3e6db2321c
This is the last piece of the old directory layout of the modules. Duplicated header files are left in audio_coding/main/include until downstream code is updated to the new location. They have pragma warnings added to them and identical header guards as the new headers to avoid breaking things. BUG=webrtc:5095 TESTED=Passing compile-trybots with --clobber flag: git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel --bot=linux_gn_rel --bot=win_x64_gn_rel --bot=mac_x64_gn_rel --bot=android_gn_rel -m tryserver.webrtc NOTRY=True NOPRESUBMIT=True Review URL: https://codereview.webrtc.org/1481493004 Cr-Commit-Position: refs/heads/master@{#10803}
121 lines
4.3 KiB
C++
121 lines
4.3 KiB
C++
/*
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* Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef WEBRTC_MODULES_AUDIO_CODING_ACM2_INITIAL_DELAY_MANAGER_H_
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#define WEBRTC_MODULES_AUDIO_CODING_ACM2_INITIAL_DELAY_MANAGER_H_
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#include "webrtc/base/scoped_ptr.h"
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#include "webrtc/modules/include/module_common_types.h"
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namespace webrtc {
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namespace acm2 {
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class InitialDelayManager {
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public:
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enum PacketType {
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kUndefinedPacket, kCngPacket, kAvtPacket, kAudioPacket, kSyncPacket };
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// Specifies a stream of sync-packets.
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struct SyncStream {
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SyncStream()
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: num_sync_packets(0),
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receive_timestamp(0),
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timestamp_step(0) {
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memset(&rtp_info, 0, sizeof(rtp_info));
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}
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int num_sync_packets;
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// RTP header of the first sync-packet in the sequence.
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WebRtcRTPHeader rtp_info;
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// Received timestamp of the first sync-packet in the sequence.
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uint32_t receive_timestamp;
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// Samples per packet.
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uint32_t timestamp_step;
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};
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InitialDelayManager(int initial_delay_ms, int late_packet_threshold);
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// Update with the last received RTP header, |header|, and received timestamp,
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// |received_timestamp|. |type| indicates the packet type. If codec is changed
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// since the last time |new_codec| should be true. |sample_rate_hz| is the
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// decoder's sampling rate in Hz. |header| has a field to store sampling rate
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// but we are not sure if that is properly set at the send side, and |header|
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// is declared constant in the caller of this function
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// (AcmReceiver::InsertPacket()). |sync_stream| contains information required
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// to generate a stream of sync packets.
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void UpdateLastReceivedPacket(const WebRtcRTPHeader& header,
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uint32_t receive_timestamp,
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PacketType type,
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bool new_codec,
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int sample_rate_hz,
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SyncStream* sync_stream);
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// Based on the last received timestamp and given the current timestamp,
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// sequence of late (or perhaps missing) packets is computed.
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void LatePackets(uint32_t timestamp_now, SyncStream* sync_stream);
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// Get playout timestamp.
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// Returns true if the timestamp is valid (when buffering), otherwise false.
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bool GetPlayoutTimestamp(uint32_t* playout_timestamp);
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// True if buffered audio is less than the given initial delay (specified at
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// the constructor). Buffering might be disabled by the client of this class.
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bool buffering() { return buffering_; }
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// Disable buffering in the class.
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void DisableBuffering();
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// True if any packet received for buffering.
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bool PacketBuffered() { return last_packet_type_ != kUndefinedPacket; }
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private:
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static const uint8_t kInvalidPayloadType = 0xFF;
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// Update playout timestamps. While buffering, this is about
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// |initial_delay_ms| millisecond behind the latest received timestamp.
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void UpdatePlayoutTimestamp(const RTPHeader& current_header,
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int sample_rate_hz);
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// Record an RTP headr and related parameter
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void RecordLastPacket(const WebRtcRTPHeader& rtp_info,
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uint32_t receive_timestamp,
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PacketType type);
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PacketType last_packet_type_;
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WebRtcRTPHeader last_packet_rtp_info_;
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uint32_t last_receive_timestamp_;
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uint32_t timestamp_step_;
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uint8_t audio_payload_type_;
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const int initial_delay_ms_;
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int buffered_audio_ms_;
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bool buffering_;
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// During the initial phase where packets are being accumulated and silence
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// is played out, |playout_ts| is a timestamp which is equal to
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// |initial_delay_ms_| milliseconds earlier than the most recently received
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// RTP timestamp.
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uint32_t playout_timestamp_;
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// If the number of late packets exceed this value (computed based on current
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// timestamp and last received timestamp), sequence of sync-packets is
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// specified.
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const int late_packet_threshold_;
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};
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} // namespace acm2
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} // namespace webrtc
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#endif // WEBRTC_MODULES_AUDIO_CODING_ACM2_INITIAL_DELAY_MANAGER_H_
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