ppapi: add AudioEncode example

ppapi/examples/audio_encode/audio_encode.cc

Index: ppapi/examples/audio_encode/audio_encode.cc
diff --git a/ppapi/examples/audio_encode/audio_encode.cc b/ppapi/examples/audio_encode/audio_encode.cc
new file mode 100644
index 0000000000000000000000000000000000000000..91d712c080f1610692df8068d04eeec0f60a7487
--- /dev/null
+++ b/ppapi/examples/audio_encode/audio_encode.cc
@@ -0,0 +1,412 @@
+// Copyright 2015 The Chromium Authors. All rights reserved.
+// Use of this source code is governed by a BSD-style license that can be
+// found in the LICENSE file.
+
+#include <stdlib.h>
+#include <string.h>
+
+#include <map>
+#include <string>
+#include <vector>
+
+#include "ppapi/cpp/audio_buffer.h"
+#include "ppapi/cpp/audio_encoder.h"
+#include "ppapi/cpp/instance.h"
+#include "ppapi/cpp/logging.h"
+#include "ppapi/cpp/media_stream_audio_track.h"
+#include "ppapi/cpp/module.h"
+#include "ppapi/cpp/var_array_buffer.h"
+#include "ppapi/cpp/var_dictionary.h"
+#include "ppapi/utility/completion_callback_factory.h"
+
+// When compiling natively on Windows, PostMessage can be #define-d to
+// something else.
+#ifdef PostMessage
+#undef PostMessage
+#endif
+
+// This example demonstrates receiving audio samples from an audio
+// track and encoding them.
+
+namespace {
+
+class AudioEncoderInstance : public pp::Instance {
+ public:
+  explicit AudioEncoderInstance(PP_Instance instance);
+  virtual ~AudioEncoderInstance();
+
+ private:
+  // pp::Instance implementation.
+  virtual void HandleMessage(const pp::Var& var_message);
+
+  void AddAudioProfile(PP_AudioProfile profile, const std::string& profile_str);
+  void InitializeAudioProfiles();
+  PP_AudioProfile ProfileFromString(const std::string& string);
+  std::string ProfileToString(PP_AudioProfile profile);
+
+  void GetSupportedProfiles();
+  void OnGetSupportedProfiles(
+      int32_t result,
+      const std::vector<PP_AudioProfileDescription> profiles);
+
+  void StartAudioTrack(const pp::Resource& resource_track);
+  void StopAudioTrack();
+  void OnGetBuffer(int32_t result, pp::AudioBuffer buffer);
+
+  void InitializeEncoder();
+  void OnEncoderInitialized(int32_t result);
+  void OnAudioTrackConfigured(int32_t result);
+  void GetEncoderBuffer(const pp::AudioBuffer& track_buffer);
+  void OnGetEncoderBuffer(int32_t result,
+                          pp::AudioBuffer buffer,
+                          pp::AudioBuffer track_buffer);
+  void OnEncodeDone(int32_t result);
+  void OnGetBitstreamBuffer(int32_t result,
+                            const PP_AudioBitstreamBuffer& buffer);
+
+  pp::VarDictionary NewCommand(const std::string& type);
+
+  void Log(const std::string& message);
+  void LogError(int32_t error, const std::string& message);
+
+  void PostAudioFormat();
+  void PostDataMessage(const void* data, uint32_t size);
+
+  pp::CompletionCallbackFactory<AudioEncoderInstance> callback_factory_;
+
+  // Set when the plugin is not performing any encoding and just passing the
+  // uncompressed data directly to web page.
+  bool no_encoding_;
+  bool is_encoding_;

[bbudge, 2015/12/08 17:12:30]

This bit of state doesn't seem to be needed.

[llandwerlin-old, 2015/12/08 18:01:01]

Done.

+
+  uint32_t channels_;
+  uint32_t sample_rate_;
+  uint32_t sample_size_;
+  uint32_t samples_per_frame_;
+
+  pp::MediaStreamAudioTrack audio_track_;
+  pp::AudioEncoder audio_encoder_;
+
+  typedef std::map<std::string, PP_AudioProfile> AudioProfileFromStringMap;

[bbudge, 2015/12/08 17:12:30]

Of all of this, only ProfileToString seems to be needed. I think you can
eliminate these types, and the ProfileFromString and ProfileToString methods,
and just use a switch in OnGetSupportedProfiles to convert profiles to strings.

[llandwerlin-old, 2015/12/08 18:01:01]

We forward all the supported profiles to JS.
Then once the UI has selected what codec is used, it sends back the encoder
configuration with the codec name in a string.
That's why I'm using this table. 

I agree this is a bit silly because there is only one codec available.
Changing to if statements.

+  AudioProfileFromStringMap profile_from_string_;
+
+  typedef std::map<PP_AudioProfile, std::string> AudioProfileToStringMap;
+  AudioProfileToStringMap profile_to_string_;
+};
+
+AudioEncoderInstance::AudioEncoderInstance(PP_Instance instance)
+    : pp::Instance(instance),
+      callback_factory_(this),
+      no_encoding_(true),
+      is_encoding_(false),
+      channels_(0),
+      sample_rate_(0),
+      sample_size_(0),
+      samples_per_frame_(0),
+      audio_encoder_(this) {

[bbudge, 2015/12/08 17:12:30]

I now see why you didn't do this before. It would be better not to have to
create a throw-away encoder just to support GetSupportedProfiles. Could you just
initialize to a null resource here and defer the call to GetSupportedProfiles
until after the audio_track_ has given you a buffer?

Also, I don't see how the results of GetSupportedProfiles are used. Are you
assuming OPUS is always supported?

[llandwerlin-old, 2015/12/08 18:01:01]

Done for the encoder initialized at null here.

The results of GetSupportedProfiles are propagated to JS, to populate the UI
with the available options.

+  InitializeAudioProfiles();
+  GetSupportedProfiles();
+}
+
+AudioEncoderInstance::~AudioEncoderInstance() {}
+
+void AudioEncoderInstance::HandleMessage(const pp::Var& var_message) {
+  if (!var_message.is_dictionary()) {
+    LogError(PP_ERROR_FAILED,
+             "Cannot handle incoming message, not a dictionary");
+    return;
+  }
+
+  pp::VarDictionary dict_message(var_message);
+  std::string command = dict_message.Get("command").AsString();
+
+  if (command == "start") {
+    pp::Var var_track = dict_message.Get("track");
+    if (!var_track.is_resource()) {
+      LogError(PP_ERROR_FAILED, "Given track is not a resource");
+      return;
+    }
+    std::string profile = dict_message.Get("profile").AsString();
+    if (profile == "wav") {
+      no_encoding_ = true;
+    } else {
+      no_encoding_ = false;
+    }
+    StartAudioTrack(var_track.AsResource());
+  } else if (command == "stop") {
+    StopAudioTrack();
+  } else {
+    LogError(PP_ERROR_FAILED, "Invalid command");
+  }
+}
+
+void AudioEncoderInstance::AddAudioProfile(PP_AudioProfile profile,
+                                           const std::string& profile_str) {
+  profile_to_string_.insert(std::make_pair(profile, profile_str));
+  profile_from_string_.insert(std::make_pair(profile_str, profile));
+}
+
+void AudioEncoderInstance::InitializeAudioProfiles() {
+  AddAudioProfile(PP_AUDIOPROFILE_OPUS, "opus");
+}
+
+PP_AudioProfile AudioEncoderInstance::ProfileFromString(
+    const std::string& string) {
+  AudioProfileFromStringMap::iterator it = profile_from_string_.find(string);
+  if (it == profile_from_string_.end())
+    return PP_AUDIOPROFILE_OPUS;
+  return it->second;
+}
+
+std::string AudioEncoderInstance::ProfileToString(PP_AudioProfile profile) {
+  AudioProfileToStringMap::iterator it = profile_to_string_.find(profile);
+  if (it == profile_to_string_.end())
+    return "unknown";
+  return it->second;
+}
+
+void AudioEncoderInstance::GetSupportedProfiles() {
+  audio_encoder_.GetSupportedProfiles(callback_factory_.NewCallbackWithOutput(
+      &AudioEncoderInstance::OnGetSupportedProfiles));
+}
+
+void AudioEncoderInstance::OnGetSupportedProfiles(
+    int32_t result,
+    const std::vector<PP_AudioProfileDescription> profiles) {
+  pp::VarDictionary dictionary = NewCommand("supportedProfiles");
+  pp::VarArray js_profiles;
+  dictionary.Set(pp::Var("profiles"), js_profiles);
+
+  if (result < 0) {
+    LogError(result, "Cannot get supported profiles");
+    PostMessage(dictionary);
+  }
+
+  int32_t idx = 0;
+  for (std::vector<PP_AudioProfileDescription>::const_iterator it =
+           profiles.begin();
+       it != profiles.end(); ++it) {
+    pp::VarDictionary js_profile;
+    js_profile.Set(pp::Var("name"), pp::Var(ProfileToString(it->profile)));
+    js_profile.Set(pp::Var("sample_size"),
+                   pp::Var(static_cast<int32_t>(it->sample_size)));
+    js_profile.Set(pp::Var("sample_rate"),
+                   pp::Var(static_cast<int32_t>(it->sample_rate)));
+    js_profiles.Set(idx++, js_profile);
+  }
+  PostMessage(dictionary);
+}
+
+void AudioEncoderInstance::StartAudioTrack(const pp::Resource& resource_track) {

[bbudge, 2015/12/08 17:12:30]

Log("StartAudioTrack!"); ?

[llandwerlin-old, 2015/12/08 18:01:01]

Dropping the traces.

+  channels_ = 0;

[bbudge, 2015/12/08 17:12:30]

It would be more robust if channels_ was always 0 when the audio_track_ was not
running, so this should be done in StopAudioTrack.

[llandwerlin-old, 2015/12/08 18:01:01]

Done.

+  audio_track_ = pp::MediaStreamAudioTrack(resource_track);
+  audio_track_.GetBuffer(callback_factory_.NewCallbackWithOutput(
+      &AudioEncoderInstance::OnGetBuffer));
+}
+
+void AudioEncoderInstance::StopAudioTrack() {
+  Log("StopAudioTrack!");
+  audio_track_.Close();
+  audio_track_ = pp::MediaStreamAudioTrack();
+  audio_encoder_.Close();
+  audio_encoder_ = pp::AudioEncoder();
+}
+
+void AudioEncoderInstance::OnGetBuffer(int32_t result, pp::AudioBuffer buffer) {
+  if (result == PP_ERROR_ABORTED)
+    return;
+  if (result != PP_OK) {
+    LogError(result, "Cannot get audio track buffer");
+    return;
+  }
+
+  // If this is the first buffer, then we need to initialize the encoder.
+  if (channels_ == 0) {
+    channels_ = buffer.GetNumberOfChannels();
+    sample_rate_ = buffer.GetSampleRate();
+    sample_size_ = buffer.GetSampleSize();
+    samples_per_frame_ = buffer.GetNumberOfSamples();
+
+    if (no_encoding_) {
+      PostAudioFormat();
+      PostDataMessage(buffer.GetDataBuffer(), buffer.GetDataBufferSize());
+      audio_track_.GetBuffer(callback_factory_.NewCallbackWithOutput(
+          &AudioEncoderInstance::OnGetBuffer));
+    } else {
+      InitializeEncoder();
+    }
+
+    audio_track_.RecycleBuffer(buffer);
+  } else {
+    if (no_encoding_) {
+      PostDataMessage(buffer.GetDataBuffer(), buffer.GetDataBufferSize());
+      audio_track_.RecycleBuffer(buffer);
+    } else {
+      GetEncoderBuffer(buffer);
+    }
+
+    audio_track_.GetBuffer(callback_factory_.NewCallbackWithOutput(
+        &AudioEncoderInstance::OnGetBuffer));
+  }
+}
+
+void AudioEncoderInstance::InitializeEncoder() {
+  audio_encoder_ = pp::AudioEncoder(this);
+  audio_encoder_.Initialize(
+      channels_, static_cast<PP_AudioBuffer_SampleRate>(sample_rate_),
+      static_cast<PP_AudioBuffer_SampleSize>(sample_size_),
+      PP_AUDIOPROFILE_OPUS, 100000, PP_HARDWAREACCELERATION_WITHFALLBACK,
+      callback_factory_.NewCallback(
+          &AudioEncoderInstance::OnEncoderInitialized));
+}
+
+void AudioEncoderInstance::OnEncoderInitialized(int32_t result) {
+  if (result == PP_ERROR_ABORTED)
+    return;
+  if (result != PP_OK) {
+    LogError(result, "Cannot initialize encoder");
+    return;
+  }
+
+  int32_t attribs[] = {
+      // Duration in milliseconds.
+      PP_MEDIASTREAMAUDIOTRACK_ATTRIB_DURATION,
+      1000 / (sample_rate_ / audio_encoder_.GetNumberOfSamples()),
+      PP_MEDIASTREAMAUDIOTRACK_ATTRIB_NONE};
+  audio_track_.Configure(attribs,
+                         callback_factory_.NewCallback(
+                             &AudioEncoderInstance::OnAudioTrackConfigured));
+
+  samples_per_frame_ = audio_encoder_.GetNumberOfSamples();
+  PostAudioFormat();
+}
+
+void AudioEncoderInstance::OnAudioTrackConfigured(int32_t result) {
+  if (result == PP_ERROR_ABORTED)
+    return;
+  if (result != PP_OK) {
+    LogError(result, "Cannot configure audio track buffer duration");
+    return;
+  }
+
+  is_encoding_ = true;
+  audio_track_.GetBuffer(callback_factory_.NewCallbackWithOutput(
+      &AudioEncoderInstance::OnGetBuffer));
+  audio_encoder_.GetBitstreamBuffer(callback_factory_.NewCallbackWithOutput(
+      &AudioEncoderInstance::OnGetBitstreamBuffer));
+}
+
+void AudioEncoderInstance::GetEncoderBuffer(
+    const pp::AudioBuffer& track_buffer) {
+  audio_encoder_.GetBuffer(callback_factory_.NewCallbackWithOutput(
+      &AudioEncoderInstance::OnGetEncoderBuffer, track_buffer));
+}
+
+void AudioEncoderInstance::OnGetEncoderBuffer(int32_t result,
+                                              pp::AudioBuffer buffer,
+                                              pp::AudioBuffer track_buffer) {
+  const char* error(nullptr);
+
+  if (result != PP_OK)
+    error = "Cannot get encoder buffer";
+  if (buffer.GetDataBufferSize() != track_buffer.GetDataBufferSize()) {
+    result = PP_ERROR_FAILED;
+    error = "Invalid buffer size";
+  }
+
+  if (result == PP_OK) {
+    memcpy(buffer.GetDataBuffer(), track_buffer.GetDataBuffer(),
+           buffer.GetDataBufferSize());
+    audio_encoder_.Encode(buffer, callback_factory_.NewCallback(
+                                      &AudioEncoderInstance::OnEncodeDone));
+  } else {
+    LogError(result, error);
+    StopAudioTrack();
+  }
+  audio_track_.RecycleBuffer(track_buffer);
+}
+
+void AudioEncoderInstance::OnEncodeDone(int32_t result) {}
+
+void AudioEncoderInstance::OnGetBitstreamBuffer(
+    int32_t result,
+    const PP_AudioBitstreamBuffer& buffer) {
+  if (result == PP_ERROR_ABORTED)
+    return;
+  if (result != PP_OK) {
+    LogError(result, "Cannot get bitstream buffer");
+    return;
+  }
+
+  audio_encoder_.GetBitstreamBuffer(callback_factory_.NewCallbackWithOutput(
+      &AudioEncoderInstance::OnGetBitstreamBuffer));
+
+  PostDataMessage(buffer.buffer, buffer.size);
+  audio_encoder_.RecycleBitstreamBuffer(buffer);
+}
+
+pp::VarDictionary AudioEncoderInstance::NewCommand(const std::string& type) {
+  pp::VarDictionary dictionary;
+  dictionary.Set(pp::Var("command"), pp::Var(type));
+  return dictionary;
+}
+
+void AudioEncoderInstance::Log(const std::string& message) {
+  pp::VarDictionary dictionary = NewCommand("log");
+  dictionary.Set(pp::Var("message"), pp::Var(message));
+  PostMessage(dictionary);
+}
+
+void AudioEncoderInstance::LogError(int32_t error, const std::string& message) {
+  pp::VarDictionary dictionary = NewCommand("error");
+  std::string msg("Error: ");
+  msg.append(message);
+  msg.append(" : ");
+  msg.append(pp::Var(error).DebugString());
+  dictionary.Set(pp::Var("message"), pp::Var(msg));
+  PostMessage(dictionary);
+}
+
+void AudioEncoderInstance::PostAudioFormat() {
+  pp::VarDictionary dictionary = NewCommand("format");
+  dictionary.Set(pp::Var("channels"), pp::Var(static_cast<int32_t>(channels_)));
+  dictionary.Set(pp::Var("sample_rate"),
+                 pp::Var(static_cast<int32_t>(sample_rate_)));
+  dictionary.Set(pp::Var("sample_size"),
+                 pp::Var(static_cast<int32_t>(sample_size_)));
+  dictionary.Set(pp::Var("sample_per_frame"),
+                 pp::Var(static_cast<int32_t>(samples_per_frame_)));
+  PostMessage(dictionary);
+}
+
+void AudioEncoderInstance::PostDataMessage(const void* data, uint32_t size) {
+  pp::VarDictionary dictionary = NewCommand("data");
+  uint8_t* buffer;
+
+  pp::VarArrayBuffer array_buffer = pp::VarArrayBuffer(size);
+  buffer = static_cast<uint8_t*>(array_buffer.Map());
+  memcpy(buffer, data, size);
+  array_buffer.Unmap();
+
+  dictionary.Set(pp::Var("buffer"), array_buffer);
+
+  PostMessage(dictionary);
+}
+
+class AudioEncoderModule : public pp::Module {
+ public:
+  virtual pp::Instance* CreateInstance(PP_Instance instance) {
+    return new AudioEncoderInstance(instance);
+  }
+};
+
+}  // namespace
+
+namespace pp {
+
+// Factory function for your specialization of the Module object.
+Module* CreateModule() {
+  return new AudioEncoderModule();
+}
+
+}  // namespace pp