diff options
| author | Martin Storsjö <martin@martin.st> | 2011-11-30 23:10:54 +0200 |
|---|---|---|
| committer | Martin Storsjö <martin@martin.st> | 2011-12-01 23:19:22 +0200 |
| commit | 77e0c7584b595edcec7bf393c0e77dbcfe2a8cb4 (patch) | |
| tree | 4cbc884895f48c7fe55f5c3cd85b24370dfa8c9b /libavformat/rtpenc.c | |
| parent | 2d31d890bfce103512dca34e35815762eb61b5da (diff) | |
| download | ffmpeg-77e0c7584b595edcec7bf393c0e77dbcfe2a8cb4.tar.gz | |
rtpenc: Change rtp_send_samples to handle sample sizes other than even bytes
Signed-off-by: Martin Storsjö <martin@martin.st>
Diffstat (limited to 'libavformat/rtpenc.c')
| -rw-r--r-- | libavformat/rtpenc.c | 21 |
1 files changed, 12 insertions, 9 deletions
diff --git a/libavformat/rtpenc.c b/libavformat/rtpenc.c index 88b85b995c..7434837a02 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: |