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
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