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Diffstat (limited to 'media/cast/audio_receiver/audio_receiver.cc')
-rw-r--r--media/cast/audio_receiver/audio_receiver.cc113
1 files changed, 100 insertions, 13 deletions
diff --git a/media/cast/audio_receiver/audio_receiver.cc b/media/cast/audio_receiver/audio_receiver.cc
index c3dc3937b9..01e0026eb0 100644
--- a/media/cast/audio_receiver/audio_receiver.cc
+++ b/media/cast/audio_receiver/audio_receiver.cc
@@ -7,6 +7,8 @@
#include "base/bind.h"
#include "base/logging.h"
#include "base/message_loop/message_loop.h"
+#include "crypto/encryptor.h"
+#include "crypto/symmetric_key.h"
#include "media/cast/audio_receiver/audio_decoder.h"
#include "media/cast/framer/framer.h"
#include "media/cast/rtcp/rtcp.h"
@@ -19,7 +21,6 @@ static const int64 kMinSchedulingDelayMs = 1;
namespace media {
namespace cast {
-
// Local implementation of RtpData (defined in rtp_rtcp_defines.h).
// Used to pass payload data into the audio receiver.
class LocalRtpAudioData : public RtpData {
@@ -80,7 +81,6 @@ AudioReceiver::AudioReceiver(scoped_refptr<CastEnvironment> cast_environment,
PacedPacketSender* const packet_sender)
: cast_environment_(cast_environment),
codec_(audio_config.codec),
- incoming_ssrc_(audio_config.incoming_ssrc),
frequency_(audio_config.frequency),
audio_buffer_(),
audio_decoder_(),
@@ -97,8 +97,20 @@ AudioReceiver::AudioReceiver(scoped_refptr<CastEnvironment> cast_environment,
true,
0));
} else {
- audio_decoder_ = new AudioDecoder(audio_config);
+ audio_decoder_.reset(new AudioDecoder(audio_config));
+ }
+ if (audio_config.aes_iv_mask.size() == kAesKeySize &&
+ audio_config.aes_key.size() == kAesKeySize) {
+ iv_mask_ = audio_config.aes_iv_mask;
+ crypto::SymmetricKey* key = crypto::SymmetricKey::Import(
+ crypto::SymmetricKey::AES, audio_config.aes_key);
+ decryptor_.reset(new crypto::Encryptor());
+ decryptor_->Init(key, crypto::Encryptor::CTR, std::string());
+ } else if (audio_config.aes_iv_mask.size() != 0 ||
+ audio_config.aes_key.size() != 0) {
+ DCHECK(false) << "Invalid crypto configuration";
}
+
rtp_receiver_.reset(new RtpReceiver(cast_environment->Clock(),
&audio_config,
NULL,
@@ -107,36 +119,61 @@ AudioReceiver::AudioReceiver(scoped_refptr<CastEnvironment> cast_environment,
new LocalRtpReceiverStatistics(rtp_receiver_.get()));
base::TimeDelta rtcp_interval_delta =
base::TimeDelta::FromMilliseconds(audio_config.rtcp_interval);
- rtcp_.reset(new Rtcp(cast_environment->Clock(),
+ rtcp_.reset(new Rtcp(cast_environment,
NULL,
packet_sender,
NULL,
rtp_audio_receiver_statistics_.get(),
audio_config.rtcp_mode,
rtcp_interval_delta,
- false,
audio_config.feedback_ssrc,
+ audio_config.incoming_ssrc,
audio_config.rtcp_c_name));
- rtcp_->SetRemoteSSRC(audio_config.incoming_ssrc);
- ScheduleNextRtcpReport();
- ScheduleNextCastMessage();
}
AudioReceiver::~AudioReceiver() {}
+void AudioReceiver::InitializeTimers() {
+ DCHECK(cast_environment_->CurrentlyOn(CastEnvironment::MAIN));
+ ScheduleNextRtcpReport();
+ ScheduleNextCastMessage();
+}
+
void AudioReceiver::IncomingParsedRtpPacket(const uint8* payload_data,
size_t payload_size,
const RtpCastHeader& rtp_header) {
+ DCHECK(cast_environment_->CurrentlyOn(CastEnvironment::MAIN));
+ cast_environment_->Logging()->InsertPacketEvent(kPacketReceived,
+ rtp_header.webrtc.header.timestamp, rtp_header.frame_id,
+ rtp_header.packet_id, rtp_header.max_packet_id, payload_size);
+
// TODO(pwestin): update this as video to refresh over time.
+ DCHECK(cast_environment_->CurrentlyOn(CastEnvironment::MAIN));
if (time_first_incoming_packet_.is_null()) {
+ InitializeTimers();
first_incoming_rtp_timestamp_ = rtp_header.webrtc.header.timestamp;
time_first_incoming_packet_ = cast_environment_->Clock()->NowTicks();
}
if (audio_decoder_) {
DCHECK(!audio_buffer_) << "Invalid internal state";
- audio_decoder_->IncomingParsedRtpPacket(payload_data, payload_size,
- rtp_header);
+ std::string plaintext(reinterpret_cast<const char*>(payload_data),
+ payload_size);
+ if (decryptor_) {
+ plaintext.clear();
+ if (!decryptor_->SetCounter(GetAesNonce(rtp_header.frame_id, iv_mask_))) {
+ NOTREACHED() << "Failed to set counter";
+ return;
+ }
+ if (!decryptor_->Decrypt(base::StringPiece(reinterpret_cast<const char*>(
+ payload_data), payload_size), &plaintext)) {
+ VLOG(0) << "Decryption error";
+ return;
+ }
+ }
+ audio_decoder_->IncomingParsedRtpPacket(
+ reinterpret_cast<const uint8*>(plaintext.data()), plaintext.size(),
+ rtp_header);
return;
}
DCHECK(audio_buffer_) << "Invalid internal state";
@@ -155,11 +192,13 @@ void AudioReceiver::IncomingParsedRtpPacket(const uint8* payload_data,
void AudioReceiver::GetRawAudioFrame(int number_of_10ms_blocks,
int desired_frequency, const AudioFrameDecodedCallback& callback) {
+ DCHECK(cast_environment_->CurrentlyOn(CastEnvironment::MAIN));
DCHECK(audio_decoder_) << "Invalid function call in this configuration";
+ // TODO(pwestin): we can skip this function by posting direct to the decoder.
cast_environment_->PostTask(CastEnvironment::AUDIO_DECODER, FROM_HERE,
base::Bind(&AudioReceiver::DecodeAudioFrameThread,
- weak_factory_.GetWeakPtr(),
+ base::Unretained(this),
number_of_10ms_blocks,
desired_frequency,
callback));
@@ -178,12 +217,17 @@ void AudioReceiver::DecodeAudioFrameThread(
desired_frequency,
audio_frame.get(),
&rtp_timestamp)) {
+ // TODO(pwestin): This looks wrong, we would loose the pending call to
+ // the application provided callback.
return;
}
base::TimeTicks now = cast_environment_->Clock()->NowTicks();
base::TimeTicks playout_time;
playout_time = GetPlayoutTime(now, rtp_timestamp);
+ cast_environment_->Logging()->InsertFrameEventWithDelay(kAudioPlayoutDelay,
+ rtp_timestamp, kFrameIdUnknown, playout_time - now);
+
// Frame is ready - Send back to the main thread.
cast_environment_->PostTask(CastEnvironment::MAIN, FROM_HERE,
base::Bind(callback,
@@ -191,6 +235,7 @@ void AudioReceiver::DecodeAudioFrameThread(
}
void AudioReceiver::PlayoutTimeout() {
+ DCHECK(cast_environment_->CurrentlyOn(CastEnvironment::MAIN));
DCHECK(audio_buffer_) << "Invalid function call in this configuration";
if (queued_encoded_callbacks_.empty()) {
// Already released by incoming packet.
@@ -209,6 +254,12 @@ void AudioReceiver::PlayoutTimeout() {
VLOG(1) << "Failed to retrieved a complete frame at this point in time";
return;
}
+
+ if (decryptor_ && !DecryptAudioFrame(&encoded_frame)) {
+ // Logging already done.
+ return;
+ }
+
if (PostEncodedAudioFrame(queued_encoded_callbacks_.front(), rtp_timestamp,
next_frame, &encoded_frame)) {
// Call succeed remove callback from list.
@@ -218,6 +269,7 @@ void AudioReceiver::PlayoutTimeout() {
void AudioReceiver::GetEncodedAudioFrame(
const AudioFrameEncodedCallback& callback) {
+ DCHECK(cast_environment_->CurrentlyOn(CastEnvironment::MAIN));
DCHECK(audio_buffer_) << "Invalid function call in this configuration";
uint32 rtp_timestamp = 0;
@@ -231,6 +283,11 @@ void AudioReceiver::GetEncodedAudioFrame(
queued_encoded_callbacks_.push_back(callback);
return;
}
+ if (decryptor_ && !DecryptAudioFrame(&encoded_frame)) {
+ // Logging already done.
+ queued_encoded_callbacks_.push_back(callback);
+ return;
+ }
if (!PostEncodedAudioFrame(callback, rtp_timestamp, next_frame,
&encoded_frame)) {
// We have an audio frame; however we are missing packets and we have time
@@ -244,7 +301,9 @@ bool AudioReceiver::PostEncodedAudioFrame(
uint32 rtp_timestamp,
bool next_frame,
scoped_ptr<EncodedAudioFrame>* encoded_frame) {
+ DCHECK(cast_environment_->CurrentlyOn(CastEnvironment::MAIN));
DCHECK(audio_buffer_) << "Invalid function call in this configuration";
+
base::TimeTicks now = cast_environment_->Clock()->NowTicks();
base::TimeTicks playout_time = GetPlayoutTime(now, rtp_timestamp);
base::TimeDelta time_until_playout = playout_time - now;
@@ -270,6 +329,7 @@ bool AudioReceiver::PostEncodedAudioFrame(
void AudioReceiver::IncomingPacket(const uint8* packet, size_t length,
const base::Closure callback) {
+ DCHECK(cast_environment_->CurrentlyOn(CastEnvironment::MAIN));
bool rtcp_packet = Rtcp::IsRtcpPacket(packet, length);
if (!rtcp_packet) {
rtp_receiver_->ReceivedPacket(packet, length);
@@ -280,11 +340,13 @@ void AudioReceiver::IncomingPacket(const uint8* packet, size_t length,
}
void AudioReceiver::CastFeedback(const RtcpCastMessage& cast_message) {
- rtcp_->SendRtcpCast(cast_message);
+ // TODO(pwestin): add logging.
+ rtcp_->SendRtcpFromRtpReceiver(&cast_message, NULL);
}
base::TimeTicks AudioReceiver::GetPlayoutTime(base::TimeTicks now,
uint32 rtp_timestamp) {
+ DCHECK(cast_environment_->CurrentlyOn(CastEnvironment::MAIN));
// Senders time in ms when this frame was recorded.
// Note: the senders clock and our local clock might not be synced.
base::TimeTicks rtp_timestamp_in_ticks;
@@ -314,7 +376,28 @@ base::TimeTicks AudioReceiver::GetPlayoutTime(base::TimeTicks now,
now;
}
+bool AudioReceiver::DecryptAudioFrame(
+ scoped_ptr<EncodedAudioFrame>* audio_frame) {
+ DCHECK(decryptor_) << "Invalid state";
+
+ if (!decryptor_->SetCounter(GetAesNonce((*audio_frame)->frame_id,
+ iv_mask_))) {
+ NOTREACHED() << "Failed to set counter";
+ return false;
+ }
+ std::string decrypted_audio_data;
+ if (!decryptor_->Decrypt((*audio_frame)->data, &decrypted_audio_data)) {
+ VLOG(0) << "Decryption error";
+ // Give up on this frame, release it from jitter buffer.
+ audio_buffer_->ReleaseFrame((*audio_frame)->frame_id);
+ return false;
+ }
+ (*audio_frame)->data.swap(decrypted_audio_data);
+ return true;
+}
+
void AudioReceiver::ScheduleNextRtcpReport() {
+ DCHECK(cast_environment_->CurrentlyOn(CastEnvironment::MAIN));
base::TimeDelta time_to_send = rtcp_->TimeToSendNextRtcpReport() -
cast_environment_->Clock()->NowTicks();
@@ -327,13 +410,16 @@ void AudioReceiver::ScheduleNextRtcpReport() {
}
void AudioReceiver::SendNextRtcpReport() {
- rtcp_->SendRtcpReport(incoming_ssrc_);
+ DCHECK(cast_environment_->CurrentlyOn(CastEnvironment::MAIN));
+ // TODO(pwestin): add logging.
+ rtcp_->SendRtcpFromRtpReceiver(NULL, NULL);
ScheduleNextRtcpReport();
}
// Cast messages should be sent within a maximum interval. Schedule a call
// if not triggered elsewhere, e.g. by the cast message_builder.
void AudioReceiver::ScheduleNextCastMessage() {
+ DCHECK(cast_environment_->CurrentlyOn(CastEnvironment::MAIN));
if (audio_buffer_) {
base::TimeTicks send_time;
audio_buffer_->TimeToSendNextCastMessage(&send_time);
@@ -349,6 +435,7 @@ void AudioReceiver::ScheduleNextCastMessage() {
}
void AudioReceiver::SendNextCastMessage() {
+ DCHECK(cast_environment_->CurrentlyOn(CastEnvironment::MAIN));
DCHECK(audio_buffer_) << "Invalid function call in this configuration";
audio_buffer_->SendCastMessage(); // Will only send a message if it is time.
ScheduleNextCastMessage();