webrtc/talk/media/base/filemediaengine.cc

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// libjingle
// Copyright 2004 Google Inc.
//
// Redistribution and use in source and binary forms, with or without
// modification, are permitted provided that the following conditions are met:
//
// 1. Redistributions of source code must retain the above copyright notice,
// this list of conditions and the following disclaimer.
// 2. Redistributions in binary form must reproduce the above copyright notice,
// this list of conditions and the following disclaimer in the documentation
// and/or other materials provided with the distribution.
// 3. The name of the author may not be used to endorse or promote products
// derived from this software without specific prior written permission.
//
// THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
// WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
// MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
// EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
// SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
// PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
// OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
// WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
// OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
// ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
#include "talk/media/base/filemediaengine.h"
#include <limits.h>
#include "talk/media/base/rtpdump.h"
#include "talk/media/base/rtputils.h"
#include "talk/media/base/streamparams.h"
#include "webrtc/base/buffer.h"
#include "webrtc/base/event.h"
#include "webrtc/base/logging.h"
#include "webrtc/base/pathutils.h"
#include "webrtc/base/stream.h"
namespace cricket {
///////////////////////////////////////////////////////////////////////////
// Implementation of FileMediaEngine.
///////////////////////////////////////////////////////////////////////////
int FileMediaEngine::GetCapabilities() {
int capabilities = 0;
if (!voice_input_filename_.empty()) {
capabilities |= AUDIO_SEND;
}
if (!voice_output_filename_.empty()) {
capabilities |= AUDIO_RECV;
}
if (!video_input_filename_.empty()) {
capabilities |= VIDEO_SEND;
}
if (!video_output_filename_.empty()) {
capabilities |= VIDEO_RECV;
}
return capabilities;
}
VoiceMediaChannel* FileMediaEngine::CreateChannel() {
rtc::FileStream* input_file_stream = NULL;
rtc::FileStream* output_file_stream = NULL;
if (voice_input_filename_.empty() && voice_output_filename_.empty())
return NULL;
if (!voice_input_filename_.empty()) {
input_file_stream = rtc::Filesystem::OpenFile(
rtc::Pathname(voice_input_filename_), "rb");
if (!input_file_stream) {
LOG(LS_ERROR) << "Not able to open the input audio stream file.";
return NULL;
}
}
if (!voice_output_filename_.empty()) {
output_file_stream = rtc::Filesystem::OpenFile(
rtc::Pathname(voice_output_filename_), "wb");
if (!output_file_stream) {
delete input_file_stream;
LOG(LS_ERROR) << "Not able to open the output audio stream file.";
return NULL;
}
}
return new FileVoiceChannel(input_file_stream, output_file_stream,
rtp_sender_thread_);
}
VideoMediaChannel* FileMediaEngine::CreateVideoChannel(
VoiceMediaChannel* voice_ch) {
rtc::FileStream* input_file_stream = NULL;
rtc::FileStream* output_file_stream = NULL;
if (video_input_filename_.empty() && video_output_filename_.empty())
return NULL;
if (!video_input_filename_.empty()) {
input_file_stream = rtc::Filesystem::OpenFile(
rtc::Pathname(video_input_filename_), "rb");
if (!input_file_stream) {
LOG(LS_ERROR) << "Not able to open the input video stream file.";
return NULL;
}
}
if (!video_output_filename_.empty()) {
output_file_stream = rtc::Filesystem::OpenFile(
rtc::Pathname(video_output_filename_), "wb");
if (!output_file_stream) {
delete input_file_stream;
LOG(LS_ERROR) << "Not able to open the output video stream file.";
return NULL;
}
}
return new FileVideoChannel(input_file_stream, output_file_stream,
rtp_sender_thread_);
}
///////////////////////////////////////////////////////////////////////////
// Definition of RtpSenderReceiver.
///////////////////////////////////////////////////////////////////////////
class RtpSenderReceiver : public rtc::MessageHandler {
public:
RtpSenderReceiver(MediaChannel* channel,
rtc::StreamInterface* input_file_stream,
rtc::StreamInterface* output_file_stream,
rtc::Thread* sender_thread);
virtual ~RtpSenderReceiver();
// Called by media channel. Context: media channel thread.
bool SetSend(bool send);
void SetSendSsrc(uint32 ssrc);
void OnPacketReceived(rtc::Buffer* packet);
// Override virtual method of parent MessageHandler. Context: Worker Thread.
virtual void OnMessage(rtc::Message* pmsg);
private:
// Read the next RTP dump packet, whose RTP SSRC is the same as first_ssrc_.
// Return true if successful.
bool ReadNextPacket(RtpDumpPacket* packet);
// Send a RTP packet to the network. The input parameter data points to the
// start of the RTP packet and len is the packet size. Return true if the sent
// size is equal to len.
bool SendRtpPacket(const void* data, size_t len);
MediaChannel* media_channel_;
rtc::scoped_ptr<rtc::StreamInterface> input_stream_;
rtc::scoped_ptr<rtc::StreamInterface> output_stream_;
rtc::scoped_ptr<RtpDumpLoopReader> rtp_dump_reader_;
rtc::scoped_ptr<RtpDumpWriter> rtp_dump_writer_;
rtc::Thread* sender_thread_;
bool own_sender_thread_;
// RTP dump packet read from the input stream.
RtpDumpPacket rtp_dump_packet_;
uint32 start_send_time_;
bool sending_;
bool first_packet_;
uint32 first_ssrc_;
DISALLOW_COPY_AND_ASSIGN(RtpSenderReceiver);
};
///////////////////////////////////////////////////////////////////////////
// Implementation of RtpSenderReceiver.
///////////////////////////////////////////////////////////////////////////
RtpSenderReceiver::RtpSenderReceiver(
MediaChannel* channel,
rtc::StreamInterface* input_file_stream,
rtc::StreamInterface* output_file_stream,
rtc::Thread* sender_thread)
: media_channel_(channel),
input_stream_(input_file_stream),
output_stream_(output_file_stream),
sending_(false),
first_packet_(true) {
if (sender_thread == NULL) {
sender_thread_ = new rtc::Thread();
own_sender_thread_ = true;
} else {
sender_thread_ = sender_thread;
own_sender_thread_ = false;
}
if (input_stream_) {
rtp_dump_reader_.reset(new RtpDumpLoopReader(input_stream_.get()));
// Start the sender thread, which reads rtp dump records, waits based on
// the record timestamps, and sends the RTP packets to the network.
if (own_sender_thread_) {
sender_thread_->Start();
}
}
// Create a rtp dump writer for the output RTP dump stream.
if (output_stream_) {
rtp_dump_writer_.reset(new RtpDumpWriter(output_stream_.get()));
}
}
RtpSenderReceiver::~RtpSenderReceiver() {
if (own_sender_thread_) {
sender_thread_->Stop();
delete sender_thread_;
}
}
bool RtpSenderReceiver::SetSend(bool send) {
bool was_sending = sending_;
sending_ = send;
if (!was_sending && sending_) {
sender_thread_->PostDelayed(0, this); // Wake up the send thread.
start_send_time_ = rtc::Time();
}
return true;
}
void RtpSenderReceiver::SetSendSsrc(uint32 ssrc) {
if (rtp_dump_reader_) {
rtp_dump_reader_->SetSsrc(ssrc);
}
}
void RtpSenderReceiver::OnPacketReceived(rtc::Buffer* packet) {
if (rtp_dump_writer_) {
rtp_dump_writer_->WriteRtpPacket(packet->data(), packet->length());
}
}
void RtpSenderReceiver::OnMessage(rtc::Message* pmsg) {
if (!sending_) {
// If the sender thread is not sending, ignore this message. The thread goes
// to sleep until SetSend(true) wakes it up.
return;
}
if (!first_packet_) {
// Send the previously read packet.
SendRtpPacket(&rtp_dump_packet_.data[0], rtp_dump_packet_.data.size());
}
if (ReadNextPacket(&rtp_dump_packet_)) {
int wait = rtc::TimeUntil(
start_send_time_ + rtp_dump_packet_.elapsed_time);
wait = rtc::_max(0, wait);
sender_thread_->PostDelayed(wait, this);
} else {
sender_thread_->Quit();
}
}
bool RtpSenderReceiver::ReadNextPacket(RtpDumpPacket* packet) {
while (rtc::SR_SUCCESS == rtp_dump_reader_->ReadPacket(packet)) {
uint32 ssrc;
if (!packet->GetRtpSsrc(&ssrc)) {
return false;
}
if (first_packet_) {
first_packet_ = false;
first_ssrc_ = ssrc;
}
if (ssrc == first_ssrc_) {
return true;
}
}
return false;
}
bool RtpSenderReceiver::SendRtpPacket(const void* data, size_t len) {
if (!media_channel_)
return false;
rtc::Buffer packet(data, len, kMaxRtpPacketLen);
return media_channel_->SendPacket(&packet);
}
///////////////////////////////////////////////////////////////////////////
// Implementation of FileVoiceChannel.
///////////////////////////////////////////////////////////////////////////
FileVoiceChannel::FileVoiceChannel(
rtc::StreamInterface* input_file_stream,
rtc::StreamInterface* output_file_stream,
rtc::Thread* rtp_sender_thread)
: send_ssrc_(0),
rtp_sender_receiver_(new RtpSenderReceiver(this, input_file_stream,
output_file_stream,
rtp_sender_thread)) {}
FileVoiceChannel::~FileVoiceChannel() {}
bool FileVoiceChannel::SetSendCodecs(const std::vector<AudioCodec>& codecs) {
// TODO(whyuan): Check the format of RTP dump input.
return true;
}
bool FileVoiceChannel::SetSend(SendFlags flag) {
return rtp_sender_receiver_->SetSend(flag != SEND_NOTHING);
}
bool FileVoiceChannel::AddSendStream(const StreamParams& sp) {
if (send_ssrc_ != 0 || sp.ssrcs.size() != 1) {
LOG(LS_ERROR) << "FileVoiceChannel only supports one send stream.";
return false;
}
send_ssrc_ = sp.ssrcs[0];
rtp_sender_receiver_->SetSendSsrc(send_ssrc_);
return true;
}
bool FileVoiceChannel::RemoveSendStream(uint32 ssrc) {
if (ssrc != send_ssrc_)
return false;
send_ssrc_ = 0;
rtp_sender_receiver_->SetSendSsrc(send_ssrc_);
return true;
}
void FileVoiceChannel::OnPacketReceived(
rtc::Buffer* packet, const rtc::PacketTime& packet_time) {
rtp_sender_receiver_->OnPacketReceived(packet);
}
///////////////////////////////////////////////////////////////////////////
// Implementation of FileVideoChannel.
///////////////////////////////////////////////////////////////////////////
FileVideoChannel::FileVideoChannel(
rtc::StreamInterface* input_file_stream,
rtc::StreamInterface* output_file_stream,
rtc::Thread* rtp_sender_thread)
: send_ssrc_(0),
rtp_sender_receiver_(new RtpSenderReceiver(this, input_file_stream,
output_file_stream,
rtp_sender_thread)) {}
FileVideoChannel::~FileVideoChannel() {}
bool FileVideoChannel::SetSendCodecs(const std::vector<VideoCodec>& codecs) {
// TODO(whyuan): Check the format of RTP dump input.
return true;
}
bool FileVideoChannel::SetSend(bool send) {
return rtp_sender_receiver_->SetSend(send);
}
bool FileVideoChannel::AddSendStream(const StreamParams& sp) {
if (send_ssrc_ != 0 || sp.ssrcs.size() != 1) {
LOG(LS_ERROR) << "FileVideoChannel only support one send stream.";
return false;
}
send_ssrc_ = sp.ssrcs[0];
rtp_sender_receiver_->SetSendSsrc(send_ssrc_);
return true;
}
bool FileVideoChannel::RemoveSendStream(uint32 ssrc) {
if (ssrc != send_ssrc_)
return false;
send_ssrc_ = 0;
rtp_sender_receiver_->SetSendSsrc(send_ssrc_);
return true;
}
void FileVideoChannel::OnPacketReceived(
rtc::Buffer* packet, const rtc::PacketTime& packet_time) {
rtp_sender_receiver_->OnPacketReceived(packet);
}
} // namespace cricket