1// 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
28#include <limits.h>
29
30#include "talk/media/base/rtpdump.h"
31#include "talk/media/base/rtputils.h"
32#include "talk/media/base/streamparams.h"
33#include "webrtc/base/buffer.h"
34#include "webrtc/base/event.h"
35#include "webrtc/base/logging.h"
36#include "webrtc/base/pathutils.h"
37#include "webrtc/base/stream.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  rtc::FileStream* input_file_stream = NULL;
63  rtc::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 = rtc::Filesystem::OpenFile(
69        rtc::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 = rtc::Filesystem::OpenFile(
78        rtc::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
86  return new FileVoiceChannel(input_file_stream, output_file_stream,
87                              rtp_sender_thread_);
88}
89
90VideoMediaChannel* FileMediaEngine::CreateVideoChannel(
91    VoiceMediaChannel* voice_ch) {
92  rtc::FileStream* input_file_stream = NULL;
93  rtc::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 = rtc::Filesystem::OpenFile(
100        rtc::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 = rtc::Filesystem::OpenFile(
109        rtc::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
117  return new FileVideoChannel(input_file_stream, output_file_stream,
118                              rtp_sender_thread_);
119}
120
121///////////////////////////////////////////////////////////////////////////
122// Definition of RtpSenderReceiver.
123///////////////////////////////////////////////////////////////////////////
124class RtpSenderReceiver : public rtc::MessageHandler {
125 public:
126  RtpSenderReceiver(MediaChannel* channel,
127                    rtc::StreamInterface* input_file_stream,
128                    rtc::StreamInterface* output_file_stream,
129                    rtc::Thread* sender_thread);
130  virtual ~RtpSenderReceiver();
131
132  // Called by media channel. Context: media channel thread.
133  bool SetSend(bool send);
134  void SetSendSsrc(uint32 ssrc);
135  void OnPacketReceived(rtc::Buffer* packet);
136
137  // Override virtual method of parent MessageHandler. Context: Worker Thread.
138  virtual void OnMessage(rtc::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  rtc::scoped_ptr<rtc::StreamInterface> input_stream_;
151  rtc::scoped_ptr<rtc::StreamInterface> output_stream_;
152  rtc::scoped_ptr<RtpDumpLoopReader> rtp_dump_reader_;
153  rtc::scoped_ptr<RtpDumpWriter> rtp_dump_writer_;
154  rtc::Thread* sender_thread_;
155  bool own_sender_thread_;
156  // 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    rtc::StreamInterface* input_file_stream,
172    rtc::StreamInterface* output_file_stream,
173    rtc::Thread* sender_thread)
174    : media_channel_(channel),
175      input_stream_(input_file_stream),
176      output_stream_(output_file_stream),
177      sending_(false),
178      first_packet_(true) {
179  if (sender_thread == NULL) {
180    sender_thread_ = new rtc::Thread();
181    own_sender_thread_ = true;
182  } else {
183    sender_thread_ = sender_thread;
184    own_sender_thread_ = false;
185  }
186
187  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.
191    if (own_sender_thread_) {
192      sender_thread_->Start();
193    }
194  }
195
196  // Create a rtp dump writer for the output RTP dump stream.
197  if (output_stream_) {
198    rtp_dump_writer_.reset(new RtpDumpWriter(output_stream_.get()));
199  }
200}
201
202RtpSenderReceiver::~RtpSenderReceiver() {
203  if (own_sender_thread_) {
204    sender_thread_->Stop();
205    delete sender_thread_;
206  }
207}
208
209bool RtpSenderReceiver::SetSend(bool send) {
210  bool was_sending = sending_;
211  sending_ = send;
212  if (!was_sending && sending_) {
213    sender_thread_->PostDelayed(0, this);  // Wake up the send thread.
214    start_send_time_ = rtc::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(rtc::Buffer* packet) {
226  if (rtp_dump_writer_) {
227    rtp_dump_writer_->WriteRtpPacket(packet->data(), packet->length());
228  }
229}
230
231void RtpSenderReceiver::OnMessage(rtc::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  }
237  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 = rtc::TimeUntil(
244        start_send_time_ + rtp_dump_packet_.elapsed_time);
245    wait = rtc::_max(0, wait);
246    sender_thread_->PostDelayed(wait, this);
247  } else {
248    sender_thread_->Quit();
249  }
250}
251
252bool RtpSenderReceiver::ReadNextPacket(RtpDumpPacket* packet) {
253  while (rtc::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) {
270  if (!media_channel_)
271    return false;
272
273  rtc::Buffer packet(data, len, kMaxRtpPacketLen);
274  return media_channel_->SendPacket(&packet);
275}
276
277///////////////////////////////////////////////////////////////////////////
278// Implementation of FileVoiceChannel.
279///////////////////////////////////////////////////////////////////////////
280FileVoiceChannel::FileVoiceChannel(
281    rtc::StreamInterface* input_file_stream,
282    rtc::StreamInterface* output_file_stream,
283    rtc::Thread* rtp_sender_thread)
284    : send_ssrc_(0),
285      rtp_sender_receiver_(new RtpSenderReceiver(this, input_file_stream,
286                                                 output_file_stream,
287                                                 rtp_sender_thread)) {}
288
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
318void FileVoiceChannel::OnPacketReceived(
319    rtc::Buffer* packet, const rtc::PacketTime& packet_time) {
320  rtp_sender_receiver_->OnPacketReceived(packet);
321}
322
323///////////////////////////////////////////////////////////////////////////
324// Implementation of FileVideoChannel.
325///////////////////////////////////////////////////////////////////////////
326FileVideoChannel::FileVideoChannel(
327    rtc::StreamInterface* input_file_stream,
328    rtc::StreamInterface* output_file_stream,
329    rtc::Thread* rtp_sender_thread)
330    : send_ssrc_(0),
331      rtp_sender_receiver_(new RtpSenderReceiver(this, input_file_stream,
332                                                 output_file_stream,
333                                                 rtp_sender_thread)) {}
334
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
364void FileVideoChannel::OnPacketReceived(
365    rtc::Buffer* packet, const rtc::PacketTime& packet_time) {
366  rtp_sender_receiver_->OnPacketReceived(packet);
367}
368
369}  // namespace cricket
370