mirror of
https://github.com/ossrs/srs.git
synced 2025-02-15 04:42:04 +00:00
Merge branch 'srs.master'
This commit is contained in:
winlin
commit
7a4901cd54
4 changed files with 355 additions and 32 deletions
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@ -35,7 +35,7 @@ int main(int argc, char** argv)
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srs_rtmp_t rtmp;
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// time
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int64_t time_startup = srs_utils_get_time_ms();
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int64_t time_startup = srs_utils_time_ms();
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int64_t time_dns_resolve = 0;
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int64_t time_socket_connect = 0;
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int64_t time_play_stream = 0;
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@ -95,14 +95,14 @@ int main(int argc, char** argv)
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goto rtmp_destroy;
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}
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srs_human_trace("dns resolve success");
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time_dns_resolve = srs_utils_get_time_ms();
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time_dns_resolve = srs_utils_time_ms();
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if ((ret = __srs_rtmp_connect_server(rtmp)) != 0) {
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srs_human_trace("socket connect failed. ret=%d", ret);
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goto rtmp_destroy;
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}
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srs_human_trace("socket connect success");
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time_socket_connect = srs_utils_get_time_ms();
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time_socket_connect = srs_utils_time_ms();
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if ((ret = __srs_rtmp_do_simple_handshake(rtmp)) != 0) {
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srs_human_trace("do simple handshake failed. ret=%d", ret);
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@ -121,7 +121,7 @@ int main(int argc, char** argv)
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goto rtmp_destroy;
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}
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srs_human_trace("play stream success");
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time_play_stream = srs_utils_get_time_ms();
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time_play_stream = srs_utils_time_ms();
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for (;;) {
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if ((ret = srs_rtmp_read_packet(rtmp, &type, ×tamp, &data, &size)) != 0) {
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@ -133,7 +133,7 @@ int main(int argc, char** argv)
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if (SRS_RTMP_TYPE_VIDEO == type || SRS_RTMP_TYPE_AUDIO == type) {
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if (time_first_packet <= 0) {
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time_first_packet = srs_utils_get_time_ms();
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time_first_packet = srs_utils_time_ms();
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}
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if (basetime <= 0) {
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basetime = timestamp;
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@ -142,7 +142,7 @@ int main(int argc, char** argv)
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free(data);
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if (srs_utils_get_time_ms() - time_startup > timeout * 1000) {
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if (srs_utils_time_ms() - time_startup > timeout * 1000) {
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srs_human_trace("timeout, terminate.");
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goto rtmp_destroy;
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}
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@ -154,11 +154,11 @@ int main(int argc, char** argv)
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}
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rtmp_destroy:
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bytes_nsend = srs_utils_get_send_bytes(rtmp);
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bytes_nrecv = srs_utils_get_recv_bytes(rtmp);
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bytes_nsend = srs_utils_send_bytes(rtmp);
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bytes_nrecv = srs_utils_recv_bytes(rtmp);
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srs_rtmp_destroy(rtmp);
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time_cleanup = srs_utils_get_time_ms();
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time_cleanup = srs_utils_time_ms();
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time_duration = (int)(time_cleanup - time_startup);
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// print result to stderr.
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@ -48,7 +48,7 @@ int main(int argc, char** argv)
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int ret = 0;
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// main function
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tools_main_entrance_startup_time = srs_utils_get_time_ms();
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tools_main_entrance_startup_time = srs_utils_time_ms();
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// user option parse index.
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int opt = 0;
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@ -215,7 +215,7 @@ int connect_oc(srs_rtmp_t ortmp)
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int64_t re_create()
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{
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// if not very precise, we can directly use this as re.
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int64_t re = srs_utils_get_time_ms();
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int64_t re = srs_utils_time_ms();
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// use the starttime to get the deviation
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int64_t deviation = re - tools_main_entrance_startup_time;
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@ -236,7 +236,7 @@ int64_t re_create()
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void re_update(int64_t re, int32_t starttime, u_int32_t time)
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{
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// send by pulse algorithm.
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int64_t now = srs_utils_get_time_ms();
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int64_t now = srs_utils_time_ms();
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int64_t diff = time - starttime - (now -re);
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if (diff > RE_PULSE_MS) {
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usleep(diff * 1000);
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@ -246,7 +246,7 @@ void re_cleanup(int64_t re, int32_t starttime, u_int32_t time)
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{
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// for the last pulse, always sleep.
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// for the virtual live encoder long time publishing.
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int64_t now = srs_utils_get_time_ms();
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int64_t now = srs_utils_time_ms();
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int64_t diff = time - starttime - (now -re);
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if (diff > 0) {
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srs_human_trace("re_cleanup, diff=%d, start=%d, last=%d ms",
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@ -1712,20 +1712,20 @@ void srs_amf0_strict_array_append(srs_amf0_t amf0, srs_amf0_t value)
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obj->append(any);
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}
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int64_t srs_utils_get_time_ms()
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int64_t srs_utils_time_ms()
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{
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srs_update_system_time_ms();
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return srs_get_system_time_ms();
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}
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int64_t srs_utils_get_send_bytes(srs_rtmp_t rtmp)
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int64_t srs_utils_send_bytes(srs_rtmp_t rtmp)
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{
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srs_assert(rtmp != NULL);
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Context* context = (Context*)rtmp;
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return context->rtmp->get_send_bytes();
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}
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int64_t srs_utils_get_recv_bytes(srs_rtmp_t rtmp)
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int64_t srs_utils_recv_bytes(srs_rtmp_t rtmp)
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{
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srs_assert(rtmp != NULL);
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Context* context = (Context*)rtmp;
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@ -1773,7 +1773,7 @@ int srs_utils_parse_timestamp(
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return ret;
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}
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char srs_utils_get_flv_video_codec_id(char* data, int size)
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char srs_utils_flv_video_codec_id(char* data, int size)
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{
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if (size < 1) {
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return 0;
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@ -1785,7 +1785,7 @@ char srs_utils_get_flv_video_codec_id(char* data, int size)
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return codec_id;
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}
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char srs_utils_get_flv_video_avc_packet_type(char* data, int size)
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char srs_utils_flv_video_avc_packet_type(char* data, int size)
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{
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if (size < 2) {
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return -1;
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@ -1804,7 +1804,7 @@ char srs_utils_get_flv_video_avc_packet_type(char* data, int size)
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return avc_packet_type;
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}
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char srs_utils_get_flv_video_frame_type(char* data, int size)
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char srs_utils_flv_video_frame_type(char* data, int size)
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{
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if (size < 1) {
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return -1;
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@ -1823,6 +1823,85 @@ char srs_utils_get_flv_video_frame_type(char* data, int size)
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return frame_type;
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}
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char srs_utils_flv_audio_sound_format(char* data, int size)
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{
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if (size < 1) {
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return -1;
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}
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u_int8_t sound_format = data[0];
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sound_format = (sound_format >> 4) & 0x0f;
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if (sound_format > 15 || sound_format == 12 || sound_format == 13) {
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return -1;
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}
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return sound_format;
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}
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char srs_utils_flv_audio_sound_rate(char* data, int size)
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{
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if (size < 1) {
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return -1;
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}
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u_int8_t sound_rate = data[0];
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sound_rate = (sound_rate >> 2) & 0x03;
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if (sound_rate > 3) {
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return -1;
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}
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return sound_rate;
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}
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char srs_utils_flv_audio_sound_size(char* data, int size)
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{
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if (size < 1) {
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return -1;
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}
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u_int8_t sound_size = data[0];
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sound_size = (sound_size >> 1) & 0x01;
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if (sound_size > 1) {
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return -1;
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}
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return sound_size;
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}
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char srs_utils_flv_audio_sound_type(char* data, int size)
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{
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if (size < 1) {
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return -1;
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}
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u_int8_t sound_type = data[0];
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sound_type = sound_type & 0x01;
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if (sound_type > 1) {
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return -1;
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}
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return sound_type;
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}
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char srs_utils_flv_audio_aac_packet_type(char* data, int size)
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{
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if (size < 2) {
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return -1;
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}
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if (srs_utils_flv_audio_sound_format(data, size) != 10) {
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return -1;
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}
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u_int8_t aac_packet_type = data[1];
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aac_packet_type = aac_packet_type;
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if (aac_packet_type > 1) {
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return -1;
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}
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return aac_packet_type;
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}
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char* srs_human_amf0_print(srs_amf0_t amf0, char** pdata, int* psize)
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{
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if (!amf0) {
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@ -1876,7 +1955,7 @@ const char* srs_human_flv_video_codec_id2string(char codec_id)
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const char* srs_human_flv_video_avc_packet_type2string(char avc_packet_type)
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{
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static const char* sps_pps = "SpsPps";
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static const char* sps_pps = "SH";
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static const char* nalu = "Nalu";
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static const char* sps_pps_end = "SpsPpsEnd";
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static const char* unknown = "Unknown";
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@ -1912,6 +1991,109 @@ const char* srs_human_flv_video_frame_type2string(char frame_type)
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return unknown;
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}
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const char* srs_human_flv_audio_sound_format2string(char sound_format)
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{
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static const char* linear_pcm = "LinearPCM";
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static const char* ad_pcm = "ADPCM";
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static const char* mp3 = "MP3";
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static const char* linear_pcm_le = "LinearPCMLe";
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static const char* nellymoser_16khz = "NellymoserKHz16";
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static const char* nellymoser_8khz = "NellymoserKHz8";
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static const char* nellymoser = "Nellymoser";
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static const char* g711_a_pcm = "G711APCM";
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static const char* g711_mu_pcm = "G711MuPCM";
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static const char* reserved = "Reserved";
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static const char* aac = "AAC";
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static const char* speex = "Speex";
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static const char* mp3_8khz = "MP3KHz8";
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static const char* device_specific = "DeviceSpecific";
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static const char* unknown = "Unknown";
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switch (sound_format) {
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case 0: return linear_pcm;
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case 1: return ad_pcm;
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case 2: return mp3;
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case 3: return linear_pcm_le;
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case 4: return nellymoser_16khz;
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case 5: return nellymoser_8khz;
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case 6: return nellymoser;
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case 7: return g711_a_pcm;
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case 8: return g711_mu_pcm;
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case 9: return reserved;
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case 10: return aac;
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case 11: return speex;
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case 14: return mp3_8khz;
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case 15: return device_specific;
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default: return unknown;
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}
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return unknown;
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}
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const char* srs_human_flv_audio_sound_rate2string(char sound_rate)
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{
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static const char* khz_5_5 = "5.5KHz";
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static const char* khz_11 = "11KHz";
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static const char* khz_22 = "22KHz";
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static const char* khz_44 = "44KHz";
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static const char* unknown = "Unknown";
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switch (sound_rate) {
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case 0: return khz_5_5;
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case 1: return khz_11;
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case 2: return khz_22;
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case 3: return khz_44;
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default: return unknown;
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}
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return unknown;
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}
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const char* srs_human_flv_audio_sound_size2string(char sound_size)
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{
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static const char* bit_8 = "8bit";
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static const char* bit_16 = "16bit";
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static const char* unknown = "Unknown";
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|
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switch (sound_size) {
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case 0: return bit_8;
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case 1: return bit_16;
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default: return unknown;
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}
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|
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return unknown;
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}
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const char* srs_human_flv_audio_sound_type2string(char sound_type)
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{
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static const char* mono = "Mono";
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static const char* stereo = "Stereo";
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static const char* unknown = "Unknown";
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switch (sound_type) {
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case 0: return mono;
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case 1: return stereo;
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default: return unknown;
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}
|
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|
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return unknown;
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}
|
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|
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const char* srs_human_flv_audio_aac_packet_type2string(char aac_packet_type)
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{
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static const char* sps_pps = "SH";
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static const char* raw = "Raw";
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static const char* unknown = "Unknown";
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|
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switch (aac_packet_type) {
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case 0: return sps_pps;
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case 1: return raw;
|
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default: return unknown;
|
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}
|
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|
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return unknown;
|
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}
|
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|
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int srs_human_print_rtmp_packet(char type, u_int32_t timestamp, char* data, int size)
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{
|
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int ret = ERROR_SUCCESS;
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|
|
@ -1924,13 +2106,19 @@ int srs_human_print_rtmp_packet(char type, u_int32_t timestamp, char* data, int
|
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if (type == SRS_RTMP_TYPE_VIDEO) {
|
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srs_human_trace("Video packet type=%s, dts=%d, pts=%d, size=%d, %s(%s,%s)",
|
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srs_human_flv_tag_type2string(type), timestamp, pts, size,
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srs_human_flv_video_codec_id2string(srs_utils_get_flv_video_codec_id(data, size)),
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srs_human_flv_video_avc_packet_type2string(srs_utils_get_flv_video_avc_packet_type(data, size)),
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srs_human_flv_video_frame_type2string(srs_utils_get_flv_video_frame_type(data, size))
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srs_human_flv_video_codec_id2string(srs_utils_flv_video_codec_id(data, size)),
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srs_human_flv_video_avc_packet_type2string(srs_utils_flv_video_avc_packet_type(data, size)),
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srs_human_flv_video_frame_type2string(srs_utils_flv_video_frame_type(data, size))
|
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);
|
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} else if (type == SRS_RTMP_TYPE_AUDIO) {
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srs_human_trace("Audio packet type=%s, dts=%d, pts=%d, size=%d",
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srs_human_flv_tag_type2string(type), timestamp, pts, size);
|
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srs_human_trace("Audio packet type=%s, dts=%d, pts=%d, size=%d, %s(%s,%s,%s,%s)",
|
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srs_human_flv_tag_type2string(type), timestamp, pts, size,
|
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srs_human_flv_audio_sound_format2string(srs_utils_flv_audio_sound_format(data, size)),
|
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srs_human_flv_audio_sound_rate2string(srs_utils_flv_audio_sound_rate(data, size)),
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srs_human_flv_audio_sound_size2string(srs_utils_flv_audio_sound_size(data, size)),
|
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srs_human_flv_audio_sound_type2string(srs_utils_flv_audio_sound_type(data, size)),
|
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srs_human_flv_audio_aac_packet_type2string(srs_utils_flv_audio_aac_packet_type(data, size))
|
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);
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} else if (type == SRS_RTMP_TYPE_SCRIPT) {
|
||||
srs_human_verbose("Data packet type=%s, time=%d, size=%d",
|
||||
srs_human_flv_tag_type2string(type), timestamp, size);
|
||||
|
|
|
|||
|
|
@ -592,17 +592,17 @@ extern void srs_amf0_strict_array_append(srs_amf0_t amf0, srs_amf0_t value);
|
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* get the current system time in ms.
|
||||
* use gettimeofday() to get system time.
|
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*/
|
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extern int64_t srs_utils_get_time_ms();
|
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extern int64_t srs_utils_time_ms();
|
||||
|
||||
/**
|
||||
* get the send bytes.
|
||||
*/
|
||||
extern int64_t srs_utils_get_send_bytes(srs_rtmp_t rtmp);
|
||||
extern int64_t srs_utils_send_bytes(srs_rtmp_t rtmp);
|
||||
|
||||
/**
|
||||
* get the recv bytes.
|
||||
*/
|
||||
extern int64_t srs_utils_get_recv_bytes(srs_rtmp_t rtmp);
|
||||
extern int64_t srs_utils_recv_bytes(srs_rtmp_t rtmp);
|
||||
|
||||
/**
|
||||
* parse the dts and pts by time in header and data in tag,
|
||||
|
|
@ -635,7 +635,7 @@ extern int srs_utils_parse_timestamp(
|
|||
* 7 = AVC
|
||||
* @return the code id. 0 for error.
|
||||
*/
|
||||
extern char srs_utils_get_flv_video_codec_id(char* data, int size);
|
||||
extern char srs_utils_flv_video_codec_id(char* data, int size);
|
||||
|
||||
/**
|
||||
* get the AVCPacketType of video tag.
|
||||
|
|
@ -646,7 +646,7 @@ extern char srs_utils_get_flv_video_codec_id(char* data, int size);
|
|||
* not required or supported)
|
||||
* @return the avc packet type. -1(0xff) for error.
|
||||
*/
|
||||
extern char srs_utils_get_flv_video_avc_packet_type(char* data, int size);
|
||||
extern char srs_utils_flv_video_avc_packet_type(char* data, int size);
|
||||
|
||||
/**
|
||||
* get the FrameType of video tag.
|
||||
|
|
@ -658,7 +658,71 @@ extern char srs_utils_get_flv_video_avc_packet_type(char* data, int size);
|
|||
* 5 = video info/command frame
|
||||
* @return the frame type. 0 for error.
|
||||
*/
|
||||
extern char srs_utils_get_flv_video_frame_type(char* data, int size);
|
||||
extern char srs_utils_flv_video_frame_type(char* data, int size);
|
||||
|
||||
/**
|
||||
* get the SoundFormat of audio tag.
|
||||
* Format of SoundData. The following values are defined:
|
||||
* 0 = Linear PCM, platform endian
|
||||
* 1 = ADPCM
|
||||
* 2 = MP3
|
||||
* 3 = Linear PCM, little endian
|
||||
* 4 = Nellymoser 16 kHz mono
|
||||
* 5 = Nellymoser 8 kHz mono
|
||||
* 6 = Nellymoser
|
||||
* 7 = G.711 A-law logarithmic PCM
|
||||
* 8 = G.711 mu-law logarithmic PCM
|
||||
* 9 = reserved
|
||||
* 10 = AAC
|
||||
* 11 = Speex
|
||||
* 14 = MP3 8 kHz
|
||||
* 15 = Device-specific sound
|
||||
* Formats 7, 8, 14, and 15 are reserved.
|
||||
* AAC is supported in Flash Player 9,0,115,0 and higher.
|
||||
* Speex is supported in Flash Player 10 and higher.
|
||||
* @return the sound format. -1(0xff) for error.
|
||||
*/
|
||||
extern char srs_utils_flv_audio_sound_format(char* data, int size);
|
||||
|
||||
/**
|
||||
* get the SoundRate of audio tag.
|
||||
* Sampling rate. The following values are defined:
|
||||
* 0 = 5.5 kHz
|
||||
* 1 = 11 kHz
|
||||
* 2 = 22 kHz
|
||||
* 3 = 44 kHz
|
||||
* @return the sound rate. -1(0xff) for error.
|
||||
*/
|
||||
extern char srs_utils_flv_audio_sound_rate(char* data, int size);
|
||||
|
||||
/**
|
||||
* get the SoundSize of audio tag.
|
||||
* Size of each audio sample. This parameter only pertains to
|
||||
* uncompressed formats. Compressed formats always decode
|
||||
* to 16 bits internally.
|
||||
* 0 = 8-bit samples
|
||||
* 1 = 16-bit samples
|
||||
* @return the sound size. -1(0xff) for error.
|
||||
*/
|
||||
extern char srs_utils_flv_audio_sound_size(char* data, int size);
|
||||
|
||||
/**
|
||||
* get the SoundType of audio tag.
|
||||
* Mono or stereo sound
|
||||
* 0 = Mono sound
|
||||
* 1 = Stereo sound
|
||||
* @return the sound type. -1(0xff) for error.
|
||||
*/
|
||||
extern char srs_utils_flv_audio_sound_type(char* data, int size);
|
||||
|
||||
/**
|
||||
* get the AACPacketType of audio tag.
|
||||
* The following values are defined:
|
||||
* 0 = AAC sequence header
|
||||
* 1 = AAC raw
|
||||
* @return the aac packet type. -1(0xff) for error.
|
||||
*/
|
||||
extern char srs_utils_flv_audio_aac_packet_type(char* data, int size);
|
||||
|
||||
/*************************************************************
|
||||
**************************************************************
|
||||
|
|
@ -699,7 +763,7 @@ extern const char* srs_human_flv_video_codec_id2string(char codec_id);
|
|||
|
||||
/**
|
||||
* get the avc packet type string.
|
||||
* SpsPps = AVC sequence header
|
||||
* SH = AVC sequence header
|
||||
* Nalu = AVC NALU
|
||||
* SpsPpsEnd = AVC end of sequence
|
||||
* otherwise, "Unknown"
|
||||
|
|
@ -721,6 +785,77 @@ extern const char* srs_human_flv_video_avc_packet_type2string(char avc_packet_ty
|
|||
*/
|
||||
extern const char* srs_human_flv_video_frame_type2string(char frame_type);
|
||||
|
||||
/**
|
||||
* get the SoundFormat string.
|
||||
* Format of SoundData. The following values are defined:
|
||||
* LinearPCM = Linear PCM, platform endian
|
||||
* ADPCM = ADPCM
|
||||
* MP3 = MP3
|
||||
* LinearPCMLe = Linear PCM, little endian
|
||||
* NellymoserKHz16 = Nellymoser 16 kHz mono
|
||||
* NellymoserKHz8 = Nellymoser 8 kHz mono
|
||||
* Nellymoser = Nellymoser
|
||||
* G711APCM = G.711 A-law logarithmic PCM
|
||||
* G711MuPCM = G.711 mu-law logarithmic PCM
|
||||
* Reserved = reserved
|
||||
* AAC = AAC
|
||||
* Speex = Speex
|
||||
* MP3KHz8 = MP3 8 kHz
|
||||
* DeviceSpecific = Device-specific sound
|
||||
* otherwise, "Unknown"
|
||||
* @remark user never free the return char*,
|
||||
* it's static shared const string.
|
||||
*/
|
||||
extern const char* srs_human_flv_audio_sound_format2string(char sound_format);
|
||||
|
||||
/**
|
||||
* get the SoundRate of audio tag.
|
||||
* Sampling rate. The following values are defined:
|
||||
* 5.5KHz = 5.5 kHz
|
||||
* 11KHz = 11 kHz
|
||||
* 22KHz = 22 kHz
|
||||
* 44KHz = 44 kHz
|
||||
* otherwise, "Unknown"
|
||||
* @remark user never free the return char*,
|
||||
* it's static shared const string.
|
||||
*/
|
||||
extern const char* srs_human_flv_audio_sound_rate2string(char sound_rate);
|
||||
|
||||
/**
|
||||
* get the SoundSize of audio tag.
|
||||
* Size of each audio sample. This parameter only pertains to
|
||||
* uncompressed formats. Compressed formats always decode
|
||||
* to 16 bits internally.
|
||||
* 8bit = 8-bit samples
|
||||
* 16bit = 16-bit samples
|
||||
* otherwise, "Unknown"
|
||||
* @remark user never free the return char*,
|
||||
* it's static shared const string.
|
||||
*/
|
||||
extern const char* srs_human_flv_audio_sound_size2string(char sound_size);
|
||||
|
||||
/**
|
||||
* get the SoundType of audio tag.
|
||||
* Mono or stereo sound
|
||||
* Mono = Mono sound
|
||||
* Stereo = Stereo sound
|
||||
* otherwise, "Unknown"
|
||||
* @remark user never free the return char*,
|
||||
* it's static shared const string.
|
||||
*/
|
||||
extern const char* srs_human_flv_audio_sound_type2string(char sound_type);
|
||||
|
||||
/**
|
||||
* get the AACPacketType of audio tag.
|
||||
* The following values are defined:
|
||||
* SH = AAC sequence header
|
||||
* Raw = AAC raw
|
||||
* otherwise, "Unknown"
|
||||
* @remark user never free the return char*,
|
||||
* it's static shared const string.
|
||||
*/
|
||||
extern const char* srs_human_flv_audio_aac_packet_type2string(char aac_packet_type);
|
||||
|
||||
/**
|
||||
* print the rtmp packet, use srs_human_trace/srs_human_verbose for packet,
|
||||
* and use srs_human_raw for script data body.
|
||||
|
|
|
|||
Loading…
Reference in a new issue