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for #133, rtsp parse aac from rtp packet.

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This commit is contained in:
winlin 2015-02-18 13:37:08 +08:00
parent 0cc693a3b8
commit a954040d29
3 changed files with 108 additions and 2 deletions
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99
trunk/src/protocol/srs_rtsp_stack.cpp
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@ -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
@ -133,6 +134,7 @@ SrsRtpPacket::SrsRtpPacket()
ssrc = 0;
payload = new SrsSimpleBuffer();
audio_samples = new SrsCodecSample();
chunked = false;
completed = false;
}
@ -140,6 +142,7 @@ SrsRtpPacket::SrsRtpPacket()
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