Cast: Fix audio pipeline

1. Add audio decoded callback  - guarantees that we will return an audio frame when invoked.
2. Bug fixes in sender.cc
3. Add a return in transport.cc when we drop a packet.

BUG=323703

Review URL: https://codereview.chromium.org/102293004

git-svn-id: svn://svn.chromium.org/chrome/trunk/src@238992 0039d316-1c4b-4281-b951-d872f2087c98

5 files changed, 51 insertions, 16 deletions

diff --git a/media/cast/audio_receiver/audio_receiver.cc b/media/cast/audio_receiver/audio_receiver.cc
index ea56195..9347f15 100644
--- a/media/cast/audio_receiver/audio_receiver.cc
+++ b/media/cast/audio_receiver/audio_receiver.cc
@@ -21,6 +21,13 @@ static const int64 kMinSchedulingDelayMs = 1;
 namespace media {
 namespace cast {
 
+DecodedAudioCallbackData::DecodedAudioCallbackData()
+    : number_of_10ms_blocks(0),
+      desired_frequency(0),
+      callback() {}
+
+DecodedAudioCallbackData::~DecodedAudioCallbackData() {}
+
 // Local implementation of RtpData (defined in rtp_rtcp_defines.h).
 // Used to pass payload data into the audio receiver.
 class LocalRtpAudioData : public RtpData {
@@ -174,27 +181,37 @@ void AudioReceiver::IncomingParsedRtpPacket(const uint8* payload_data,
     audio_decoder_->IncomingParsedRtpPacket(
         reinterpret_cast<const uint8*>(plaintext.data()), plaintext.size(),
         rtp_header);
+    if (!queued_decoded_callbacks_.empty()) {
+      DecodedAudioCallbackData decoded_data = queued_decoded_callbacks_.front();
+      queued_decoded_callbacks_.pop_front();
+      cast_environment_->PostTask(CastEnvironment::AUDIO_DECODER, FROM_HERE,
+        base::Bind(&AudioReceiver::DecodeAudioFrameThread,
+                   base::Unretained(this),
+                   decoded_data.number_of_10ms_blocks,
+                   decoded_data.desired_frequency,
+                   decoded_data.callback));
+    }
     return;
   }
+
   DCHECK(audio_buffer_) << "Invalid internal state";
   DCHECK(!audio_decoder_) << "Invalid internal state";
+
   bool complete = audio_buffer_->InsertPacket(payload_data, payload_size,
                                               rtp_header);
   if (!complete) return;  // Audio frame not complete; wait for more packets.
-  if (queued_encoded_callbacks_.empty()) return;  // No pending callback.
-
+  if (queued_encoded_callbacks_.empty()) return;
   AudioFrameEncodedCallback callback = queued_encoded_callbacks_.front();
   queued_encoded_callbacks_.pop_front();
   cast_environment_->PostTask(CastEnvironment::MAIN, FROM_HERE,
       base::Bind(&AudioReceiver::GetEncodedAudioFrame,
-          weak_factory_.GetWeakPtr(), callback));
+      weak_factory_.GetWeakPtr(), callback));
 }
 
 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,
@@ -217,8 +234,11 @@ 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.
+    DecodedAudioCallbackData callback_data;
+    callback_data.number_of_10ms_blocks = number_of_10ms_blocks;
+    callback_data.desired_frequency = desired_frequency;
+    callback_data.callback = callback;
+    queued_decoded_callbacks_.push_back(callback_data);
     return;
   }
   base::TimeTicks now = cast_environment_->Clock()->NowTicks();
diff --git a/media/cast/audio_receiver/audio_receiver.h b/media/cast/audio_receiver/audio_receiver.h
index 53e85db..0298641 100644
--- a/media/cast/audio_receiver/audio_receiver.h
+++ b/media/cast/audio_receiver/audio_receiver.h
@@ -34,6 +34,14 @@ class PacedPacketSender;
 class RtpReceiver;
 class RtpReceiverStatistics;
 
+struct DecodedAudioCallbackData {
+  DecodedAudioCallbackData();
+  ~DecodedAudioCallbackData();
+  int number_of_10ms_blocks;
+  int desired_frequency;
+  AudioFrameDecodedCallback callback;
+};
+
 // This class is not thread safe. Should only be called from the Main cast
 // thread.
 class AudioReceiver : public base::NonThreadSafe,
@@ -126,6 +134,7 @@ class AudioReceiver : public base::NonThreadSafe,
   std::string iv_mask_;
 
   std::list<AudioFrameEncodedCallback> queued_encoded_callbacks_;
+  std::list<DecodedAudioCallbackData> queued_decoded_callbacks_;
 };
 
 }  // namespace cast
diff --git a/media/cast/test/receiver.cc b/media/cast/test/receiver.cc
index 356d14a9..1321f53 100644
--- a/media/cast/test/receiver.cc
+++ b/media/cast/test/receiver.cc
@@ -29,13 +29,13 @@
 
 namespace media {
 namespace cast {
-
+// Settings chosen to match default sender settings.
 #define DEFAULT_SEND_PORT "2346"
 #define DEFAULT_RECEIVE_PORT "2344"
 #define DEFAULT_SEND_IP "127.0.0.1"
 #define DEFAULT_RESTART "0"
-#define DEFAULT_AUDIO_FEEDBACK_SSRC "2"
-#define DEFAULT_AUDIO_INCOMING_SSRC "1"
+#define DEFAULT_AUDIO_FEEDBACK_SSRC "1"
+#define DEFAULT_AUDIO_INCOMING_SSRC "2"
 #define DEFAULT_AUDIO_PAYLOAD_TYPE "127"
 #define DEFAULT_VIDEO_FEEDBACK_SSRC "12"
 #define DEFAULT_VIDEO_INCOMING_SSRC "11"
@@ -180,7 +180,7 @@ class ReceiveProcess : public base::RefCountedThreadSafe<ReceiveProcess> {
         base::TimeDelta::FromMilliseconds(kFrameTimerMs);
     if (!last_playout_time_.is_null()){
         time_diff = playout_time - last_playout_time_;
-        VLOG(0) << " PlayoutDelay[mS] =  " << time_diff.InMilliseconds();
+        VLOG(0) << " ***PlayoutDelay[mS] =  " << time_diff.InMilliseconds();
     }
     last_playout_time_ = playout_time;
     GetAudioFrame(time_diff);
diff --git a/media/cast/test/sender.cc b/media/cast/test/sender.cc
index 4e491bf..e4c26de 100644
--- a/media/cast/test/sender.cc
+++ b/media/cast/test/sender.cc
@@ -21,6 +21,9 @@
 #include "media/cast/test/video_utility.h"
 #include "ui/gfx/size.h"
 
+namespace media {
+namespace cast {
+// Settings chosen to match default receiver settings.
 #define DEFAULT_SEND_PORT "2344"
 #define DEFAULT_RECEIVE_PORT "2346"
 #define DEFAULT_SEND_IP "127.0.0.1"
@@ -38,9 +41,6 @@
 #define DEFAULT_VIDEO_CODEC_MAX_BITRATE "4000"
 #define DEFAULT_VIDEO_CODEC_MIN_BITRATE "1000"
 
-namespace media {
-namespace cast {
-
 namespace {
 static const int kAudioChannels = 2;
 static const int kAudioSamplingFrequency = 48000;
@@ -49,6 +49,12 @@ static const int kSoundFrequency = 1234;  // Frequency of sinusoid wave.
 // a normal video is 30 fps hence the 33 ms between frames.
 static const float kSoundVolume = 0.5f;
 static const int kFrameTimerMs = 33;
+
+// Dummy callback function that does nothing except to accept ownership of
+// |audio_bus| for destruction. This guarantees that the audio_bus is valid for
+// the entire duration of the encode/send process (not equivalent to DoNothing).
+void OwnThatAudioBus(scoped_ptr<AudioBus> audio_bus) {
+}
 } // namespace
 
 void GetPorts(int* tx_port, int* rx_port) {
@@ -231,7 +237,7 @@ class SendProcess {
         base::TimeDelta::FromMilliseconds(10) * num_10ms_blocks));
     AudioBus* const audio_bus_ptr = audio_bus.get();
     frame_input_->InsertAudio(audio_bus_ptr, clock_->NowTicks(),
-        base::Bind(base::DoNothing));
+        base::Bind(&OwnThatAudioBus, base::Passed(&audio_bus)));
 
     gfx::Size size(video_config_.width, video_config_.height);
     // TODO(mikhal): Use the provided timestamp.
diff --git a/media/cast/test/transport/transport.cc b/media/cast/test/transport/transport.cc
index 32ee246..9312470 100644
--- a/media/cast/test/transport/transport.cc
+++ b/media/cast/test/transport/transport.cc
@@ -114,8 +114,7 @@ class LocalPacketSender : public PacketSender,
 
   virtual bool SendPacket(const Packet& packet) OVERRIDE {
     io_thread_proxy_->PostTask(FROM_HERE,
-    base::Bind(&LocalPacketSender::SendPacketToNetwork,
-               this, packet));
+    base::Bind(&LocalPacketSender::SendPacketToNetwork, this, packet));
     return true;
   }
 
@@ -128,6 +127,7 @@ class LocalPacketSender : public PacketSender,
         VLOG(1) << "Drop packet f:" << static_cast<int>(data[12 + 1])
                 << " p:" << static_cast<int>(data[12 + 3])
                 << " m:" << static_cast<int>(data[12 + 5]);
+        return;
       }
     }
     net::TestCompletionCallback callback;