blob: 2a23c100597f661f09a2cd4e42675651d076ef33 [file] [log] [blame]
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001/*
2 * libjingle
3 * Copyright 2012 Google Inc.
4 *
5 * Redistribution and use in source and binary forms, with or without
6 * modification, are permitted provided that the following conditions are met:
7 *
8 * 1. Redistributions of source code must retain the above copyright notice,
9 * this list of conditions and the following disclaimer.
10 * 2. Redistributions in binary form must reproduce the above copyright notice,
11 * this list of conditions and the following disclaimer in the documentation
12 * and/or other materials provided with the distribution.
13 * 3. The name of the author may not be used to endorse or promote products
14 * derived from this software without specific prior written permission.
15 *
16 * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
17 * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
18 * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
19 * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
20 * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
21 * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
22 * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
23 * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
24 * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
25 * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
26 */
27
28#include "talk/media/base/rtpdataengine.h"
29
30#include "talk/base/buffer.h"
31#include "talk/base/helpers.h"
32#include "talk/base/logging.h"
33#include "talk/base/ratelimiter.h"
34#include "talk/base/timing.h"
35#include "talk/media/base/codec.h"
36#include "talk/media/base/constants.h"
37#include "talk/media/base/rtputils.h"
38#include "talk/media/base/streamparams.h"
39
40namespace cricket {
41
42// We want to avoid IP fragmentation.
43static const size_t kDataMaxRtpPacketLen = 1200U;
44// We reserve space after the RTP header for future wiggle room.
45static const unsigned char kReservedSpace[] = {
46 0x00, 0x00, 0x00, 0x00
47};
48
49// Amount of overhead SRTP may take. We need to leave room in the
50// buffer for it, otherwise SRTP will fail later. If SRTP ever uses
51// more than this, we need to increase this number.
52static const size_t kMaxSrtpHmacOverhead = 16;
53
54RtpDataEngine::RtpDataEngine() {
55 data_codecs_.push_back(
56 DataCodec(kGoogleRtpDataCodecId,
57 kGoogleRtpDataCodecName, 0));
58 SetTiming(new talk_base::Timing());
59}
60
61DataMediaChannel* RtpDataEngine::CreateChannel(
62 DataChannelType data_channel_type) {
63 if (data_channel_type != DCT_RTP) {
64 return NULL;
65 }
66 return new RtpDataMediaChannel(timing_.get());
67}
68
69// TODO(pthatcher): Should we move these find/get functions somewhere
70// common?
71bool FindCodecById(const std::vector<DataCodec>& codecs,
72 int id, DataCodec* codec_out) {
73 std::vector<DataCodec>::const_iterator iter;
74 for (iter = codecs.begin(); iter != codecs.end(); ++iter) {
75 if (iter->id == id) {
76 *codec_out = *iter;
77 return true;
78 }
79 }
80 return false;
81}
82
83bool FindCodecByName(const std::vector<DataCodec>& codecs,
84 const std::string& name, DataCodec* codec_out) {
85 std::vector<DataCodec>::const_iterator iter;
86 for (iter = codecs.begin(); iter != codecs.end(); ++iter) {
87 if (iter->name == name) {
88 *codec_out = *iter;
89 return true;
90 }
91 }
92 return false;
93}
94
95RtpDataMediaChannel::RtpDataMediaChannel(talk_base::Timing* timing) {
96 Construct(timing);
97}
98
99RtpDataMediaChannel::RtpDataMediaChannel() {
100 Construct(NULL);
101}
102
103void RtpDataMediaChannel::Construct(talk_base::Timing* timing) {
104 sending_ = false;
105 receiving_ = false;
106 timing_ = timing;
107 send_limiter_.reset(new talk_base::RateLimiter(kDataMaxBandwidth / 8, 1.0));
108}
109
110
111RtpDataMediaChannel::~RtpDataMediaChannel() {
112 std::map<uint32, RtpClock*>::const_iterator iter;
113 for (iter = rtp_clock_by_send_ssrc_.begin();
114 iter != rtp_clock_by_send_ssrc_.end();
115 ++iter) {
116 delete iter->second;
117 }
118}
119
120void RtpClock::Tick(
121 double now, int* seq_num, uint32* timestamp) {
122 *seq_num = ++last_seq_num_;
123 *timestamp = timestamp_offset_ + static_cast<uint32>(now * clockrate_);
124}
125
126const DataCodec* FindUnknownCodec(const std::vector<DataCodec>& codecs) {
127 DataCodec data_codec(kGoogleRtpDataCodecId, kGoogleRtpDataCodecName, 0);
128 std::vector<DataCodec>::const_iterator iter;
129 for (iter = codecs.begin(); iter != codecs.end(); ++iter) {
130 if (!iter->Matches(data_codec)) {
131 return &(*iter);
132 }
133 }
134 return NULL;
135}
136
137const DataCodec* FindKnownCodec(const std::vector<DataCodec>& codecs) {
138 DataCodec data_codec(kGoogleRtpDataCodecId, kGoogleRtpDataCodecName, 0);
139 std::vector<DataCodec>::const_iterator iter;
140 for (iter = codecs.begin(); iter != codecs.end(); ++iter) {
141 if (iter->Matches(data_codec)) {
142 return &(*iter);
143 }
144 }
145 return NULL;
146}
147
148bool RtpDataMediaChannel::SetRecvCodecs(const std::vector<DataCodec>& codecs) {
149 const DataCodec* unknown_codec = FindUnknownCodec(codecs);
150 if (unknown_codec) {
151 LOG(LS_WARNING) << "Failed to SetRecvCodecs because of unknown codec: "
152 << unknown_codec->ToString();
153 return false;
154 }
155
156 recv_codecs_ = codecs;
157 return true;
158}
159
160bool RtpDataMediaChannel::SetSendCodecs(const std::vector<DataCodec>& codecs) {
161 const DataCodec* known_codec = FindKnownCodec(codecs);
162 if (!known_codec) {
163 LOG(LS_WARNING) <<
164 "Failed to SetSendCodecs because there is no known codec.";
165 return false;
166 }
167
168 send_codecs_ = codecs;
169 return true;
170}
171
172bool RtpDataMediaChannel::AddSendStream(const StreamParams& stream) {
173 if (!stream.has_ssrcs()) {
174 return false;
175 }
176
177 StreamParams found_stream;
178 if (GetStreamBySsrc(send_streams_, stream.first_ssrc(), &found_stream)) {
179 LOG(LS_WARNING) << "Not adding data send stream '" << stream.id
180 << "' with ssrc=" << stream.first_ssrc()
181 << " because stream already exists.";
182 return false;
183 }
184
185 send_streams_.push_back(stream);
186 // TODO(pthatcher): This should be per-stream, not per-ssrc.
187 // And we should probably allow more than one per stream.
188 rtp_clock_by_send_ssrc_[stream.first_ssrc()] = new RtpClock(
189 kDataCodecClockrate,
190 talk_base::CreateRandomNonZeroId(), talk_base::CreateRandomNonZeroId());
191
192 LOG(LS_INFO) << "Added data send stream '" << stream.id
193 << "' with ssrc=" << stream.first_ssrc();
194 return true;
195}
196
197bool RtpDataMediaChannel::RemoveSendStream(uint32 ssrc) {
198 StreamParams found_stream;
199 if (!GetStreamBySsrc(send_streams_, ssrc, &found_stream)) {
200 return false;
201 }
202
203 RemoveStreamBySsrc(&send_streams_, ssrc);
204 delete rtp_clock_by_send_ssrc_[ssrc];
205 rtp_clock_by_send_ssrc_.erase(ssrc);
206 return true;
207}
208
209bool RtpDataMediaChannel::AddRecvStream(const StreamParams& stream) {
210 if (!stream.has_ssrcs()) {
211 return false;
212 }
213
214 StreamParams found_stream;
215 if (GetStreamBySsrc(recv_streams_, stream.first_ssrc(), &found_stream)) {
216 LOG(LS_WARNING) << "Not adding data recv stream '" << stream.id
217 << "' with ssrc=" << stream.first_ssrc()
218 << " because stream already exists.";
219 return false;
220 }
221
222 recv_streams_.push_back(stream);
223 LOG(LS_INFO) << "Added data recv stream '" << stream.id
224 << "' with ssrc=" << stream.first_ssrc();
225 return true;
226}
227
228bool RtpDataMediaChannel::RemoveRecvStream(uint32 ssrc) {
229 RemoveStreamBySsrc(&recv_streams_, ssrc);
230 return true;
231}
232
233void RtpDataMediaChannel::OnPacketReceived(talk_base::Buffer* packet) {
234 RtpHeader header;
235 if (!GetRtpHeader(packet->data(), packet->length(), &header)) {
236 // Don't want to log for every corrupt packet.
237 // LOG(LS_WARNING) << "Could not read rtp header from packet of length "
238 // << packet->length() << ".";
239 return;
240 }
241
242 size_t header_length;
243 if (!GetRtpHeaderLen(packet->data(), packet->length(), &header_length)) {
244 // Don't want to log for every corrupt packet.
245 // LOG(LS_WARNING) << "Could not read rtp header"
246 // << length from packet of length "
247 // << packet->length() << ".";
248 return;
249 }
250 const char* data = packet->data() + header_length + sizeof(kReservedSpace);
251 size_t data_len = packet->length() - header_length - sizeof(kReservedSpace);
252
253 if (!receiving_) {
254 LOG(LS_WARNING) << "Not receiving packet "
255 << header.ssrc << ":" << header.seq_num
256 << " before SetReceive(true) called.";
257 return;
258 }
259
260 DataCodec codec;
261 if (!FindCodecById(recv_codecs_, header.payload_type, &codec)) {
262 LOG(LS_WARNING) << "Not receiving packet "
263 << header.ssrc << ":" << header.seq_num
264 << " (" << data_len << ")"
265 << " because unknown payload id: " << header.payload_type;
266 return;
267 }
268
269 StreamParams found_stream;
270 if (!GetStreamBySsrc(recv_streams_, header.ssrc, &found_stream)) {
271 LOG(LS_WARNING) << "Received packet for unknown ssrc: " << header.ssrc;
272 return;
273 }
274
275 // Uncomment this for easy debugging.
276 // LOG(LS_INFO) << "Received packet"
277 // << " groupid=" << found_stream.groupid
278 // << ", ssrc=" << header.ssrc
279 // << ", seqnum=" << header.seq_num
280 // << ", timestamp=" << header.timestamp
281 // << ", len=" << data_len;
282
283 ReceiveDataParams params;
284 params.ssrc = header.ssrc;
285 params.seq_num = header.seq_num;
286 params.timestamp = header.timestamp;
287 SignalDataReceived(params, data, data_len);
288}
289
290bool RtpDataMediaChannel::SetSendBandwidth(bool autobw, int bps) {
291 if (autobw || bps <= 0) {
292 bps = kDataMaxBandwidth;
293 }
294 send_limiter_.reset(new talk_base::RateLimiter(bps / 8, 1.0));
295 LOG(LS_INFO) << "RtpDataMediaChannel::SetSendBandwidth to " << bps << "bps.";
296 return true;
297}
298
299bool RtpDataMediaChannel::SendData(
300 const SendDataParams& params,
301 const talk_base::Buffer& payload,
302 SendDataResult* result) {
303 if (result) {
304 // If we return true, we'll set this to SDR_SUCCESS.
305 *result = SDR_ERROR;
306 }
307 if (!sending_) {
308 LOG(LS_WARNING) << "Not sending packet with ssrc=" << params.ssrc
309 << " len=" << payload.length() << " before SetSend(true).";
310 return false;
311 }
312
313 if (params.type != cricket::DMT_TEXT) {
314 LOG(LS_WARNING) << "Not sending data because binary type is unsupported.";
315 return false;
316 }
317
318 StreamParams found_stream;
319 if (!GetStreamBySsrc(send_streams_, params.ssrc, &found_stream)) {
320 LOG(LS_WARNING) << "Not sending data because ssrc is unknown: "
321 << params.ssrc;
322 return false;
323 }
324
325 DataCodec found_codec;
326 if (!FindCodecByName(send_codecs_, kGoogleRtpDataCodecName, &found_codec)) {
327 LOG(LS_WARNING) << "Not sending data because codec is unknown: "
328 << kGoogleRtpDataCodecName;
329 return false;
330 }
331
332 size_t packet_len = (kMinRtpPacketLen + sizeof(kReservedSpace)
333 + payload.length() + kMaxSrtpHmacOverhead);
334 if (packet_len > kDataMaxRtpPacketLen) {
335 return false;
336 }
337
338 double now = timing_->TimerNow();
339
340 if (!send_limiter_->CanUse(packet_len, now)) {
341 LOG(LS_VERBOSE) << "Dropped data packet of len=" << packet_len
342 << "; already sent " << send_limiter_->used_in_period()
343 << "/" << send_limiter_->max_per_period();
344 return false;
345 } else {
346 LOG(LS_VERBOSE) << "Sending data packet of len=" << packet_len
347 << "; already sent " << send_limiter_->used_in_period()
348 << "/" << send_limiter_->max_per_period();
349 }
350
351 RtpHeader header;
352 header.payload_type = found_codec.id;
353 header.ssrc = params.ssrc;
354 rtp_clock_by_send_ssrc_[header.ssrc]->Tick(
355 now, &header.seq_num, &header.timestamp);
356
357 talk_base::Buffer packet;
358 packet.SetCapacity(packet_len);
359 packet.SetLength(kMinRtpPacketLen);
360 if (!SetRtpHeader(packet.data(), packet.length(), header)) {
361 return false;
362 }
363 packet.AppendData(&kReservedSpace, sizeof(kReservedSpace));
364 packet.AppendData(payload.data(), payload.length());
365
366 // Uncomment this for easy debugging.
367 // LOG(LS_INFO) << "Sent packet: "
368 // << " stream=" << found_stream.id
369 // << ", seqnum=" << header.seq_num
370 // << ", timestamp=" << header.timestamp
371 // << ", len=" << data_len;
372
henrike@webrtc.org1e09a712013-07-26 19:17:59 +0000373 MediaChannel::SendPacket(&packet);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000374 send_limiter_->Use(packet_len, now);
375 if (result) {
376 *result = SDR_SUCCESS;
377 }
378 return true;
379}
380
381} // namespace cricket