a9890800e0
https://webrtc-codereview.appspot.com/5309005/. R=mallinath@webrtc.org, niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/5719004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5277 4adac7df-926f-26a2-2b94-8c16560cd09d
370 lines
12 KiB
C++
370 lines
12 KiB
C++
// libjingle
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// Copyright 2004 Google Inc.
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//
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// Redistribution and use in source and binary forms, with or without
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// modification, are permitted provided that the following conditions are met:
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//
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// 1. Redistributions of source code must retain the above copyright notice,
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// this list of conditions and the following disclaimer.
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// 2. Redistributions in binary form must reproduce the above copyright notice,
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// this list of conditions and the following disclaimer in the documentation
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// and/or other materials provided with the distribution.
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// 3. The name of the author may not be used to endorse or promote products
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// derived from this software without specific prior written permission.
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//
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// THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
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// WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
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// MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
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// EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
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// SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
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// PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
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// OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
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// WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
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// OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
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// ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
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#include "talk/media/base/filemediaengine.h"
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#include <climits>
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#include "talk/base/buffer.h"
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#include "talk/base/event.h"
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#include "talk/base/logging.h"
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#include "talk/base/pathutils.h"
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#include "talk/base/stream.h"
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#include "talk/media/base/rtpdump.h"
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#include "talk/media/base/rtputils.h"
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#include "talk/media/base/streamparams.h"
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namespace cricket {
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///////////////////////////////////////////////////////////////////////////
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// Implementation of FileMediaEngine.
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///////////////////////////////////////////////////////////////////////////
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int FileMediaEngine::GetCapabilities() {
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int capabilities = 0;
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if (!voice_input_filename_.empty()) {
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capabilities |= AUDIO_SEND;
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}
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if (!voice_output_filename_.empty()) {
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capabilities |= AUDIO_RECV;
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}
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if (!video_input_filename_.empty()) {
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capabilities |= VIDEO_SEND;
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}
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if (!video_output_filename_.empty()) {
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capabilities |= VIDEO_RECV;
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}
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return capabilities;
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}
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VoiceMediaChannel* FileMediaEngine::CreateChannel() {
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talk_base::FileStream* input_file_stream = NULL;
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talk_base::FileStream* output_file_stream = NULL;
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if (voice_input_filename_.empty() && voice_output_filename_.empty())
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return NULL;
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if (!voice_input_filename_.empty()) {
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input_file_stream = talk_base::Filesystem::OpenFile(
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talk_base::Pathname(voice_input_filename_), "rb");
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if (!input_file_stream) {
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LOG(LS_ERROR) << "Not able to open the input audio stream file.";
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return NULL;
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}
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}
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if (!voice_output_filename_.empty()) {
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output_file_stream = talk_base::Filesystem::OpenFile(
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talk_base::Pathname(voice_output_filename_), "wb");
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if (!output_file_stream) {
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delete input_file_stream;
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LOG(LS_ERROR) << "Not able to open the output audio stream file.";
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return NULL;
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}
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}
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return new FileVoiceChannel(input_file_stream, output_file_stream,
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rtp_sender_thread_);
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}
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VideoMediaChannel* FileMediaEngine::CreateVideoChannel(
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VoiceMediaChannel* voice_ch) {
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talk_base::FileStream* input_file_stream = NULL;
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talk_base::FileStream* output_file_stream = NULL;
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if (video_input_filename_.empty() && video_output_filename_.empty())
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return NULL;
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if (!video_input_filename_.empty()) {
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input_file_stream = talk_base::Filesystem::OpenFile(
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talk_base::Pathname(video_input_filename_), "rb");
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if (!input_file_stream) {
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LOG(LS_ERROR) << "Not able to open the input video stream file.";
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return NULL;
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}
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}
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if (!video_output_filename_.empty()) {
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output_file_stream = talk_base::Filesystem::OpenFile(
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talk_base::Pathname(video_output_filename_), "wb");
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if (!output_file_stream) {
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delete input_file_stream;
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LOG(LS_ERROR) << "Not able to open the output video stream file.";
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return NULL;
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}
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}
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return new FileVideoChannel(input_file_stream, output_file_stream,
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rtp_sender_thread_);
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}
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///////////////////////////////////////////////////////////////////////////
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// Definition of RtpSenderReceiver.
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///////////////////////////////////////////////////////////////////////////
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class RtpSenderReceiver : public talk_base::MessageHandler {
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public:
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RtpSenderReceiver(MediaChannel* channel,
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talk_base::StreamInterface* input_file_stream,
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talk_base::StreamInterface* output_file_stream,
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talk_base::Thread* sender_thread);
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virtual ~RtpSenderReceiver();
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// Called by media channel. Context: media channel thread.
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bool SetSend(bool send);
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void SetSendSsrc(uint32 ssrc);
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void OnPacketReceived(talk_base::Buffer* packet);
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// Override virtual method of parent MessageHandler. Context: Worker Thread.
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virtual void OnMessage(talk_base::Message* pmsg);
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private:
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// Read the next RTP dump packet, whose RTP SSRC is the same as first_ssrc_.
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// Return true if successful.
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bool ReadNextPacket(RtpDumpPacket* packet);
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// Send a RTP packet to the network. The input parameter data points to the
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// start of the RTP packet and len is the packet size. Return true if the sent
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// size is equal to len.
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bool SendRtpPacket(const void* data, size_t len);
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MediaChannel* media_channel_;
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talk_base::scoped_ptr<talk_base::StreamInterface> input_stream_;
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talk_base::scoped_ptr<talk_base::StreamInterface> output_stream_;
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talk_base::scoped_ptr<RtpDumpLoopReader> rtp_dump_reader_;
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talk_base::scoped_ptr<RtpDumpWriter> rtp_dump_writer_;
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talk_base::Thread* sender_thread_;
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bool own_sender_thread_;
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// RTP dump packet read from the input stream.
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RtpDumpPacket rtp_dump_packet_;
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uint32 start_send_time_;
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bool sending_;
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bool first_packet_;
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uint32 first_ssrc_;
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DISALLOW_COPY_AND_ASSIGN(RtpSenderReceiver);
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};
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///////////////////////////////////////////////////////////////////////////
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// Implementation of RtpSenderReceiver.
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///////////////////////////////////////////////////////////////////////////
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RtpSenderReceiver::RtpSenderReceiver(
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MediaChannel* channel,
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talk_base::StreamInterface* input_file_stream,
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talk_base::StreamInterface* output_file_stream,
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talk_base::Thread* sender_thread)
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: media_channel_(channel),
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input_stream_(input_file_stream),
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output_stream_(output_file_stream),
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sending_(false),
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first_packet_(true) {
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if (sender_thread == NULL) {
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sender_thread_ = new talk_base::Thread();
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own_sender_thread_ = true;
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} else {
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sender_thread_ = sender_thread;
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own_sender_thread_ = false;
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}
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if (input_stream_) {
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rtp_dump_reader_.reset(new RtpDumpLoopReader(input_stream_.get()));
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// Start the sender thread, which reads rtp dump records, waits based on
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// the record timestamps, and sends the RTP packets to the network.
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if (own_sender_thread_) {
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sender_thread_->Start();
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}
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}
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// Create a rtp dump writer for the output RTP dump stream.
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if (output_stream_) {
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rtp_dump_writer_.reset(new RtpDumpWriter(output_stream_.get()));
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}
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}
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RtpSenderReceiver::~RtpSenderReceiver() {
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if (own_sender_thread_) {
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sender_thread_->Stop();
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delete sender_thread_;
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}
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}
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bool RtpSenderReceiver::SetSend(bool send) {
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bool was_sending = sending_;
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sending_ = send;
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if (!was_sending && sending_) {
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sender_thread_->PostDelayed(0, this); // Wake up the send thread.
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start_send_time_ = talk_base::Time();
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}
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return true;
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}
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void RtpSenderReceiver::SetSendSsrc(uint32 ssrc) {
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if (rtp_dump_reader_) {
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rtp_dump_reader_->SetSsrc(ssrc);
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}
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}
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void RtpSenderReceiver::OnPacketReceived(talk_base::Buffer* packet) {
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if (rtp_dump_writer_) {
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rtp_dump_writer_->WriteRtpPacket(packet->data(), packet->length());
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}
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}
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void RtpSenderReceiver::OnMessage(talk_base::Message* pmsg) {
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if (!sending_) {
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// If the sender thread is not sending, ignore this message. The thread goes
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// to sleep until SetSend(true) wakes it up.
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return;
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}
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if (!first_packet_) {
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// Send the previously read packet.
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SendRtpPacket(&rtp_dump_packet_.data[0], rtp_dump_packet_.data.size());
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}
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if (ReadNextPacket(&rtp_dump_packet_)) {
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int wait = talk_base::TimeUntil(
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start_send_time_ + rtp_dump_packet_.elapsed_time);
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wait = talk_base::_max(0, wait);
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sender_thread_->PostDelayed(wait, this);
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} else {
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sender_thread_->Quit();
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}
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}
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bool RtpSenderReceiver::ReadNextPacket(RtpDumpPacket* packet) {
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while (talk_base::SR_SUCCESS == rtp_dump_reader_->ReadPacket(packet)) {
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uint32 ssrc;
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if (!packet->GetRtpSsrc(&ssrc)) {
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return false;
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}
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if (first_packet_) {
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first_packet_ = false;
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first_ssrc_ = ssrc;
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}
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if (ssrc == first_ssrc_) {
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return true;
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}
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}
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return false;
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}
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bool RtpSenderReceiver::SendRtpPacket(const void* data, size_t len) {
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if (!media_channel_)
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return false;
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talk_base::Buffer packet(data, len, kMaxRtpPacketLen);
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return media_channel_->SendPacket(&packet);
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}
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///////////////////////////////////////////////////////////////////////////
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// Implementation of FileVoiceChannel.
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///////////////////////////////////////////////////////////////////////////
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FileVoiceChannel::FileVoiceChannel(
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talk_base::StreamInterface* input_file_stream,
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talk_base::StreamInterface* output_file_stream,
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talk_base::Thread* rtp_sender_thread)
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: send_ssrc_(0),
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rtp_sender_receiver_(new RtpSenderReceiver(this, input_file_stream,
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output_file_stream,
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rtp_sender_thread)) {}
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FileVoiceChannel::~FileVoiceChannel() {}
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bool FileVoiceChannel::SetSendCodecs(const std::vector<AudioCodec>& codecs) {
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// TODO(whyuan): Check the format of RTP dump input.
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return true;
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}
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bool FileVoiceChannel::SetSend(SendFlags flag) {
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return rtp_sender_receiver_->SetSend(flag != SEND_NOTHING);
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}
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bool FileVoiceChannel::AddSendStream(const StreamParams& sp) {
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if (send_ssrc_ != 0 || sp.ssrcs.size() != 1) {
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LOG(LS_ERROR) << "FileVoiceChannel only supports one send stream.";
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return false;
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}
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send_ssrc_ = sp.ssrcs[0];
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rtp_sender_receiver_->SetSendSsrc(send_ssrc_);
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return true;
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}
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bool FileVoiceChannel::RemoveSendStream(uint32 ssrc) {
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if (ssrc != send_ssrc_)
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return false;
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send_ssrc_ = 0;
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rtp_sender_receiver_->SetSendSsrc(send_ssrc_);
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return true;
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}
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void FileVoiceChannel::OnPacketReceived(
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talk_base::Buffer* packet, const talk_base::PacketTime& packet_time) {
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rtp_sender_receiver_->OnPacketReceived(packet);
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}
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///////////////////////////////////////////////////////////////////////////
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// Implementation of FileVideoChannel.
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///////////////////////////////////////////////////////////////////////////
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FileVideoChannel::FileVideoChannel(
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talk_base::StreamInterface* input_file_stream,
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talk_base::StreamInterface* output_file_stream,
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talk_base::Thread* rtp_sender_thread)
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: send_ssrc_(0),
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rtp_sender_receiver_(new RtpSenderReceiver(this, input_file_stream,
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output_file_stream,
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rtp_sender_thread)) {}
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FileVideoChannel::~FileVideoChannel() {}
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bool FileVideoChannel::SetSendCodecs(const std::vector<VideoCodec>& codecs) {
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// TODO(whyuan): Check the format of RTP dump input.
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return true;
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}
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bool FileVideoChannel::SetSend(bool send) {
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return rtp_sender_receiver_->SetSend(send);
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}
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bool FileVideoChannel::AddSendStream(const StreamParams& sp) {
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if (send_ssrc_ != 0 || sp.ssrcs.size() != 1) {
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LOG(LS_ERROR) << "FileVideoChannel only support one send stream.";
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return false;
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}
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send_ssrc_ = sp.ssrcs[0];
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rtp_sender_receiver_->SetSendSsrc(send_ssrc_);
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return true;
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}
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bool FileVideoChannel::RemoveSendStream(uint32 ssrc) {
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if (ssrc != send_ssrc_)
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return false;
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send_ssrc_ = 0;
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rtp_sender_receiver_->SetSendSsrc(send_ssrc_);
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return true;
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}
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void FileVideoChannel::OnPacketReceived(
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talk_base::Buffer* packet, const talk_base::PacketTime& packet_time) {
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rtp_sender_receiver_->OnPacketReceived(packet);
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}
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} // namespace cricket
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