Feat/support dummy io

BUILD.gn

@@ -2,7 +2,9 @@ import("../webrtc.gni")
 
 declare_args() {
   libwebrtc_intel_media_sdk = false
-  libwebrtc_desktop_capture = true
+  libwebrtc_desktop_capture = false
+  libwebrtc_video_capture = false
+  libwebrtc_dummy_audio_device = true
 }
 
 if (is_android) {
@@ -34,16 +36,27 @@ config("external_config") {
 
 rtc_shared_library("libwebrtc") {
   libs = []
+  visibility = [
+      "//:default",
+  ]
+  #complete_static_lib = true
+  #suppressed_configs += [ "//build/config/compiler:thin_archive" ]
   include_dirs = [
     ".",
     "include",
   ]
 
   defines = [
     "USE_LIBYUV",
-    "WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE",
+    "SHARED_BUILD",
+    #"USE_PORTABLE_TYPES",
   ]
 
+  if(!libwebrtc_dummy_audio_device) {
+    defines += [ "RTC_AUDIO_DEVICE" ]
+    defines += [ "WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE" ]
+  }
+
   if (is_win) {
     defines += [
       "LIB_WEBRTC_API_EXPORTS",
@@ -69,6 +82,7 @@ rtc_shared_library("libwebrtc") {
     "include/base/scoped_ref_ptr.h",
     "include/libwebrtc.h",
     "include/rtc_audio_device.h",
+    "include/rtc_audio_frame.h",
     "include/rtc_audio_source.h",
     "include/rtc_audio_track.h",
     "include/rtc_data_channel.h",
@@ -88,21 +102,18 @@ rtc_shared_library("libwebrtc") {
     "include/rtc_rtp_transceiver.h",
     "include/rtc_session_description.h",
     "include/rtc_types.h",
-    "include/rtc_video_device.h",
     "include/rtc_video_frame.h",
     "include/rtc_video_renderer.h",
     "include/rtc_video_source.h",
     "include/rtc_video_track.h",
     "include/helper.h",
     "src/helper.cc",
     "src/base/portable.cc",
-    "src/internal/vcm_capturer.cc",
-    "src/internal/vcm_capturer.h",
-    "src/internal/video_capturer.cc",
-    "src/internal/video_capturer.h",
     "src/libwebrtc.cc",
     "src/rtc_audio_device_impl.cc",
     "src/rtc_audio_device_impl.h",
+    "src/rtc_audio_frame_impl.cc",
+    "src/rtc_audio_frame_impl.h",
     "src/rtc_audio_source_impl.cc",
     "src/rtc_audio_source_impl.h",
     "src/rtc_audio_track_impl.cc",
@@ -137,8 +148,6 @@ rtc_shared_library("libwebrtc") {
     "src/rtc_rtp_transceiver_impl.h",
     "src/rtc_session_description_impl.cc",
     "src/rtc_session_description_impl.h",
-    "src/rtc_video_device_impl.cc",
-    "src/rtc_video_device_impl.h",
     "src/rtc_video_frame_impl.cc",
     "src/rtc_video_frame_impl.h",
     "src/rtc_video_sink_adapter.cc",
@@ -149,6 +158,28 @@ rtc_shared_library("libwebrtc") {
     "src/rtc_video_track_impl.h",
   ]
 
+  if(libwebrtc_dummy_audio_device) {
+    sources += [
+      "src/audio_device_dummy.cc",
+      "src/audio_device_dummy.h",
+    ]
+  }
+
+
+  # video capture device
+  if (libwebrtc_video_capture) {
+    defines += [ "RTC_VIDEO_CAPTURE_DEVICE" ]
+    sources += [
+      "include/rtc_video_capturer.h",
+      "src/internal/video_capturer.h",
+      "src/internal/video_capturer.cc",
+      "src/internal/vcm_capturer.cc",
+      "src/internal/vcm_capturer.h",
+      "src/rtc_video_device_impl.cc",
+      "src/rtc_video_device_impl.h",
+    ]
+  }
+
   # intel media sdk
   if (is_win && libwebrtc_intel_media_sdk) {
     sources += [
@@ -235,7 +266,7 @@ rtc_shared_library("libwebrtc") {
 
   # screen capture device
   if (libwebrtc_desktop_capture) {
-    defines += [ "RTC_DESKTOP_DEVICE" ]
+    defines += [ "RTC_DESKTOP_CAPTURE_DEVICE" ]
     sources += [
       "include/rtc_desktop_capturer.h",
       "include/rtc_desktop_device.h",

include/base/portable.h

@@ -1,15 +1,7 @@
 #ifndef INFINISPAN_HOTROD_PORTABLE_H
 #define INFINISPAN_HOTROD_PORTABLE_H
 
-#ifdef LIB_WEBRTC_API_EXPORTS
-#define LIB_PORTABLE_API __declspec(dllexport)
-#elif defined(LIB_WEBRTC_API_DLL)
-#define LIB_PORTABLE_API __declspec(dllimport)
-#elif !defined(WIN32)
-#define LIB_PORTABLE_API __attribute__((visibility("default")))
-#else
-#define LIB_PORTABLE_API
-#endif
+#include "../rtc_config.h"
 
 #include <cstdlib>
 #include <cstring>
@@ -45,9 +37,9 @@ class string {
   size_t m_length;
 
  public:
-  LIB_PORTABLE_API string();
-  LIB_PORTABLE_API void init(const char* str, size_t len);
-  LIB_PORTABLE_API void destroy();
+  LIB_WEBRTC_API string();
+  LIB_WEBRTC_API void init(const char* str, size_t len);
+  LIB_WEBRTC_API void destroy();
 
   inline string(const char* str) { init(str, strlen(str)); }
 
@@ -63,7 +55,7 @@ class string {
     return *this;
   }
 
-  LIB_PORTABLE_API ~string();
+  LIB_WEBRTC_API ~string();
 
   inline string& operator=(const std::string& str) {
     destroy();
@@ -73,7 +65,7 @@ class string {
 
   inline size_t size() { return m_length; }
 
-  inline const char* c_string() const {
+  inline const char* c_str() const {
     return m_dynamic == 0 ? m_buf : m_dynamic;
   }
 
@@ -82,8 +74,6 @@ class string {
   }
 };
 
-inline std::string to_std_string(const string& str) { return str.std_string(); }
-
 template <typename T>
 class identity {
   T operator()(const T& x) { return x; }

include/rtc_audio_device.h

@@ -12,7 +12,7 @@ namespace libwebrtc {
  */
 class RTCAudioDevice : public RefCountInterface {
  public:
-  typedef fixed_size_function<void()> OnDeviceChangeCallback;
+  typedef callback_function_t<void()> OnDeviceChangeCallback;
 
  public:
   static const int kAdmMaxDeviceNameSize = 128;

include/rtc_audio_frame.h

@@ -1,108 +1,40 @@
-#ifndef AUDIO_FRAME_HXX
-#define AUDIO_FRAME_HXX
+#ifndef LIB_WEBRTC_RTC_AUDIO_FRAME_HXX
+#define LIB_WEBRTC_RTC_AUDIO_FRAME_HXX
 
-#include "media_manager_types.h"
+#include "rtc_types.h"
 
-namespace b2bua {
+namespace libwebrtc {
 
-class AudioFrame {
+class RTCAudioFrame : public RefCountInterface {
  public:
-  /**
-   * @brief Creates a new instance of AudioFrame.
-   * @return AudioFrame*: a pointer to the newly created AudioFrame.
-   */
-  MEDIA_MANAGER_API static AudioFrame* Create();
+ LIB_WEBRTC_API static scoped_refptr<RTCAudioFrame> Create();
 
-  /**
-   * @brief Creates a new instance of AudioFrame with specified parameters.
-   * @param id: the unique identifier of the frame.
-   * @param timestamp: the timestamp of the frame.
-   * @param data: a pointer to the audio data buffer.
-   * @param samples_per_channel: the number of samples per channel.
-   * @param sample_rate_hz: the sample rate in Hz.
-   * @param num_channels: the number of audio channels.
-   * @return AudioFrame*: a pointer to the newly created AudioFrame.
-   */
-  MEDIA_MANAGER_API static AudioFrame* Create(int id, uint32_t timestamp,
-                                              const int16_t* data,
-                                              size_t samples_per_channel,
-                                              int sample_rate_hz,
-                                              size_t num_channels = 1);
-
-  /**
-   * @brief Releases the memory of this AudioFrame.
-   */
-  virtual void Release() = 0;
+ LIB_WEBRTC_API static scoped_refptr<RTCAudioFrame> Create(
+      uint32_t timestamp, const int16_t* data, size_t samples_per_channel,
+      int sample_rate_hz, size_t num_channels = 1);
 
  public:
-  /**
-   * @brief Updates the audio frame with specified parameters.
-   * @param id: the unique identifier of the frame.
-   * @param timestamp: the timestamp of the frame.
-   * @param data: a pointer to the audio data buffer.
-   * @param samples_per_channel: the number of samples per channel.
-   * @param sample_rate_hz: the sample rate in Hz.
-   * @param num_channels: the number of audio channels.
-   */
-  virtual void UpdateFrame(int id, uint32_t timestamp, const int16_t* data,
+  virtual void UpdateFrame(uint32_t timestamp, const int16_t* data,
                            size_t samples_per_channel, int sample_rate_hz,
                            size_t num_channels = 1) = 0;
 
-  /**
-   * @brief Copies the contents of another AudioFrame.
-   * @param src: the source AudioFrame to copy from.
-   */
-  virtual void CopyFrom(const AudioFrame& src) = 0;
+  virtual void CopyFrom(const scoped_refptr<RTCAudioFrame> src) = 0;
 
-  /**
-   * @brief Adds another AudioFrame to this one.
-   * @param frame_to_add: the AudioFrame to add.
-   */
-  virtual void Add(const AudioFrame& frame_to_add) = 0;
+  virtual void Add(const scoped_refptr<RTCAudioFrame> frame_to_add) = 0;
 
-  /**
-   * @brief Mutes the audio data in this AudioFrame.
-   */
   virtual void Mute() = 0;
 
-  /**
-   * @brief Returns a pointer to the audio data buffer.
-   * @return const int16_t*: a pointer to the audio data buffer.
-   */
   virtual const int16_t* data() = 0;
 
-  /**
-   * @brief Returns the number of samples per channel.
-   * @return size_t: the number of samples per channel.
-   */
   virtual size_t samples_per_channel() = 0;
 
-  /**
-   * @brief Returns the sample rate in Hz.
-   * @return int: the sample rate in Hz.
-   */
   virtual int sample_rate_hz() = 0;
 
-  /**
-   * @brief Returns the number of audio channels.
-   * @return size_t: the number of audio channels.
-   */
   virtual size_t num_channels() = 0;
 
-  /**
-   * @brief Returns the timestamp of the AudioFrame.
-   * @return uint32_t: the timestamp of the AudioFrame.
-   */
   virtual uint32_t timestamp() = 0;
-
-  /**
-   * @brief Returns the unique identifier of the AudioFrame.
-   * @return int: the unique identifier of the AudioFrame.
-   */
-
-  virtual int id() = 0;
 };
 
-};  // namespace b2bua
+}  // namespace libwebrtc
 
 #endif

include/rtc_audio_source.h

@@ -2,6 +2,7 @@
 #define LIB_WEBRTC_RTC_AUDIO_SOURCE_HXX
 
 #include "rtc_types.h"
+#include "rtc_audio_frame.h"
 
 namespace libwebrtc {
 
@@ -20,6 +21,21 @@ class RTCAudioSource : public RefCountInterface {
   virtual ~RTCAudioSource() {}
 };
 
+class VirtualAudioCapturer : public RefCountInterface {
+ public:
+  LIB_WEBRTC_API static scoped_refptr<VirtualAudioCapturer> Create();
+
+  virtual void OnFrame(scoped_refptr<RTCAudioFrame> data) = 0;
+
+  virtual void OnData(const void* audio_data,
+                      int bits_per_sample,
+                      int sample_rate,
+                      size_t number_of_channels,
+                      size_t number_of_frames) = 0;
+
+  virtual scoped_refptr<RTCAudioSource> source() = 0;
+};
+
 }  // namespace libwebrtc
 
 #endif  // LIB_WEBRTC_RTC_AUDIO_TRACK_HXX

include/rtc_audio_track.h

@@ -1,11 +1,24 @@
 #ifndef LIB_WEBRTC_RTC_AUDIO_TRACK_HXX
 #define LIB_WEBRTC_RTC_AUDIO_TRACK_HXX
 
+#include "rtc_audio_frame.h"
 #include "rtc_media_track.h"
 #include "rtc_types.h"
 
 namespace libwebrtc {
 
+template <typename AudioFrameT>
+class RTCAudioRenderer {
+ public:
+  virtual void OnFrame(AudioFrameT frame) = 0;
+
+  virtual void OnData(const void* audio_data, int bits_per_sample,
+                            int sample_rate, size_t number_of_channels,
+                            size_t number_of_frames) = 0;
+ protected:
+  virtual ~RTCAudioRenderer() {}
+};
+
 /**
  * The RTCAudioTrack class represents an audio track in WebRTC.
  * Audio tracks are used to transmit audio data over a WebRTC peer connection.
@@ -17,6 +30,12 @@ class RTCAudioTrack : public RTCMediaTrack {
   // volume in [0-10]
   virtual void SetVolume(double volume) = 0;
 
+  virtual void AddAudioSink(
+      RTCAudioRenderer<scoped_refptr<RTCAudioFrame>>* sink) = 0;
+
+  virtual void RemoveAudioSink(
+      RTCAudioRenderer<scoped_refptr<RTCAudioFrame>>* sink) = 0;
+
  protected:
   /**
    * The destructor for the RTCAudioTrack class.

include/rtc_config.h

@@ -0,0 +1,22 @@
+#ifndef WEBRTC_CONFIG_H_
+#define WEBRTC_CONFIG_H_
+
+#ifdef SHARED_BUILD
+
+#ifdef LIB_WEBRTC_API_EXPORTS
+#define LIB_WEBRTC_API __declspec(dllexport)
+#elif defined(LIB_WEBRTC_API_DLL)
+#define LIB_WEBRTC_API __declspec(dllimport)
+#elif !defined(WIN32)
+#define LIB_WEBRTC_API __attribute__((visibility("default")))
+#else
+#define LIB_WEBRTC_API
+#endif
+
+#else
+
+#define LIB_WEBRTC_API 
+
+#endif
+
+#endif // WEBRTC_CONFIG_H_