for ossrs#133, rtsp parse aac from rtp packet.

trunk/src/kernel/srs_kernel_error.hpp

@@ -145,6 +145,8 @@ CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
 #define ERROR_RTSP_TOKEN_NOT_NORMAL         2042
 #define ERROR_RTSP_REQUEST_HEADER_EOF       2043
 #define ERROR_RTP_HEADER_CORRUPT            2044
+#define ERROR_RTP_TYPE96_CORRUPT            2045
+#define ERROR_RTP_TYPE97_CORRUPT            2046
 //                                           
 // system control message, 
 // not an error, but special control logic.

trunk/src/protocol/srs_rtsp_stack.cpp

@@ -35,6 +35,7 @@ using namespace std;
 #include <srs_core_autofree.hpp>
 #include <srs_kernel_utility.hpp>
 #include <srs_kernel_stream.hpp>
+#include <srs_kernel_codec.hpp>
 
 #ifdef SRS_AUTO_STREAM_CASTER
 
@@ -133,13 +134,15 @@ SrsRtpPacket::SrsRtpPacket()
     ssrc = 0;
 
     payload = new SrsSimpleBuffer();
+    audio_samples = new SrsCodecSample();
     chunked = false;
     completed = false;
 }
 
 SrsRtpPacket::~SrsRtpPacket()
 {
     srs_freep(payload);
+    srs_freep(audio_samples);
 }
 
 void SrsRtpPacket::copy(SrsRtpPacket* src)
@@ -156,22 +159,28 @@ void SrsRtpPacket::copy(SrsRtpPacket* src)
 
     chunked = src->chunked;
     completed = src->completed;
+    audio_samples = new SrsCodecSample();
 }
 
 void SrsRtpPacket::reap(SrsRtpPacket* src)
 {
     copy(src);
 
+    srs_freep(payload);
     payload = src->payload;
     src->payload = NULL;
+
+    srs_freep(audio_samples);
+    audio_samples = src->audio_samples;
+    src->audio_samples = NULL;
 }
 
 int SrsRtpPacket::decode(SrsStream* stream)
 {
     int ret = ERROR_SUCCESS;
 
-    // 12bytes header, atleast 2bytes content.
-    if (!stream->require(14)) {
+    // 12bytes header
+    if (!stream->require(12)) {
         ret = ERROR_RTP_HEADER_CORRUPT;
         srs_error("rtsp: rtp header corrupt. ret=%d", ret);
         return ret;
@@ -191,6 +200,92 @@ int SrsRtpPacket::decode(SrsStream* stream)
     timestamp = stream->read_4bytes();
     ssrc = stream->read_4bytes();
 
+    // video codec.
+    if (payload_type == 96) {
+        return decode_96(stream);
+    } else if (payload_type == 97) {
+        return decode_97(stream);
+    }
+
+    return ret;
+}
+
+int SrsRtpPacket::decode_97(SrsStream* stream)
+{
+    int ret = ERROR_SUCCESS;
+
+    // atleast 2bytes content.
+    if (!stream->require(2)) {
+        ret = ERROR_RTP_TYPE97_CORRUPT;
+        srs_error("rtsp: rtp type97 corrupt. ret=%d", ret);
+        return ret;
+    }
+
+    int8_t hasv = stream->read_1bytes();
+    int8_t lasv = stream->read_1bytes();
+    u_int16_t au_size = ((hasv << 5) & 0xE0) | ((lasv >> 3) & 0x1f);
+
+    if (!stream->require(au_size)) {
+        ret = ERROR_RTP_TYPE97_CORRUPT;
+        srs_error("rtsp: rtp type97 au_size corrupt. ret=%d", ret);
+        return ret;
+    }
+
+    int nb_samples = au_size / 2;
+    int guess_sample_size = (stream->size() - stream->pos() - au_size) / nb_samples;
+    int required_size = 0;
+
+    // append left bytes to payload.
+    payload->append(
+        stream->data() + stream->pos() + au_size, 
+        stream->size() - stream->pos() - au_size
+    );
+    char* p = payload->bytes();
+
+    for (int i = 0; i < au_size; i += 2) {
+        hasv = stream->read_1bytes();
+        lasv = stream->read_1bytes();
+
+        u_int16_t sample_size = ((hasv << 5) & 0xE0) | ((lasv >> 3) & 0x1f);
+        if (sample_size != guess_sample_size) {
+            // guess the size lost 0x100.
+            if (guess_sample_size == (sample_size | 0x100)) {
+                sample_size = guess_sample_size;
+            }
+        }
+
+        char* sample = p + required_size;
+        required_size += sample_size;
+
+        if (!stream->require(required_size)) {
+            ret = ERROR_RTP_TYPE97_CORRUPT;
+            srs_error("rtsp: rtp type97 samples corrupt. ret=%d", ret);
+            return ret;
+        }
+
+        if ((ret = audio_samples->add_sample_unit(sample, sample_size)) != ERROR_SUCCESS) {
+            srs_error("rtsp: rtp type97 add sample failed. ret=%d", ret);
+            return ret;
+        }
+    }
+
+    // parsed ok.
+    completed = true;
+
+    return ret;
+}
+
+int SrsRtpPacket::decode_96(SrsStream* stream)
+{
+    int ret = ERROR_SUCCESS;
+
+    // atleast 2bytes content.
+    if (!stream->require(2)) {
+        ret = ERROR_RTP_TYPE96_CORRUPT;
+        srs_error("rtsp: rtp type96 corrupt. ret=%d", ret);
+        return ret;
+    }
+
     // frame type
     // 0... .... reserverd
     // .11. .... NALU[0]&0x60

trunk/src/protocol/srs_rtsp_stack.hpp

@@ -39,6 +39,7 @@ CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
 
 class SrsStream;
 class SrsSimpleBuffer;
+class SrsCodecSample;
 class ISrsProtocolReaderWriter;
 
 // rtsp specification
@@ -292,6 +293,11 @@ class SrsRtpPacket
     // normal message always completed.
     // while chunked completed when the last chunk arriaved.
     bool completed;
+
+    /**
+    * the audio samples, one rtp packets may contains multiple audio samples.
+    */
+    SrsCodecSample* audio_samples;
 public:
     SrsRtpPacket();
     virtual ~SrsRtpPacket();
@@ -308,6 +314,9 @@ class SrsRtpPacket
     * decode rtp packet from stream.
     */
     virtual int decode(SrsStream* stream);
+private:
+    virtual int decode_97(SrsStream* stream);
+    virtual int decode_96(SrsStream* stream);
 };
 
 /**