add RTCAudioFrame.

Author: cloudwebrtc

BUILD.gn

@@ -76,6 +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",
@@ -105,6 +106,8 @@ rtc_shared_library("libwebrtc") {
     "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",

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

src/rtc_audio_frame_impl.cc

@@ -0,0 +1,62 @@
+#include "rtc_audio_frame_impl.h"
+
+#include "audio/utility/audio_frame_operations.h"
+
+namespace libwebrtc {
+
+scoped_refptr<RTCAudioFrame> RTCAudioFrame::Create() {
+  return scoped_refptr<RTCAudioFrame>(
+      new RefCountedObject<RTCAudioFrameImpl>());
+}
+
+scoped_refptr<RTCAudioFrame> RTCAudioFrame::Create(
+    uint32_t timestamp, const int16_t* data, size_t samples_per_channel,
+    int sample_rate_hz, size_t num_channels /*= 1*/) {
+  auto audio_frame =
+      scoped_refptr<RTCAudioFrame>(new RefCountedObject<RTCAudioFrameImpl>());
+  audio_frame->UpdateFrame(timestamp, data, samples_per_channel, sample_rate_hz,
+                           num_channels);
+  return audio_frame;
+}
+
+RTCAudioFrameImpl::RTCAudioFrameImpl(webrtc::AudioFrame& buffer) {
+  buffer_.CopyFrom(buffer);
+}
+
+RTCAudioFrameImpl::~RTCAudioFrameImpl() {}
+
+void RTCAudioFrameImpl::UpdateFrame(uint32_t timestamp, const int16_t* data,
+                                    size_t samples_per_channel,
+                                    int sample_rate_hz,
+                                    size_t num_channels /*= 1*/) {
+  buffer_.UpdateFrame(timestamp, data, samples_per_channel, sample_rate_hz,
+                      webrtc::AudioFrame::kNormalSpeech,
+                      webrtc::AudioFrame::kVadUnknown, num_channels);
+}
+
+void RTCAudioFrameImpl::CopyFrom(const scoped_refptr<RTCAudioFrame> src) {
+  RTCAudioFrameImpl* frame = static_cast<RTCAudioFrameImpl*>(src.get());
+  buffer_.CopyFrom(frame->buffer_);
+}
+
+void RTCAudioFrameImpl::Add(const scoped_refptr<RTCAudioFrame> frame_to_add) {
+  RTCAudioFrameImpl* frame =
+      static_cast<RTCAudioFrameImpl*>(frame_to_add.get());
+  webrtc::AudioFrameOperations::Add(frame->buffer_, &buffer_);
+}
+
+void RTCAudioFrameImpl::Mute() { buffer_.Mute(); }
+
+const int16_t* RTCAudioFrameImpl::data() { return buffer_.data(); }
+
+size_t RTCAudioFrameImpl::samples_per_channel() {
+  return buffer_.samples_per_channel_;
+}
+
+int RTCAudioFrameImpl::sample_rate_hz() { return buffer_.sample_rate_hz_; }
+
+size_t RTCAudioFrameImpl::num_channels() { return buffer_.num_channels_; }
+
+uint32_t RTCAudioFrameImpl::timestamp() { return buffer_.timestamp_; }
+
+}  // namespace libwebrtc

src/rtc_audio_frame_impl.h

@@ -0,0 +1,37 @@
+#include "api/audio/audio_frame.h"
+#include "include/rtc_audio_frame.h"
+
+namespace libwebrtc {
+
+class RTCAudioFrameImpl : public RTCAudioFrame {
+ public:
+  RTCAudioFrameImpl() {}
+  RTCAudioFrameImpl(webrtc::AudioFrame& buffer);
+  ~RTCAudioFrameImpl();
+
+ public:
+  virtual void UpdateFrame(uint32_t timestamp, const int16_t* data,
+                           size_t samples_per_channel, int sample_rate_hz,
+                           size_t num_channels = 1) override;
+
+  virtual void CopyFrom(const scoped_refptr<RTCAudioFrame> src) override;
+
+  virtual void Add(const scoped_refptr<RTCAudioFrame> frame_to_add) override;
+
+  virtual void Mute() override;
+
+  virtual const int16_t* data() override;
+
+  virtual size_t samples_per_channel() override;
+
+  virtual int sample_rate_hz() override;
+
+  virtual size_t num_channels() override;
+
+  virtual uint32_t timestamp() override;
+
+ private:
+  webrtc::AudioFrame buffer_;
+};
+
+}  // namespace libwebrtc