a6cb80fc0 (diff) |
rtpenc: Remove an av_abort() that depends on user-supplied data
Signed-off-by: Martin Storsjö <martin@martin.st>
Diffstat (limited to 'libavformat')
-rw-r--r-- | libavformat/rtpenc.c | 21 |
1 files changed, 9 insertions, 12 deletions
diff --git a/libavformat/rtpenc.c b/libavformat/rtpenc.c index ada5a3b4f3..3f2231ffef 100644 --- a/libavformat/rtpenc.c +++ b/libavformat/rtpenc.c @@ -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); |