Move talk/media to webrtc/media
I removed the 'libjingle' target in talk/libjingle.gyp and replaced
all users of it with base/base.gyp:rtc_base. It seems the jsoncpp
and expat dependencies were not used by it's previous references.
The files in talk/media/testdata were uploaded to Google Storage and
added .sha1 files in resources/media instead of simply moving them.
The previously disabled warnings that were inherited from
talk/build/common.gypi are now replaced by target-specific disabling
of only the failing warnings. Additional disabling was needed since the stricter
compilation warnings that applies to code in webrtc/.
License headers will be updated in a follow-up CL in order to not
break Git history.
Other modifications:
* Updated the header guards.
* Sorted the includes using chromium/src/tools/sort-headers.py
except for these files:
talk/app/webrtc/peerconnectionendtoend_unittest.cc
talk/app/webrtc/java/jni/androidmediadecoder_jni.cc
talk/app/webrtc/java/jni/androidmediaencoder_jni.cc
webrtc/media/devices/win32devicemanager.cc.
* Unused GYP reference to libjingle_tests_additional_deps was removed.
* Removed duplicated GYP entries of
webrtc/base/testutils.cc
webrtc/base/testutils.h
The HAVE_WEBRTC_VIDEO and HAVE_WEBRTC_VOICE defines were used by only talk/media,
so they were moved to the media.gyp.
I also checked that none of
EXPAT_RELATIVE_PATH,
FEATURE_ENABLE_VOICEMAIL,
GTEST_RELATIVE_PATH,
JSONCPP_RELATIVE_PATH,
LOGGING=1,
SRTP_RELATIVE_PATH,
FEATURE_ENABLE_SSL,
FEATURE_ENABLE_VOICEMAIL,
FEATURE_ENABLE_PSTN,
HAVE_SCTP,
HAVE_SRTP,
are used by the talk/media code.
For Chromium, the following changes will need to be applied to the roll CL that updates the
DEPS for WebRTC and libjingle: https://codereview.chromium.org/1604303002/
BUG=webrtc:5420
NOPRESUBMIT=True
TBR=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1587193006
Cr-Commit-Position: refs/heads/master@{#11495}
diff --git a/webrtc/media/base/rtpdataengine.cc b/webrtc/media/base/rtpdataengine.cc
new file mode 100644
index 0000000..df104f6
--- /dev/null
+++ b/webrtc/media/base/rtpdataengine.cc
@@ -0,0 +1,370 @@
+/*
+ * libjingle
+ * Copyright 2012 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 "webrtc/media/base/rtpdataengine.h"
+
+#include "webrtc/base/buffer.h"
+#include "webrtc/base/helpers.h"
+#include "webrtc/base/logging.h"
+#include "webrtc/base/ratelimiter.h"
+#include "webrtc/base/timing.h"
+#include "webrtc/media/base/codec.h"
+#include "webrtc/media/base/constants.h"
+#include "webrtc/media/base/rtputils.h"
+#include "webrtc/media/base/streamparams.h"
+
+namespace cricket {
+
+// We want to avoid IP fragmentation.
+static const size_t kDataMaxRtpPacketLen = 1200U;
+// We reserve space after the RTP header for future wiggle room.
+static const unsigned char kReservedSpace[] = {
+ 0x00, 0x00, 0x00, 0x00
+};
+
+// Amount of overhead SRTP may take. We need to leave room in the
+// buffer for it, otherwise SRTP will fail later. If SRTP ever uses
+// more than this, we need to increase this number.
+static const size_t kMaxSrtpHmacOverhead = 16;
+
+RtpDataEngine::RtpDataEngine() {
+ data_codecs_.push_back(
+ DataCodec(kGoogleRtpDataCodecId,
+ kGoogleRtpDataCodecName, 0));
+ SetTiming(new rtc::Timing());
+}
+
+DataMediaChannel* RtpDataEngine::CreateChannel(
+ DataChannelType data_channel_type) {
+ if (data_channel_type != DCT_RTP) {
+ return NULL;
+ }
+ return new RtpDataMediaChannel(timing_.get());
+}
+
+bool FindCodecByName(const std::vector<DataCodec>& codecs,
+ const std::string& name, DataCodec* codec_out) {
+ std::vector<DataCodec>::const_iterator iter;
+ for (iter = codecs.begin(); iter != codecs.end(); ++iter) {
+ if (iter->name == name) {
+ *codec_out = *iter;
+ return true;
+ }
+ }
+ return false;
+}
+
+RtpDataMediaChannel::RtpDataMediaChannel(rtc::Timing* timing) {
+ Construct(timing);
+}
+
+RtpDataMediaChannel::RtpDataMediaChannel() {
+ Construct(NULL);
+}
+
+void RtpDataMediaChannel::Construct(rtc::Timing* timing) {
+ sending_ = false;
+ receiving_ = false;
+ timing_ = timing;
+ send_limiter_.reset(new rtc::RateLimiter(kDataMaxBandwidth / 8, 1.0));
+}
+
+
+RtpDataMediaChannel::~RtpDataMediaChannel() {
+ std::map<uint32_t, RtpClock*>::const_iterator iter;
+ for (iter = rtp_clock_by_send_ssrc_.begin();
+ iter != rtp_clock_by_send_ssrc_.end();
+ ++iter) {
+ delete iter->second;
+ }
+}
+
+void RtpClock::Tick(double now, int* seq_num, uint32_t* timestamp) {
+ *seq_num = ++last_seq_num_;
+ *timestamp = timestamp_offset_ + static_cast<uint32_t>(now * clockrate_);
+}
+
+const DataCodec* FindUnknownCodec(const std::vector<DataCodec>& codecs) {
+ DataCodec data_codec(kGoogleRtpDataCodecId, kGoogleRtpDataCodecName, 0);
+ std::vector<DataCodec>::const_iterator iter;
+ for (iter = codecs.begin(); iter != codecs.end(); ++iter) {
+ if (!iter->Matches(data_codec)) {
+ return &(*iter);
+ }
+ }
+ return NULL;
+}
+
+const DataCodec* FindKnownCodec(const std::vector<DataCodec>& codecs) {
+ DataCodec data_codec(kGoogleRtpDataCodecId, kGoogleRtpDataCodecName, 0);
+ std::vector<DataCodec>::const_iterator iter;
+ for (iter = codecs.begin(); iter != codecs.end(); ++iter) {
+ if (iter->Matches(data_codec)) {
+ return &(*iter);
+ }
+ }
+ return NULL;
+}
+
+bool RtpDataMediaChannel::SetRecvCodecs(const std::vector<DataCodec>& codecs) {
+ const DataCodec* unknown_codec = FindUnknownCodec(codecs);
+ if (unknown_codec) {
+ LOG(LS_WARNING) << "Failed to SetRecvCodecs because of unknown codec: "
+ << unknown_codec->ToString();
+ return false;
+ }
+
+ recv_codecs_ = codecs;
+ return true;
+}
+
+bool RtpDataMediaChannel::SetSendCodecs(const std::vector<DataCodec>& codecs) {
+ const DataCodec* known_codec = FindKnownCodec(codecs);
+ if (!known_codec) {
+ LOG(LS_WARNING) <<
+ "Failed to SetSendCodecs because there is no known codec.";
+ return false;
+ }
+
+ send_codecs_ = codecs;
+ return true;
+}
+
+bool RtpDataMediaChannel::SetSendParameters(const DataSendParameters& params) {
+ return (SetSendCodecs(params.codecs) &&
+ SetMaxSendBandwidth(params.max_bandwidth_bps));
+}
+
+bool RtpDataMediaChannel::SetRecvParameters(const DataRecvParameters& params) {
+ return SetRecvCodecs(params.codecs);
+}
+
+bool RtpDataMediaChannel::AddSendStream(const StreamParams& stream) {
+ if (!stream.has_ssrcs()) {
+ return false;
+ }
+
+ if (GetStreamBySsrc(send_streams_, stream.first_ssrc())) {
+ LOG(LS_WARNING) << "Not adding data send stream '" << stream.id
+ << "' with ssrc=" << stream.first_ssrc()
+ << " because stream already exists.";
+ return false;
+ }
+
+ send_streams_.push_back(stream);
+ // TODO(pthatcher): This should be per-stream, not per-ssrc.
+ // And we should probably allow more than one per stream.
+ rtp_clock_by_send_ssrc_[stream.first_ssrc()] = new RtpClock(
+ kDataCodecClockrate,
+ rtc::CreateRandomNonZeroId(), rtc::CreateRandomNonZeroId());
+
+ LOG(LS_INFO) << "Added data send stream '" << stream.id
+ << "' with ssrc=" << stream.first_ssrc();
+ return true;
+}
+
+bool RtpDataMediaChannel::RemoveSendStream(uint32_t ssrc) {
+ if (!GetStreamBySsrc(send_streams_, ssrc)) {
+ return false;
+ }
+
+ RemoveStreamBySsrc(&send_streams_, ssrc);
+ delete rtp_clock_by_send_ssrc_[ssrc];
+ rtp_clock_by_send_ssrc_.erase(ssrc);
+ return true;
+}
+
+bool RtpDataMediaChannel::AddRecvStream(const StreamParams& stream) {
+ if (!stream.has_ssrcs()) {
+ return false;
+ }
+
+ if (GetStreamBySsrc(recv_streams_, stream.first_ssrc())) {
+ LOG(LS_WARNING) << "Not adding data recv stream '" << stream.id
+ << "' with ssrc=" << stream.first_ssrc()
+ << " because stream already exists.";
+ return false;
+ }
+
+ recv_streams_.push_back(stream);
+ LOG(LS_INFO) << "Added data recv stream '" << stream.id
+ << "' with ssrc=" << stream.first_ssrc();
+ return true;
+}
+
+bool RtpDataMediaChannel::RemoveRecvStream(uint32_t ssrc) {
+ RemoveStreamBySsrc(&recv_streams_, ssrc);
+ return true;
+}
+
+void RtpDataMediaChannel::OnPacketReceived(
+ rtc::Buffer* packet, const rtc::PacketTime& packet_time) {
+ RtpHeader header;
+ if (!GetRtpHeader(packet->data(), packet->size(), &header)) {
+ // Don't want to log for every corrupt packet.
+ // LOG(LS_WARNING) << "Could not read rtp header from packet of length "
+ // << packet->length() << ".";
+ return;
+ }
+
+ size_t header_length;
+ if (!GetRtpHeaderLen(packet->data(), packet->size(), &header_length)) {
+ // Don't want to log for every corrupt packet.
+ // LOG(LS_WARNING) << "Could not read rtp header"
+ // << length from packet of length "
+ // << packet->length() << ".";
+ return;
+ }
+ const char* data =
+ packet->data<char>() + header_length + sizeof(kReservedSpace);
+ size_t data_len = packet->size() - header_length - sizeof(kReservedSpace);
+
+ if (!receiving_) {
+ LOG(LS_WARNING) << "Not receiving packet "
+ << header.ssrc << ":" << header.seq_num
+ << " before SetReceive(true) called.";
+ return;
+ }
+
+ DataCodec codec;
+ if (!FindCodecById(recv_codecs_, header.payload_type, &codec)) {
+ // For bundling, this will be logged for every message.
+ // So disable this logging.
+ // LOG(LS_WARNING) << "Not receiving packet "
+ // << header.ssrc << ":" << header.seq_num
+ // << " (" << data_len << ")"
+ // << " because unknown payload id: " << header.payload_type;
+ return;
+ }
+
+ if (!GetStreamBySsrc(recv_streams_, header.ssrc)) {
+ LOG(LS_WARNING) << "Received packet for unknown ssrc: " << header.ssrc;
+ return;
+ }
+
+ // Uncomment this for easy debugging.
+ // const auto* found_stream = GetStreamBySsrc(recv_streams_, header.ssrc);
+ // LOG(LS_INFO) << "Received packet"
+ // << " groupid=" << found_stream.groupid
+ // << ", ssrc=" << header.ssrc
+ // << ", seqnum=" << header.seq_num
+ // << ", timestamp=" << header.timestamp
+ // << ", len=" << data_len;
+
+ ReceiveDataParams params;
+ params.ssrc = header.ssrc;
+ params.seq_num = header.seq_num;
+ params.timestamp = header.timestamp;
+ SignalDataReceived(params, data, data_len);
+}
+
+bool RtpDataMediaChannel::SetMaxSendBandwidth(int bps) {
+ if (bps <= 0) {
+ bps = kDataMaxBandwidth;
+ }
+ send_limiter_.reset(new rtc::RateLimiter(bps / 8, 1.0));
+ LOG(LS_INFO) << "RtpDataMediaChannel::SetSendBandwidth to " << bps << "bps.";
+ return true;
+}
+
+bool RtpDataMediaChannel::SendData(
+ const SendDataParams& params,
+ const rtc::Buffer& payload,
+ SendDataResult* result) {
+ if (result) {
+ // If we return true, we'll set this to SDR_SUCCESS.
+ *result = SDR_ERROR;
+ }
+ if (!sending_) {
+ LOG(LS_WARNING) << "Not sending packet with ssrc=" << params.ssrc
+ << " len=" << payload.size() << " before SetSend(true).";
+ return false;
+ }
+
+ if (params.type != cricket::DMT_TEXT) {
+ LOG(LS_WARNING) << "Not sending data because binary type is unsupported.";
+ return false;
+ }
+
+ const StreamParams* found_stream =
+ GetStreamBySsrc(send_streams_, params.ssrc);
+ if (!found_stream) {
+ LOG(LS_WARNING) << "Not sending data because ssrc is unknown: "
+ << params.ssrc;
+ return false;
+ }
+
+ DataCodec found_codec;
+ if (!FindCodecByName(send_codecs_, kGoogleRtpDataCodecName, &found_codec)) {
+ LOG(LS_WARNING) << "Not sending data because codec is unknown: "
+ << kGoogleRtpDataCodecName;
+ return false;
+ }
+
+ size_t packet_len = (kMinRtpPacketLen + sizeof(kReservedSpace) +
+ payload.size() + kMaxSrtpHmacOverhead);
+ if (packet_len > kDataMaxRtpPacketLen) {
+ return false;
+ }
+
+ double now = timing_->TimerNow();
+
+ if (!send_limiter_->CanUse(packet_len, now)) {
+ LOG(LS_VERBOSE) << "Dropped data packet of len=" << packet_len
+ << "; already sent " << send_limiter_->used_in_period()
+ << "/" << send_limiter_->max_per_period();
+ return false;
+ }
+
+ RtpHeader header;
+ header.payload_type = found_codec.id;
+ header.ssrc = params.ssrc;
+ rtp_clock_by_send_ssrc_[header.ssrc]->Tick(
+ now, &header.seq_num, &header.timestamp);
+
+ rtc::Buffer packet(kMinRtpPacketLen, packet_len);
+ if (!SetRtpHeader(packet.data(), packet.size(), header)) {
+ return false;
+ }
+ packet.AppendData(kReservedSpace);
+ packet.AppendData(payload);
+
+ LOG(LS_VERBOSE) << "Sent RTP data packet: "
+ << " stream=" << found_stream->id << " ssrc=" << header.ssrc
+ << ", seqnum=" << header.seq_num
+ << ", timestamp=" << header.timestamp
+ << ", len=" << payload.size();
+
+ MediaChannel::SendPacket(&packet, rtc::PacketOptions());
+ send_limiter_->Use(packet_len, now);
+ if (result) {
+ *result = SDR_SUCCESS;
+ }
+ return true;
+}
+
+} // namespace cricket