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https://github.com/ossrs/srs.git
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refine librtmp, add audio video detail.
This commit is contained in:
winlin
parent
d80b580c2f
commit
d3c770d252
2 changed files with 327 additions and 4 deletions
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@ -1823,6 +1823,85 @@ char srs_utils_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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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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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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return unknown;
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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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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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return unknown;
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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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@ -1929,8 +2111,14 @@ int srs_human_print_rtmp_packet(char type, u_int32_t timestamp, char* data, int
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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) {
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srs_human_verbose("Data packet type=%s, time=%d, size=%d",
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srs_human_flv_tag_type2string(type), timestamp, size);
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@ -660,6 +660,70 @@ extern char srs_utils_flv_video_avc_packet_type(char* data, int size);
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*/
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extern char srs_utils_flv_video_frame_type(char* data, int size);
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/**
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* get the SoundFormat of audio tag.
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* Format of SoundData. The following values are defined:
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* 0 = Linear PCM, platform endian
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* 1 = ADPCM
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* 2 = MP3
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* 3 = Linear PCM, little endian
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* 4 = Nellymoser 16 kHz mono
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* 5 = Nellymoser 8 kHz mono
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* 6 = Nellymoser
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* 7 = G.711 A-law logarithmic PCM
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* 8 = G.711 mu-law logarithmic PCM
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* 9 = reserved
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* 10 = AAC
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* 11 = Speex
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* 14 = MP3 8 kHz
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* 15 = Device-specific sound
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* Formats 7, 8, 14, and 15 are reserved.
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* AAC is supported in Flash Player 9,0,115,0 and higher.
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* Speex is supported in Flash Player 10 and higher.
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* @return the sound format. -1(0xff) for error.
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*/
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extern char srs_utils_flv_audio_sound_format(char* data, int size);
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/**
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* get the SoundRate of audio tag.
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* Sampling rate. The following values are defined:
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* 0 = 5.5 kHz
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* 1 = 11 kHz
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* 2 = 22 kHz
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* 3 = 44 kHz
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* @return the sound rate. -1(0xff) for error.
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*/
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extern char srs_utils_flv_audio_sound_rate(char* data, int size);
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/**
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* get the SoundSize of audio tag.
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* Size of each audio sample. This parameter only pertains to
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* uncompressed formats. Compressed formats always decode
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* to 16 bits internally.
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* 0 = 8-bit samples
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* 1 = 16-bit samples
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* @return the sound size. -1(0xff) for error.
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*/
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extern char srs_utils_flv_audio_sound_size(char* data, int size);
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/**
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* get the SoundType of audio tag.
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* Mono or stereo sound
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* 0 = Mono sound
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* 1 = Stereo sound
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* @return the sound type. -1(0xff) for error.
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*/
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extern char srs_utils_flv_audio_sound_type(char* data, int size);
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/**
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* get the AACPacketType of audio tag.
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* The following values are defined:
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* 0 = AAC sequence header
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* 1 = AAC raw
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* @return the aac packet type. -1(0xff) for error.
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*/
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extern char srs_utils_flv_audio_aac_packet_type(char* data, int size);
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/*************************************************************
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**************************************************************
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* human readable print.
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@ -699,7 +763,7 @@ extern const char* srs_human_flv_video_codec_id2string(char codec_id);
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/**
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* get the avc packet type string.
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* SpsPps = AVC sequence header
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* SH = AVC sequence header
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* Nalu = AVC NALU
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* SpsPpsEnd = AVC end of sequence
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* otherwise, "Unknown"
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@ -721,6 +785,77 @@ extern const char* srs_human_flv_video_avc_packet_type2string(char avc_packet_ty
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*/
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extern const char* srs_human_flv_video_frame_type2string(char frame_type);
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/**
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* get the SoundFormat string.
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* Format of SoundData. The following values are defined:
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* LinearPCM = Linear PCM, platform endian
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* ADPCM = ADPCM
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* MP3 = MP3
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* LinearPCMLe = Linear PCM, little endian
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* NellymoserKHz16 = Nellymoser 16 kHz mono
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* NellymoserKHz8 = Nellymoser 8 kHz mono
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* Nellymoser = Nellymoser
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* G711APCM = G.711 A-law logarithmic PCM
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* G711MuPCM = G.711 mu-law logarithmic PCM
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* Reserved = reserved
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* AAC = AAC
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* Speex = Speex
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* MP3KHz8 = MP3 8 kHz
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* DeviceSpecific = Device-specific sound
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* otherwise, "Unknown"
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* @remark user never free the return char*,
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* it's static shared const string.
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*/
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extern const char* srs_human_flv_audio_sound_format2string(char sound_format);
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/**
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* get the SoundRate of audio tag.
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* Sampling rate. The following values are defined:
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* 5.5KHz = 5.5 kHz
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* 11KHz = 11 kHz
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* 22KHz = 22 kHz
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* 44KHz = 44 kHz
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* otherwise, "Unknown"
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* @remark user never free the return char*,
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* it's static shared const string.
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*/
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extern const char* srs_human_flv_audio_sound_rate2string(char sound_rate);
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/**
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* get the SoundSize of audio tag.
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* Size of each audio sample. This parameter only pertains to
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* uncompressed formats. Compressed formats always decode
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* to 16 bits internally.
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* 8bit = 8-bit samples
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* 16bit = 16-bit samples
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* otherwise, "Unknown"
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* @remark user never free the return char*,
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* it's static shared const string.
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*/
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extern const char* srs_human_flv_audio_sound_size2string(char sound_size);
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/**
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* get the SoundType of audio tag.
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* Mono or stereo sound
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* Mono = Mono sound
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* Stereo = Stereo sound
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* otherwise, "Unknown"
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* @remark user never free the return char*,
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* it's static shared const string.
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*/
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extern const char* srs_human_flv_audio_sound_type2string(char sound_type);
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/**
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* get the AACPacketType of audio tag.
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* The following values are defined:
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* SH = AAC sequence header
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* Raw = AAC raw
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* otherwise, "Unknown"
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* @remark user never free the return char*,
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* it's static shared const string.
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*/
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extern const char* srs_human_flv_audio_aac_packet_type2string(char aac_packet_type);
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/**
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* print the rtmp packet, use srs_human_trace/srs_human_verbose for packet,
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* and use srs_human_raw for script data body.
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