diff options
author | Michael Niedermayer <michaelni@gmx.at> | 2012-08-09 21:09:39 +0400 |
---|---|---|
committer | Michael Niedermayer <michaelni@gmx.at> | 2012-08-09 21:31:56 +0400 |
commit | 9f088a1ed4d34f0cf4244a4b80960af9e8f23dfc (patch) | |
tree | 049c8db5ee42c462bfe3d999146a224cff29bf03 /libavformat/rtpenc.c | |
parent | e1a983e6010930ab742bede275de1ccf921485b7 (diff) | |
parent | f69f4036f8cc3b673864dce01d2714fd5e49e8da (diff) |
Merge remote-tracking branch 'qatar/master'
* qatar/master:
mpegvideo: reduce excessive inlining of mpeg_motion()
mpegvideo: convert mpegvideo_common.h to a .c file
build: factor out mpegvideo.o dependencies to CONFIG_MPEGVIDEO
Move MASK_ABS macro to libavcodec/mathops.h
x86: move MANGLE() and related macros to libavutil/x86/asm.h
x86: rename libavutil/x86_cpu.h to libavutil/x86/asm.h
aacdec: Don't fall back to the old output configuration when no old configuration is present.
rtmp: Add message tracking
rtsp: Support mpegts in raw udp packets
rtsp: Support receiving plain data over UDP without any RTP encapsulation
rtpdec: Remove an unused include
rtpenc: Remove an av_abort() that depends on user-supplied data
vsrc_movie: discourage its use with avconv.
avconv: allow no input files.
avconv: prevent invalid reads in transcode_init()
avconv: rename OutputStream.is_past_recording_time to finished.
Conflicts:
configure
doc/filters.texi
ffmpeg.c
ffmpeg.h
libavcodec/Makefile
libavcodec/aacdec.c
libavcodec/mpegvideo.c
libavformat/version.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Diffstat (limited to 'libavformat/rtpenc.c')
-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 6af23f760a..7cd7034fbf 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); |