Adding upstream version 124.0.1.upstream/124.0.1

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>

1 files changed, 116 insertions, 0 deletions

diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sender_audio.h b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sender_audio.h
new file mode 100644
index 0000000000..662f908216
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_sender_audio.h
@@ -0,0 +1,116 @@
+/*
+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_RTP_RTCP_SOURCE_RTP_SENDER_AUDIO_H_
+#define MODULES_RTP_RTCP_SOURCE_RTP_SENDER_AUDIO_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include <memory>
+
+#include "absl/strings/string_view.h"
+#include "modules/audio_coding/include/audio_coding_module_typedefs.h"
+#include "modules/rtp_rtcp/source/absolute_capture_time_sender.h"
+#include "modules/rtp_rtcp/source/dtmf_queue.h"
+#include "modules/rtp_rtcp/source/rtp_sender.h"
+#include "rtc_base/one_time_event.h"
+#include "rtc_base/synchronization/mutex.h"
+#include "rtc_base/thread_annotations.h"
+#include "system_wrappers/include/clock.h"
+
+namespace webrtc {
+
+class RTPSenderAudio {
+ public:
+  RTPSenderAudio(Clock* clock, RTPSender* rtp_sender);
+
+  RTPSenderAudio() = delete;
+  RTPSenderAudio(const RTPSenderAudio&) = delete;
+  RTPSenderAudio& operator=(const RTPSenderAudio&) = delete;
+
+  ~RTPSenderAudio();
+
+  int32_t RegisterAudioPayload(absl::string_view payload_name,
+                               int8_t payload_type,
+                               uint32_t frequency,
+                               size_t channels,
+                               uint32_t rate);
+
+  struct RtpAudioFrame {
+    AudioFrameType type = AudioFrameType::kAudioFrameSpeech;
+    rtc::ArrayView<const uint8_t> payload;
+
+    // Payload id to write to the payload type field of the rtp packet.
+    int payload_id = -1;
+
+    // capture time of the audio frame represented as rtp timestamp.
+    uint32_t rtp_timestamp = 0;
+
+    // capture time of the audio frame in the same epoch as `clock->CurrentTime`
+    absl::optional<Timestamp> capture_time;
+
+    // Audio level in dBov for
+    // header-extension-for-audio-level-indication.
+    // Valid range is [0,127]. Actual value is negative.
+    absl::optional<int> audio_level_dbov;
+  };
+  bool SendAudio(const RtpAudioFrame& frame);
+
+  // Send a DTMF tone using RFC 2833 (4733)
+  int32_t SendTelephoneEvent(uint8_t key, uint16_t time_ms, uint8_t level);
+
+ protected:
+  bool SendTelephoneEventPacket(
+      bool ended,
+      uint32_t dtmf_timestamp,
+      uint16_t duration,
+      bool marker_bit);  // set on first packet in talk burst
+
+  bool MarkerBit(AudioFrameType frame_type, int8_t payload_type);
+
+ private:
+  Clock* const clock_ = nullptr;
+  RTPSender* const rtp_sender_ = nullptr;
+
+  Mutex send_audio_mutex_;
+
+  // DTMF.
+  bool dtmf_event_is_on_ = false;
+  bool dtmf_event_first_packet_sent_ = false;
+  int8_t dtmf_payload_type_ RTC_GUARDED_BY(send_audio_mutex_) = -1;
+  uint32_t dtmf_payload_freq_ RTC_GUARDED_BY(send_audio_mutex_) = 8000;
+  uint32_t dtmf_timestamp_ = 0;
+  uint32_t dtmf_length_samples_ = 0;
+  int64_t dtmf_time_last_sent_ = 0;
+  uint32_t dtmf_timestamp_last_sent_ = 0;
+  DtmfQueue::Event dtmf_current_event_;
+  DtmfQueue dtmf_queue_;
+
+  // VAD detection, used for marker bit.
+  bool inband_vad_active_ RTC_GUARDED_BY(send_audio_mutex_) = false;
+  int8_t cngnb_payload_type_ RTC_GUARDED_BY(send_audio_mutex_) = -1;
+  int8_t cngwb_payload_type_ RTC_GUARDED_BY(send_audio_mutex_) = -1;
+  int8_t cngswb_payload_type_ RTC_GUARDED_BY(send_audio_mutex_) = -1;
+  int8_t cngfb_payload_type_ RTC_GUARDED_BY(send_audio_mutex_) = -1;
+  int8_t last_payload_type_ RTC_GUARDED_BY(send_audio_mutex_) = -1;
+
+  OneTimeEvent first_packet_sent_;
+
+  absl::optional<int> encoder_rtp_timestamp_frequency_
+      RTC_GUARDED_BY(send_audio_mutex_);
+
+  AbsoluteCaptureTimeSender absolute_capture_time_sender_
+      RTC_GUARDED_BY(send_audio_mutex_);
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_RTP_RTCP_SOURCE_RTP_SENDER_AUDIO_H_