diff --git a/README.md b/README.md
index c7d532688..25fd07033 100755
--- a/README.md
+++ b/README.md
@@ -482,6 +482,7 @@ Supported operating systems and hardware:
* 2013-10-17, Created.
## History
+* v2.0, 2014-11-20, fix [#212](https://github.com/winlinvip/simple-rtmp-server/issues/212), support publish audio raw frames. 2.0.27
* v2.0, 2014-11-19, fix [#213](https://github.com/winlinvip/simple-rtmp-server/issues/213), support compile [srs-librtmp on windows](https://github.com/winlinvip/srs.librtmp), [bug #213](https://github.com/winlinvip/simple-rtmp-server/issues/213). 2.0.26
* v2.0, 2014-11-18, all wiki translated to English. 2.0.23.
* v2.0, 2014-11-15, fix [#204](https://github.com/winlinvip/simple-rtmp-server/issues/204), srs-librtmp drop duplicated sps/pps(sequence header). 2.0.22.
diff --git a/trunk/research/librtmp/Makefile b/trunk/research/librtmp/Makefile
index acc2cb3d1..d5f19606e 100755
--- a/trunk/research/librtmp/Makefile
+++ b/trunk/research/librtmp/Makefile
@@ -6,7 +6,8 @@ else
ST_ALL = objs/srs_flv_parser \
objs/srs_flv_injecter objs/srs_publish objs/srs_play \
objs/srs_ingest_flv objs/srs_ingest_rtmp objs/srs_detect_rtmp \
- objs/srs_bandwidth_check objs/srs_h264_raw_publish
+ objs/srs_bandwidth_check objs/srs_h264_raw_publish \
+ objs/srs_audio_raw_publish
endif
.PHONY: default clean help ssl nossl
@@ -24,6 +25,7 @@ help:
@echo " srs_flv_injecter inject keyframes information to metadata."
@echo " srs_publish publish program using srs-librtmp"
@echo " srs_h264_raw_publish publish raw h.264 stream to SSR by srs-librtmp"
+ @echo " srs_audio_raw_publish publish raw audio stream to SSR by srs-librtmp"
@echo " srs_play play program using srs-librtmp"
@echo " srs_ingest_flv ingest flv file and publish to RTMP server."
@echo " srs_ingest_rtmp ingest RTMP and publish to RTMP server."
@@ -85,6 +87,9 @@ objs/srs_publish: srs_publish.c $(SRS_RESEARCH_DEPS) $(SRS_LIBRTMP_I) $(SRS_LIBR
objs/srs_h264_raw_publish: srs_h264_raw_publish.c $(SRS_RESEARCH_DEPS) $(SRS_LIBRTMP_I) $(SRS_LIBRTMP_L) $(SRS_LIBSSL_L)
$(GCC) srs_h264_raw_publish.c $(SRS_LIBRTMP_L) $(SRS_LIBSSL_L) $(EXTRA_CXX_FLAG) -o objs/srs_h264_raw_publish
+objs/srs_audio_raw_publish: srs_audio_raw_publish.c $(SRS_RESEARCH_DEPS) $(SRS_LIBRTMP_I) $(SRS_LIBRTMP_L) $(SRS_LIBSSL_L)
+ $(GCC) srs_audio_raw_publish.c $(SRS_LIBRTMP_L) $(SRS_LIBSSL_L) $(EXTRA_CXX_FLAG) -o objs/srs_audio_raw_publish
+
objs/srs_play: srs_play.c $(SRS_RESEARCH_DEPS) $(SRS_LIBRTMP_I) $(SRS_LIBRTMP_L) $(SRS_LIBSSL_L)
$(GCC) srs_play.c $(SRS_LIBRTMP_L) $(SRS_LIBSSL_L) $(EXTRA_CXX_FLAG) -o objs/srs_play
diff --git a/trunk/research/librtmp/srs_audio_raw_publish.c b/trunk/research/librtmp/srs_audio_raw_publish.c
new file mode 100644
index 000000000..4c3dbb3a9
--- /dev/null
+++ b/trunk/research/librtmp/srs_audio_raw_publish.c
@@ -0,0 +1,190 @@
+/*
+The MIT License (MIT)
+
+Copyright (c) 2013-2014 winlin
+
+Permission is hereby granted, free of charge, to any person obtaining a copy of
+this software and associated documentation files (the "Software"), to deal in
+the Software without restriction, including without limitation the rights to
+use, copy, modify, merge, publish, distribute, sublicense, and/or sell copies of
+the Software, and to permit persons to whom the Software is furnished to do so,
+subject to the following conditions:
+
+The above copyright notice and this permission notice shall be included in all
+copies or substantial portions of the Software.
+
+THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
+IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, FITNESS
+FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE AUTHORS OR
+COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER
+IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN
+CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
+*/
+/**
+gcc srs_audio_raw_publish.c ../../objs/lib/srs_librtmp.a -g -O0 -lstdc++ -o srs_audio_raw_publish
+*/
+
+#include
+#include
+#include
+
+// for open audio raw file.
+#include
+#include
+#include
+
+#include "../../objs/include/srs_librtmp.h"
+
+// https://github.com/winlinvip/simple-rtmp-server/issues/212#issuecomment-63648892
+// allspace:
+// Take this file as an example: https://github.com/allspace/files/blob/master/srs.pcm
+// It's captured using SDK callback method. I have filtered out h264 video, so it's audio only now.
+// For every frame, it's a 8 bytes vendor specific header, following 160 bytes audio frame.
+// The header part can be ignored.
+int read_audio_frame(char* audio_raw, int file_size, char** pp, char** pdata, int* psize)
+{
+ char* p = *pp;
+
+ if (file_size - (p - audio_raw) < 168) {
+ srs_lib_trace("audio must be 160+8 bytes. left %d bytes.",
+ file_size - (p - audio_raw));
+ return - 1;
+ }
+
+ // ignore 8bytes vendor specific header.
+ p += 8;
+
+ // 160 bytes audio frame
+ *pdata = p;
+ *psize = 160;
+
+ // next frame.
+ *pp = p + *psize;
+
+ return 0;
+}
+
+int main(int argc, char** argv)
+{
+ printf("publish raw audio as rtmp stream to server like FMLE/FFMPEG/Encoder\n");
+ printf("SRS(simple-rtmp-server) client librtmp library.\n");
+ printf("version: %d.%d.%d\n", srs_version_major(), srs_version_minor(), srs_version_revision());
+
+ if (argc <= 2) {
+ printf("Usage: %s \n", argv[0]);
+ printf(" audio_raw_file: the audio raw steam file.\n");
+ printf(" rtmp_publish_url: the rtmp publish url.\n");
+ printf("For example:\n");
+ printf(" %s ./audio.raw.pcm rtmp://127.0.0.1:1935/live/livestream\n", argv[0]);
+ printf("Where the file: http://winlinvip.github.io/srs.release/3rdparty/audio.raw.pcm\n");
+ printf("See: https://github.com/winlinvip/simple-rtmp-server/issues/212\n");
+ exit(-1);
+ }
+
+ const char* raw_file = argv[1];
+ const char* rtmp_url = argv[2];
+ srs_lib_trace("raw_file=%s, rtmp_url=%s", raw_file, rtmp_url);
+
+ // open file
+ int raw_fd = open(raw_file, O_RDONLY);
+ if (raw_fd < 0) {
+ srs_lib_trace("open audio raw file %s failed.", raw_fd);
+ goto rtmp_destroy;
+ }
+
+ off_t file_size = lseek(raw_fd, 0, SEEK_END);
+ if (file_size <= 0) {
+ srs_lib_trace("audio raw file %s empty.", raw_file);
+ goto rtmp_destroy;
+ }
+ srs_lib_trace("read entirely audio raw file, size=%dKB", (int)(file_size / 1024));
+
+ char* audio_raw = (char*)malloc(file_size);
+ if (!audio_raw) {
+ srs_lib_trace("alloc raw buffer failed for file %s.", raw_file);
+ goto rtmp_destroy;
+ }
+
+ lseek(raw_fd, 0, SEEK_SET);
+ ssize_t nb_read = 0;
+ if ((nb_read = read(raw_fd, audio_raw, file_size)) != file_size) {
+ srs_lib_trace("buffer %s failed, expect=%dKB, actual=%dKB.",
+ raw_file, (int)(file_size / 1024), (int)(nb_read / 1024));
+ goto rtmp_destroy;
+ }
+
+ // connect rtmp context
+ srs_rtmp_t rtmp = srs_rtmp_create(rtmp_url);
+
+ if (srs_simple_handshake(rtmp) != 0) {
+ srs_lib_trace("simple handshake failed.");
+ goto rtmp_destroy;
+ }
+ srs_lib_trace("simple handshake success");
+
+ if (srs_connect_app(rtmp) != 0) {
+ srs_lib_trace("connect vhost/app failed.");
+ goto rtmp_destroy;
+ }
+ srs_lib_trace("connect vhost/app success");
+
+ if (srs_publish_stream(rtmp) != 0) {
+ srs_lib_trace("publish stream failed.");
+ goto rtmp_destroy;
+ }
+ srs_lib_trace("publish stream success");
+
+ u_int32_t timestamp = 0;
+ u_int32_t time_delta = 17;
+ // @remark, to decode the file.
+ char* p = audio_raw;
+ for (;p < audio_raw + file_size;) {
+ // @remark, read a frame from file buffer.
+ char* data = NULL;
+ int size = 0;
+ if (read_audio_frame(audio_raw, file_size, &p, &data, &size) < 0) {
+ srs_lib_trace("read a frame from file buffer failed.");
+ goto rtmp_destroy;
+ }
+
+ // 0 = Linear PCM, platform endian
+ // 1 = ADPCM
+ // 2 = MP3
+ // 7 = G.711 A-law logarithmic PCM
+ // 8 = G.711 mu-law logarithmic PCM
+ // 10 = AAC
+ // 11 = Speex
+ char sound_format = 1;
+ // 3 = 44 kHz
+ char sound_rate = 3;
+ // 1 = 16-bit samples
+ char sound_size = 1;
+ // 1 = Stereo sound
+ char sound_type = 1;
+
+ timestamp += time_delta;
+
+ if (srs_audio_write_raw_frame(rtmp,
+ sound_format, sound_rate, sound_size, sound_type,
+ 0, data, size, timestamp) != 0
+ ) {
+ srs_lib_trace("send audio raw data failed.");
+ goto rtmp_destroy;
+ }
+
+ srs_lib_trace("sent packet: type=%s, time=%d, size=%d, codec=%d, rate=%d, sample=%d, channel=%d",
+ srs_type2string(SRS_RTMP_TYPE_AUDIO), timestamp, size, sound_format, sound_rate, sound_size,
+ sound_type);
+
+ // @remark, when use encode device, it not need to sleep.
+ usleep(1000 * time_delta);
+ }
+
+rtmp_destroy:
+ srs_rtmp_destroy(rtmp);
+ close(raw_fd);
+ free(audio_raw);
+
+ return 0;
+}
+
diff --git a/trunk/src/core/srs_core.hpp b/trunk/src/core/srs_core.hpp
index 390e5427f..c277518df 100644
--- a/trunk/src/core/srs_core.hpp
+++ b/trunk/src/core/srs_core.hpp
@@ -31,7 +31,7 @@ CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
// current release version
#define VERSION_MAJOR 2
#define VERSION_MINOR 0
-#define VERSION_REVISION 26
+#define VERSION_REVISION 27
// server info.
#define RTMP_SIG_SRS_KEY "SRS"
#define RTMP_SIG_SRS_ROLE "origin/edge server"
diff --git a/trunk/src/libs/srs_librtmp.cpp b/trunk/src/libs/srs_librtmp.cpp
index f2be538f4..e789d21a8 100644
--- a/trunk/src/libs/srs_librtmp.cpp
+++ b/trunk/src/libs/srs_librtmp.cpp
@@ -1438,6 +1438,46 @@ char* srs_amf0_human_print(srs_amf0_t amf0, char** pdata, int* psize)
return any->human_print(pdata, psize);
}
+/**
+* write audio raw frame to SRS.
+*/
+int srs_audio_write_raw_frame(srs_rtmp_t rtmp,
+ char sound_format, char sound_rate, char sound_size, char sound_type,
+ char aac_packet_type, char* frame, int frame_size, u_int32_t timestamp
+) {
+ int ret = ERROR_SUCCESS;
+
+ Context* context = (Context*)rtmp;
+ srs_assert(context);
+
+ // TODO: FIXME: for aac, must send the sequence header first.
+
+ // for audio frame, there is 1 or 2 bytes header:
+ // 1bytes, SoundFormat|SoundRate|SoundSize|SoundType
+ // 1bytes, AACPacketType for SoundFormat == 10
+ int size = frame_size + 1;
+ if (aac_packet_type == SrsCodecAudioAAC) {
+ size += 1;
+ }
+ char* data = new char[size];
+ char* p = data;
+
+ u_int8_t audio_header = sound_type & 0x01;
+ audio_header |= (sound_size << 1) & 0x02;
+ audio_header |= (sound_rate << 2) & 0x0c;
+ audio_header |= (sound_format << 4) & 0xf0;
+
+ *p++ = audio_header;
+
+ if (aac_packet_type == SrsCodecAudioAAC) {
+ *p++ = aac_packet_type;
+ }
+
+ memcpy(p, frame, frame_size);
+
+ return srs_write_packet(context, SRS_RTMP_TYPE_AUDIO, timestamp, data, size);
+}
+
/**
* write h264 packet, with rtmp header.
* @param frame_type, SrsCodecVideoAVCFrameKeyFrame or SrsCodecVideoAVCFrameInterFrame.
@@ -1458,7 +1498,6 @@ int __srs_write_h264_packet(Context* context,
// @see: E.4.3 Video Tags, video_file_format_spec_v10_1.pdf, page 78
int size = h264_raw_size + 5;
char* data = new char[size];
- memcpy(data + 5, h264_raw_data, h264_raw_size);
char* p = data;
// @see: E.4.3 Video Tags, video_file_format_spec_v10_1.pdf, page 78
@@ -1480,6 +1519,9 @@ int __srs_write_h264_packet(Context* context,
*p++ = pp[1];
*p++ = pp[0];
+ // h.264 raw data.
+ memcpy(p, h264_raw_data, h264_raw_size);
+
return srs_write_packet(context, SRS_RTMP_TYPE_VIDEO, timestamp, data, size);
}
diff --git a/trunk/src/libs/srs_librtmp.hpp b/trunk/src/libs/srs_librtmp.hpp
index 8a9d148c9..6df3be0ba 100644
--- a/trunk/src/libs/srs_librtmp.hpp
+++ b/trunk/src/libs/srs_librtmp.hpp
@@ -461,6 +461,64 @@ extern void srs_amf0_strict_array_append(srs_amf0_t amf0, srs_amf0_t value);
*/
extern char* srs_amf0_human_print(srs_amf0_t amf0, char** pdata, int* psize);
+/*************************************************************
+**************************************************************
+* audio raw codec
+**************************************************************
+*************************************************************/
+/**
+* write an audio raw frame to srs.
+* not similar to h.264 video, the audio never aggregated, always
+* encoded one frame by one, so this api is used to write a frame.
+*
+* @param sound_format 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.
+* @param sound_rate Sampling rate. The following values are defined:
+* 0 = 5.5 kHz
+* 1 = 11 kHz
+* 2 = 22 kHz
+* 3 = 44 kHz
+* @param sound_size 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
+* @param sound_type Mono or stereo sound
+* 0 = Mono sound
+* 1 = Stereo sound
+* @param aac_packet_type The following values are defined:
+* 0 = AAC sequence header
+* 1 = AAC raw
+* @param timestamp The timestamp of audio.
+*
+* @remark Ignore aac_packet_type if not aac(sound_format!=10).
+*
+* @see https://github.com/winlinvip/simple-rtmp-server/issues/212
+* @see E.4.2.1 AUDIODATA of video_file_format_spec_v10_1.pdf
+*
+* @return 0, success; otherswise, failed.
+*/
+extern int srs_audio_write_raw_frame(srs_rtmp_t rtmp,
+ char sound_format, char sound_rate, char sound_size, char sound_type,
+ char aac_packet_type, char* frame, int frame_size, u_int32_t timestamp
+);
+
/*************************************************************
**************************************************************
* h264 raw codec
@@ -474,7 +532,7 @@ typedef int srs_h264_bool;
* each frame prefixed h.264 annexb header, by N[00] 00 00 01, where N>=0,
* for instance, frame = header(00 00 00 01) + payload(67 42 80 29 95 A0 14 01 6E 40)
* about annexb, @see H.264-AVC-ISO_IEC_14496-10.pdf, page 211.
-* @paam frames_size the size of h264 raw data.
+* @param frames_size the size of h264 raw data.
* assert frames_size > 0, at least has 1 bytes header.
* @param dts the dts of h.264 raw data.
* @param pts the pts of h.264 raw data.
diff --git a/trunk/src/srs/srs.upp b/trunk/src/srs/srs.upp
index 5fa0bfbf9..da92f3ebf 100755
--- a/trunk/src/srs/srs.upp
+++ b/trunk/src/srs/srs.upp
@@ -128,6 +128,7 @@ file
..\utest\srs_utest_reload.hpp,
..\utest\srs_utest_reload.cpp,
research readonly separator,
+ ..\..\research\librtmp\srs_audio_raw_publish.c,
..\..\research\librtmp\srs_bandwidth_check.c,
..\..\research\librtmp\srs_detect_rtmp.c,
..\..\research\librtmp\srs_flv_injecter.c,