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henrike@webrtc.org28e20752013-07-10 00:45:36 +00001// libjingle
2// Copyright 2004 Google Inc.
3//
4// Redistribution and use in source and binary forms, with or without
5// modification, are permitted provided that the following conditions are met:
6//
7// 1. Redistributions of source code must retain the above copyright notice,
8// this list of conditions and the following disclaimer.
9// 2. Redistributions in binary form must reproduce the above copyright notice,
10// this list of conditions and the following disclaimer in the documentation
11// and/or other materials provided with the distribution.
12// 3. The name of the author may not be used to endorse or promote products
13// derived from this software without specific prior written permission.
14//
15// THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
16// WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
17// MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
18// EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
19// SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
20// PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
21// OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
22// WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
23// OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
24// ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
25
26#include "talk/media/base/filemediaengine.h"
27
pbos@webrtc.org371243d2014-03-07 15:22:04 +000028#include <limits.h>
henrike@webrtc.org28e20752013-07-10 00:45:36 +000029
30#include "talk/base/buffer.h"
31#include "talk/base/event.h"
32#include "talk/base/logging.h"
33#include "talk/base/pathutils.h"
34#include "talk/base/stream.h"
35#include "talk/media/base/rtpdump.h"
36#include "talk/media/base/rtputils.h"
37#include "talk/media/base/streamparams.h"
38
39namespace cricket {
40
41///////////////////////////////////////////////////////////////////////////
42// Implementation of FileMediaEngine.
43///////////////////////////////////////////////////////////////////////////
44int FileMediaEngine::GetCapabilities() {
45 int capabilities = 0;
46 if (!voice_input_filename_.empty()) {
47 capabilities |= AUDIO_SEND;
48 }
49 if (!voice_output_filename_.empty()) {
50 capabilities |= AUDIO_RECV;
51 }
52 if (!video_input_filename_.empty()) {
53 capabilities |= VIDEO_SEND;
54 }
55 if (!video_output_filename_.empty()) {
56 capabilities |= VIDEO_RECV;
57 }
58 return capabilities;
59}
60
61VoiceMediaChannel* FileMediaEngine::CreateChannel() {
62 talk_base::FileStream* input_file_stream = NULL;
63 talk_base::FileStream* output_file_stream = NULL;
64
65 if (voice_input_filename_.empty() && voice_output_filename_.empty())
66 return NULL;
67 if (!voice_input_filename_.empty()) {
68 input_file_stream = talk_base::Filesystem::OpenFile(
69 talk_base::Pathname(voice_input_filename_), "rb");
70 if (!input_file_stream) {
71 LOG(LS_ERROR) << "Not able to open the input audio stream file.";
72 return NULL;
73 }
74 }
75
76 if (!voice_output_filename_.empty()) {
77 output_file_stream = talk_base::Filesystem::OpenFile(
78 talk_base::Pathname(voice_output_filename_), "wb");
79 if (!output_file_stream) {
80 delete input_file_stream;
81 LOG(LS_ERROR) << "Not able to open the output audio stream file.";
82 return NULL;
83 }
84 }
85
wu@webrtc.org9caf2762013-12-11 18:25:07 +000086 return new FileVoiceChannel(input_file_stream, output_file_stream,
87 rtp_sender_thread_);
henrike@webrtc.org28e20752013-07-10 00:45:36 +000088}
89
90VideoMediaChannel* FileMediaEngine::CreateVideoChannel(
91 VoiceMediaChannel* voice_ch) {
92 talk_base::FileStream* input_file_stream = NULL;
93 talk_base::FileStream* output_file_stream = NULL;
94
95 if (video_input_filename_.empty() && video_output_filename_.empty())
96 return NULL;
97
98 if (!video_input_filename_.empty()) {
99 input_file_stream = talk_base::Filesystem::OpenFile(
100 talk_base::Pathname(video_input_filename_), "rb");
101 if (!input_file_stream) {
102 LOG(LS_ERROR) << "Not able to open the input video stream file.";
103 return NULL;
104 }
105 }
106
107 if (!video_output_filename_.empty()) {
108 output_file_stream = talk_base::Filesystem::OpenFile(
109 talk_base::Pathname(video_output_filename_), "wb");
110 if (!output_file_stream) {
111 delete input_file_stream;
112 LOG(LS_ERROR) << "Not able to open the output video stream file.";
113 return NULL;
114 }
115 }
116
wu@webrtc.org9caf2762013-12-11 18:25:07 +0000117 return new FileVideoChannel(input_file_stream, output_file_stream,
118 rtp_sender_thread_);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000119}
120
121///////////////////////////////////////////////////////////////////////////
122// Definition of RtpSenderReceiver.
123///////////////////////////////////////////////////////////////////////////
wu@webrtc.org9caf2762013-12-11 18:25:07 +0000124class RtpSenderReceiver : public talk_base::MessageHandler {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000125 public:
126 RtpSenderReceiver(MediaChannel* channel,
127 talk_base::StreamInterface* input_file_stream,
wu@webrtc.org9caf2762013-12-11 18:25:07 +0000128 talk_base::StreamInterface* output_file_stream,
129 talk_base::Thread* sender_thread);
wu@webrtc.org3c5d2b42013-10-18 16:27:26 +0000130 virtual ~RtpSenderReceiver();
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000131
132 // Called by media channel. Context: media channel thread.
133 bool SetSend(bool send);
134 void SetSendSsrc(uint32 ssrc);
135 void OnPacketReceived(talk_base::Buffer* packet);
136
137 // Override virtual method of parent MessageHandler. Context: Worker Thread.
138 virtual void OnMessage(talk_base::Message* pmsg);
139
140 private:
141 // Read the next RTP dump packet, whose RTP SSRC is the same as first_ssrc_.
142 // Return true if successful.
143 bool ReadNextPacket(RtpDumpPacket* packet);
144 // Send a RTP packet to the network. The input parameter data points to the
145 // start of the RTP packet and len is the packet size. Return true if the sent
146 // size is equal to len.
147 bool SendRtpPacket(const void* data, size_t len);
148
149 MediaChannel* media_channel_;
150 talk_base::scoped_ptr<talk_base::StreamInterface> input_stream_;
151 talk_base::scoped_ptr<talk_base::StreamInterface> output_stream_;
152 talk_base::scoped_ptr<RtpDumpLoopReader> rtp_dump_reader_;
153 talk_base::scoped_ptr<RtpDumpWriter> rtp_dump_writer_;
wu@webrtc.org9caf2762013-12-11 18:25:07 +0000154 talk_base::Thread* sender_thread_;
155 bool own_sender_thread_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000156 // RTP dump packet read from the input stream.
157 RtpDumpPacket rtp_dump_packet_;
158 uint32 start_send_time_;
159 bool sending_;
160 bool first_packet_;
161 uint32 first_ssrc_;
162
163 DISALLOW_COPY_AND_ASSIGN(RtpSenderReceiver);
164};
165
166///////////////////////////////////////////////////////////////////////////
167// Implementation of RtpSenderReceiver.
168///////////////////////////////////////////////////////////////////////////
169RtpSenderReceiver::RtpSenderReceiver(
170 MediaChannel* channel,
171 talk_base::StreamInterface* input_file_stream,
wu@webrtc.org9caf2762013-12-11 18:25:07 +0000172 talk_base::StreamInterface* output_file_stream,
173 talk_base::Thread* sender_thread)
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000174 : media_channel_(channel),
wu@webrtc.org9caf2762013-12-11 18:25:07 +0000175 input_stream_(input_file_stream),
176 output_stream_(output_file_stream),
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000177 sending_(false),
178 first_packet_(true) {
wu@webrtc.org9caf2762013-12-11 18:25:07 +0000179 if (sender_thread == NULL) {
180 sender_thread_ = new talk_base::Thread();
181 own_sender_thread_ = true;
182 } else {
183 sender_thread_ = sender_thread;
184 own_sender_thread_ = false;
185 }
186
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000187 if (input_stream_) {
188 rtp_dump_reader_.reset(new RtpDumpLoopReader(input_stream_.get()));
189 // Start the sender thread, which reads rtp dump records, waits based on
190 // the record timestamps, and sends the RTP packets to the network.
wu@webrtc.org9caf2762013-12-11 18:25:07 +0000191 if (own_sender_thread_) {
192 sender_thread_->Start();
193 }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000194 }
195
196 // Create a rtp dump writer for the output RTP dump stream.
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000197 if (output_stream_) {
198 rtp_dump_writer_.reset(new RtpDumpWriter(output_stream_.get()));
199 }
200}
201
wu@webrtc.org3c5d2b42013-10-18 16:27:26 +0000202RtpSenderReceiver::~RtpSenderReceiver() {
wu@webrtc.org9caf2762013-12-11 18:25:07 +0000203 if (own_sender_thread_) {
204 sender_thread_->Stop();
205 delete sender_thread_;
206 }
wu@webrtc.org3c5d2b42013-10-18 16:27:26 +0000207}
208
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000209bool RtpSenderReceiver::SetSend(bool send) {
210 bool was_sending = sending_;
211 sending_ = send;
212 if (!was_sending && sending_) {
wu@webrtc.org9caf2762013-12-11 18:25:07 +0000213 sender_thread_->PostDelayed(0, this); // Wake up the send thread.
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000214 start_send_time_ = talk_base::Time();
215 }
216 return true;
217}
218
219void RtpSenderReceiver::SetSendSsrc(uint32 ssrc) {
220 if (rtp_dump_reader_) {
221 rtp_dump_reader_->SetSsrc(ssrc);
222 }
223}
224
225void RtpSenderReceiver::OnPacketReceived(talk_base::Buffer* packet) {
226 if (rtp_dump_writer_) {
227 rtp_dump_writer_->WriteRtpPacket(packet->data(), packet->length());
228 }
229}
230
231void RtpSenderReceiver::OnMessage(talk_base::Message* pmsg) {
232 if (!sending_) {
233 // If the sender thread is not sending, ignore this message. The thread goes
234 // to sleep until SetSend(true) wakes it up.
235 return;
236 }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000237 if (!first_packet_) {
238 // Send the previously read packet.
239 SendRtpPacket(&rtp_dump_packet_.data[0], rtp_dump_packet_.data.size());
240 }
241
242 if (ReadNextPacket(&rtp_dump_packet_)) {
243 int wait = talk_base::TimeUntil(
244 start_send_time_ + rtp_dump_packet_.elapsed_time);
245 wait = talk_base::_max(0, wait);
wu@webrtc.org9caf2762013-12-11 18:25:07 +0000246 sender_thread_->PostDelayed(wait, this);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000247 } else {
wu@webrtc.org9caf2762013-12-11 18:25:07 +0000248 sender_thread_->Quit();
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000249 }
250}
251
252bool RtpSenderReceiver::ReadNextPacket(RtpDumpPacket* packet) {
253 while (talk_base::SR_SUCCESS == rtp_dump_reader_->ReadPacket(packet)) {
254 uint32 ssrc;
255 if (!packet->GetRtpSsrc(&ssrc)) {
256 return false;
257 }
258 if (first_packet_) {
259 first_packet_ = false;
260 first_ssrc_ = ssrc;
261 }
262 if (ssrc == first_ssrc_) {
263 return true;
264 }
265 }
266 return false;
267}
268
269bool RtpSenderReceiver::SendRtpPacket(const void* data, size_t len) {
henrike@webrtc.org1e09a712013-07-26 19:17:59 +0000270 if (!media_channel_)
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000271 return false;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000272
273 talk_base::Buffer packet(data, len, kMaxRtpPacketLen);
henrike@webrtc.org1e09a712013-07-26 19:17:59 +0000274 return media_channel_->SendPacket(&packet);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000275}
276
277///////////////////////////////////////////////////////////////////////////
278// Implementation of FileVoiceChannel.
279///////////////////////////////////////////////////////////////////////////
280FileVoiceChannel::FileVoiceChannel(
281 talk_base::StreamInterface* input_file_stream,
wu@webrtc.org9caf2762013-12-11 18:25:07 +0000282 talk_base::StreamInterface* output_file_stream,
283 talk_base::Thread* rtp_sender_thread)
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000284 : send_ssrc_(0),
285 rtp_sender_receiver_(new RtpSenderReceiver(this, input_file_stream,
wu@webrtc.org9caf2762013-12-11 18:25:07 +0000286 output_file_stream,
287 rtp_sender_thread)) {}
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000288
289FileVoiceChannel::~FileVoiceChannel() {}
290
291bool FileVoiceChannel::SetSendCodecs(const std::vector<AudioCodec>& codecs) {
292 // TODO(whyuan): Check the format of RTP dump input.
293 return true;
294}
295
296bool FileVoiceChannel::SetSend(SendFlags flag) {
297 return rtp_sender_receiver_->SetSend(flag != SEND_NOTHING);
298}
299
300bool FileVoiceChannel::AddSendStream(const StreamParams& sp) {
301 if (send_ssrc_ != 0 || sp.ssrcs.size() != 1) {
302 LOG(LS_ERROR) << "FileVoiceChannel only supports one send stream.";
303 return false;
304 }
305 send_ssrc_ = sp.ssrcs[0];
306 rtp_sender_receiver_->SetSendSsrc(send_ssrc_);
307 return true;
308}
309
310bool FileVoiceChannel::RemoveSendStream(uint32 ssrc) {
311 if (ssrc != send_ssrc_)
312 return false;
313 send_ssrc_ = 0;
314 rtp_sender_receiver_->SetSendSsrc(send_ssrc_);
315 return true;
316}
317
wu@webrtc.orga9890802013-12-13 00:21:03 +0000318void FileVoiceChannel::OnPacketReceived(
319 talk_base::Buffer* packet, const talk_base::PacketTime& packet_time) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000320 rtp_sender_receiver_->OnPacketReceived(packet);
321}
322
323///////////////////////////////////////////////////////////////////////////
324// Implementation of FileVideoChannel.
325///////////////////////////////////////////////////////////////////////////
326FileVideoChannel::FileVideoChannel(
327 talk_base::StreamInterface* input_file_stream,
wu@webrtc.org9caf2762013-12-11 18:25:07 +0000328 talk_base::StreamInterface* output_file_stream,
329 talk_base::Thread* rtp_sender_thread)
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000330 : send_ssrc_(0),
331 rtp_sender_receiver_(new RtpSenderReceiver(this, input_file_stream,
wu@webrtc.org9caf2762013-12-11 18:25:07 +0000332 output_file_stream,
333 rtp_sender_thread)) {}
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000334
335FileVideoChannel::~FileVideoChannel() {}
336
337bool FileVideoChannel::SetSendCodecs(const std::vector<VideoCodec>& codecs) {
338 // TODO(whyuan): Check the format of RTP dump input.
339 return true;
340}
341
342bool FileVideoChannel::SetSend(bool send) {
343 return rtp_sender_receiver_->SetSend(send);
344}
345
346bool FileVideoChannel::AddSendStream(const StreamParams& sp) {
347 if (send_ssrc_ != 0 || sp.ssrcs.size() != 1) {
348 LOG(LS_ERROR) << "FileVideoChannel only support one send stream.";
349 return false;
350 }
351 send_ssrc_ = sp.ssrcs[0];
352 rtp_sender_receiver_->SetSendSsrc(send_ssrc_);
353 return true;
354}
355
356bool FileVideoChannel::RemoveSendStream(uint32 ssrc) {
357 if (ssrc != send_ssrc_)
358 return false;
359 send_ssrc_ = 0;
360 rtp_sender_receiver_->SetSendSsrc(send_ssrc_);
361 return true;
362}
363
wu@webrtc.orga9890802013-12-13 00:21:03 +0000364void FileVideoChannel::OnPacketReceived(
365 talk_base::Buffer* packet, const talk_base::PacketTime& packet_time) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000366 rtp_sender_receiver_->OnPacketReceived(packet);
367}
368
369} // namespace cricket