mirror of
https://github.com/nyanmisaka/ffmpeg-rockchip.git
synced 2025-12-24 11:52:06 +08:00
64 bit pts for writing - more const usage
Originally committed as revision 2255 to svn://svn.ffmpeg.org/ffmpeg/trunk
This commit is contained in:
@@ -425,7 +425,7 @@ static void rtcp_send_sr(AVFormatContext *s1, int64_t ntp_time)
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/* send an rtp packet. sequence number is incremented, but the caller
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must update the timestamp itself */
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static void rtp_send_data(AVFormatContext *s1, uint8_t *buf1, int len)
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static void rtp_send_data(AVFormatContext *s1, const uint8_t *buf1, int len)
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{
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RTPContext *s = s1->priv_data;
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@@ -451,7 +451,7 @@ static void rtp_send_data(AVFormatContext *s1, uint8_t *buf1, int len)
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/* send an integer number of samples and compute time stamp and fill
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the rtp send buffer before sending. */
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static void rtp_send_samples(AVFormatContext *s1,
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uint8_t *buf1, int size, int sample_size)
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const uint8_t *buf1, int size, int sample_size)
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{
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RTPContext *s = s1->priv_data;
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int len, max_packet_size, n;
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@@ -484,7 +484,7 @@ static void rtp_send_samples(AVFormatContext *s1,
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/* NOTE: we suppose that exactly one frame is given as argument here */
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/* XXX: test it */
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static void rtp_send_mpegaudio(AVFormatContext *s1,
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uint8_t *buf1, int size)
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const uint8_t *buf1, int size)
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{
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RTPContext *s = s1->priv_data;
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AVStream *st = s1->streams[0];
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@@ -540,7 +540,7 @@ static void rtp_send_mpegaudio(AVFormatContext *s1,
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/* NOTE: a single frame must be passed with sequence header if
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needed. XXX: use slices. */
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static void rtp_send_mpegvideo(AVFormatContext *s1,
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uint8_t *buf1, int size)
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const uint8_t *buf1, int size)
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{
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RTPContext *s = s1->priv_data;
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AVStream *st = s1->streams[0];
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@@ -587,7 +587,7 @@ static void rtp_send_mpegvideo(AVFormatContext *s1,
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}
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static void rtp_send_raw(AVFormatContext *s1,
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uint8_t *buf1, int size)
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const uint8_t *buf1, int size)
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{
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RTPContext *s = s1->priv_data;
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AVStream *st = s1->streams[0];
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@@ -613,7 +613,7 @@ static void rtp_send_raw(AVFormatContext *s1,
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/* write an RTP packet. 'buf1' must contain a single specific frame. */
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static int rtp_write_packet(AVFormatContext *s1, int stream_index,
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uint8_t *buf1, int size, int force_pts)
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const uint8_t *buf1, int size, int64_t pts)
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{
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RTPContext *s = s1->priv_data;
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AVStream *st = s1->streams[0];
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@@ -630,7 +630,7 @@ static int rtp_write_packet(AVFormatContext *s1, int stream_index,
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if (s->first_packet || rtcp_bytes >= 28) {
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/* compute NTP time */
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/* XXX: 90 kHz timestamp hardcoded */
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ntp_time = ((int64_t)force_pts << 28) / 5625;
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ntp_time = (pts << 28) / 5625;
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rtcp_send_sr(s1, ntp_time);
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s->last_octet_count = s->octet_count;
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s->first_packet = 0;
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