Add dummy I/O support.

Author: cloudwebrtc

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) {
@@ -41,9 +43,14 @@ rtc_shared_library("libwebrtc") {
 
   defines = [
     "USE_LIBYUV",
-    "WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE",
   ]
 
+  if(libwebrtc_dummy_audio_device) {
+    defines += [ "LIB_WEBRTC_USE_DUMMY_AUDIO_DEVICE" ]
+  } else {
+    defines += [ "WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE" ]
+  }
+
   if (is_win) {
     defines += [
       "LIB_WEBRTC_API_EXPORTS",
@@ -69,6 +76,7 @@ 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 +96,18 @@ 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 +142,6 @@ 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 +152,28 @@ 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 +260,7 @@ 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/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_desktop_device.h

@@ -1,5 +1,5 @@
-#ifndef LIB_WEBRTC_RTC_DESKTOP_DEVICE_HXX
-#define LIB_WEBRTC_RTC_DESKTOP_DEVICE_HXX
+#ifndef LIB_WEBRTC_RTC_DESKTOP_CAPTURE_DEVICE_HXX
+#define LIB_WEBRTC_RTC_DESKTOP_CAPTURE_DEVICE_HXX
 
 #include "rtc_types.h"
 

include/rtc_peerconnection_factory.h

@@ -4,7 +4,7 @@
 #include "rtc_audio_source.h"
 #include "rtc_audio_track.h"
 #include "rtc_types.h"
-#ifdef RTC_DESKTOP_DEVICE
+#ifdef RTC_DESKTOP_CAPTURE_DEVICE
 #include "rtc_desktop_device.h"
 #endif
 #include "rtc_media_stream.h"
@@ -31,24 +31,28 @@ class RTCPeerConnectionFactory : public RefCountInterface {
 
   virtual void Delete(scoped_refptr<RTCPeerConnection> peerconnection) = 0;
 
+#if !defined(LIB_WEBRTC_USE_DUMMY_AUDIO_DEVICE)
   virtual scoped_refptr<RTCAudioDevice> GetAudioDevice() = 0;
-
-  virtual scoped_refptr<RTCVideoDevice> GetVideoDevice() = 0;
-#ifdef RTC_DESKTOP_DEVICE
-  virtual scoped_refptr<RTCDesktopDevice> GetDesktopDevice() = 0;
 #endif
-  virtual scoped_refptr<RTCAudioSource> CreateAudioSource(
-      const string audio_source_label) = 0;
 
+#ifdef RTC_VIDEO_CAPTURE_DEVICE
+  virtual scoped_refptr<RTCVideoDevice> GetVideoDevice() = 0;
   virtual scoped_refptr<RTCVideoSource> CreateVideoSource(
       scoped_refptr<RTCVideoCapturer> capturer, const string video_source_label,
       scoped_refptr<RTCMediaConstraints> constraints) = 0;
-#ifdef RTC_DESKTOP_DEVICE
+#endif
+
+#ifdef RTC_DESKTOP_CAPTURE_DEVICE
+  virtual scoped_refptr<RTCDesktopDevice> GetDesktopDevice() = 0;
   virtual scoped_refptr<RTCVideoSource> CreateDesktopSource(
       scoped_refptr<RTCDesktopCapturer> capturer,
       const string video_source_label,
       scoped_refptr<RTCMediaConstraints> constraints) = 0;
 #endif
+
+  virtual scoped_refptr<RTCAudioSource> CreateAudioSource(
+      const string audio_source_label) = 0;
+
   virtual scoped_refptr<RTCAudioTrack> CreateAudioTrack(
       scoped_refptr<RTCAudioSource> source, const string track_id) = 0;
 

include/rtc_video_source.h

@@ -2,13 +2,24 @@
 #define LIB_WEBRTC_RTC_VIDEO_SOURCE_HXX
 
 #include "rtc_types.h"
+#include "rtc_video_frame.h"
 
 namespace libwebrtc {
 
 class RTCVideoSource : public RefCountInterface {
  public:
   ~RTCVideoSource() {}
 };
+
+class VirtualVideoCapturer : public RefCountInterface {
+ public:
+  LIB_WEBRTC_API static scoped_refptr<VirtualVideoCapturer> Create();
+
+  virtual void OnFrameCaptured(scoped_refptr<RTCVideoFrame> frame) = 0; 
+
+  virtual scoped_refptr<RTCVideoSource> source() = 0;
+};
+
 }  // namespace libwebrtc
 
 #endif  // LIB_WEBRTC_RTC_VIDEO_SOURCE_HXX