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authorDaniel Baumann <daniel.baumann@progress-linux.org>2024-04-07 19:33:14 +0000
committerDaniel Baumann <daniel.baumann@progress-linux.org>2024-04-07 19:33:14 +0000
commit36d22d82aa202bb199967e9512281e9a53db42c9 (patch)
tree105e8c98ddea1c1e4784a60a5a6410fa416be2de /third_party/libwebrtc/audio/voip/audio_egress.h
parentInitial commit. (diff)
downloadfirefox-esr-36d22d82aa202bb199967e9512281e9a53db42c9.tar.xz
firefox-esr-36d22d82aa202bb199967e9512281e9a53db42c9.zip
Adding upstream version 115.7.0esr.upstream/115.7.0esr
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'third_party/libwebrtc/audio/voip/audio_egress.h')
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+/*
+ * Copyright (c) 2020 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 AUDIO_VOIP_AUDIO_EGRESS_H_
+#define AUDIO_VOIP_AUDIO_EGRESS_H_
+
+#include <memory>
+#include <string>
+
+#include "api/audio_codecs/audio_format.h"
+#include "api/sequence_checker.h"
+#include "api/task_queue/task_queue_factory.h"
+#include "audio/audio_level.h"
+#include "audio/utility/audio_frame_operations.h"
+#include "call/audio_sender.h"
+#include "modules/audio_coding/include/audio_coding_module.h"
+#include "modules/rtp_rtcp/include/report_block_data.h"
+#include "modules/rtp_rtcp/source/rtp_rtcp_interface.h"
+#include "modules/rtp_rtcp/source/rtp_sender_audio.h"
+#include "rtc_base/synchronization/mutex.h"
+#include "rtc_base/task_queue.h"
+#include "rtc_base/time_utils.h"
+
+namespace webrtc {
+
+// AudioEgress receives input samples from AudioDeviceModule via
+// AudioTransportImpl through AudioSender interface. Once it encodes the sample
+// via selected encoder through AudioPacketizationCallback interface, the
+// encoded payload will be packetized by the RTP stack, resulting in ready to
+// send RTP packet to remote endpoint.
+//
+// TaskQueue is used to encode and send RTP asynchrounously as some OS platform
+// uses the same thread for both audio input and output sample deliveries which
+// can affect audio quality.
+//
+// Note that this class is originally based on ChannelSend in
+// audio/channel_send.cc with non-audio related logic trimmed as aimed for
+// smaller footprint.
+class AudioEgress : public AudioSender, public AudioPacketizationCallback {
+ public:
+ AudioEgress(RtpRtcpInterface* rtp_rtcp,
+ Clock* clock,
+ TaskQueueFactory* task_queue_factory);
+ ~AudioEgress() override;
+
+ // Set the encoder format and payload type for AudioCodingModule.
+ // It's possible to change the encoder type during its active usage.
+ // `payload_type` must be the type that is negotiated with peer through
+ // offer/answer.
+ void SetEncoder(int payload_type,
+ const SdpAudioFormat& encoder_format,
+ std::unique_ptr<AudioEncoder> encoder);
+
+ // Start or stop sending operation of AudioEgress. This will start/stop
+ // the RTP stack also causes encoder queue thread to start/stop
+ // processing input audio samples. StartSend will return false if
+ // a send codec has not been set.
+ bool StartSend();
+ void StopSend();
+
+ // Query the state of the RTP stack. This returns true if StartSend()
+ // called and false if StopSend() is called.
+ bool IsSending() const;
+
+ // Enable or disable Mute state.
+ void SetMute(bool mute);
+
+ // Retrieve current encoder format info. This returns encoder format set
+ // by SetEncoder() and if encoder is not set, this will return nullopt.
+ absl::optional<SdpAudioFormat> GetEncoderFormat() const {
+ MutexLock lock(&lock_);
+ return encoder_format_;
+ }
+
+ // Register the payload type and sample rate for DTMF (RFC 4733) payload.
+ void RegisterTelephoneEventType(int rtp_payload_type, int sample_rate_hz);
+
+ // Send DTMF named event as specified by
+ // https://tools.ietf.org/html/rfc4733#section-3.2
+ // `duration_ms` specifies the duration of DTMF packets that will be emitted
+ // in place of real RTP packets instead.
+ // This will return true when requested dtmf event is successfully scheduled
+ // otherwise false when the dtmf queue reached maximum of 20 events.
+ bool SendTelephoneEvent(int dtmf_event, int duration_ms);
+
+ // See comments on LevelFullRange, TotalEnergy, TotalDuration from
+ // audio/audio_level.h.
+ int GetInputAudioLevel() const { return input_audio_level_.LevelFullRange(); }
+ double GetInputTotalEnergy() const {
+ return input_audio_level_.TotalEnergy();
+ }
+ double GetInputTotalDuration() const {
+ return input_audio_level_.TotalDuration();
+ }
+
+ // Implementation of AudioSender interface.
+ void SendAudioData(std::unique_ptr<AudioFrame> audio_frame) override;
+
+ // Implementation of AudioPacketizationCallback interface.
+ int32_t SendData(AudioFrameType frame_type,
+ uint8_t payload_type,
+ uint32_t timestamp,
+ const uint8_t* payload_data,
+ size_t payload_size) override;
+
+ private:
+ void SetEncoderFormat(const SdpAudioFormat& encoder_format) {
+ MutexLock lock(&lock_);
+ encoder_format_ = encoder_format;
+ }
+
+ mutable Mutex lock_;
+
+ // Current encoder format selected by caller.
+ absl::optional<SdpAudioFormat> encoder_format_ RTC_GUARDED_BY(lock_);
+
+ // Synchronization is handled internally by RtpRtcp.
+ RtpRtcpInterface* const rtp_rtcp_;
+
+ // Synchronization is handled internally by RTPSenderAudio.
+ RTPSenderAudio rtp_sender_audio_;
+
+ // Synchronization is handled internally by AudioCodingModule.
+ const std::unique_ptr<AudioCodingModule> audio_coding_;
+
+ // Synchronization is handled internally by voe::AudioLevel.
+ voe::AudioLevel input_audio_level_;
+
+ // Struct that holds all variables used by encoder task queue.
+ struct EncoderContext {
+ // Offset used to mark rtp timestamp in sample rate unit in
+ // newly received audio frame from AudioTransport.
+ uint32_t frame_rtp_timestamp_ = 0;
+
+ // Flag to track mute state from caller. `previously_muted_` is used to
+ // track previous state as part of input to AudioFrameOperations::Mute
+ // to implement fading effect when (un)mute is invoked.
+ bool mute_ = false;
+ bool previously_muted_ = false;
+ };
+
+ EncoderContext encoder_context_ RTC_GUARDED_BY(encoder_queue_);
+
+ // Defined last to ensure that there are no running tasks when the other
+ // members are destroyed.
+ rtc::TaskQueue encoder_queue_;
+};
+
+} // namespace webrtc
+
+#endif // AUDIO_VOIP_AUDIO_EGRESS_H_