ee2bac26dd
Instead of separate pointer and size arguments. Review URL: https://codereview.webrtc.org/1429943004 Cr-Commit-Position: refs/heads/master@{#10606}
310 lines
10 KiB
C++
310 lines
10 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_MAIN_ACM2_ACM_RECEIVER_H_
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#define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_RECEIVER_H_
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#include <map>
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#include <vector>
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#include "webrtc/base/array_view.h"
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#include "webrtc/base/scoped_ptr.h"
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#include "webrtc/base/thread_annotations.h"
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#include "webrtc/common_audio/vad/include/webrtc_vad.h"
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#include "webrtc/engine_configurations.h"
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#include "webrtc/modules/audio_coding/main/include/audio_coding_module.h"
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#include "webrtc/modules/audio_coding/main/acm2/acm_resampler.h"
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#include "webrtc/modules/audio_coding/main/acm2/call_statistics.h"
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#include "webrtc/modules/audio_coding/main/acm2/initial_delay_manager.h"
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#include "webrtc/modules/audio_coding/neteq/include/neteq.h"
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#include "webrtc/modules/include/module_common_types.h"
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#include "webrtc/typedefs.h"
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namespace webrtc {
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struct CodecInst;
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class CriticalSectionWrapper;
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class NetEq;
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namespace acm2 {
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class AcmReceiver {
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public:
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struct Decoder {
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int acm_codec_id;
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uint8_t payload_type;
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// This field is meaningful for codecs where both mono and
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// stereo versions are registered under the same ID.
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int channels;
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int sample_rate_hz;
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};
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// Constructor of the class
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explicit AcmReceiver(const AudioCodingModule::Config& config);
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// Destructor of the class.
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~AcmReceiver();
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//
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// Inserts a payload with its associated RTP-header into NetEq.
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//
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// Input:
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// - rtp_header : RTP header for the incoming payload containing
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// information about payload type, sequence number,
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// timestamp, SSRC and marker bit.
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// - incoming_payload : Incoming audio payload.
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// - length_payload : Length of incoming audio payload in bytes.
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//
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// Return value : 0 if OK.
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// <0 if NetEq returned an error.
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//
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int InsertPacket(const WebRtcRTPHeader& rtp_header,
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rtc::ArrayView<const uint8_t> incoming_payload);
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//
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// Asks NetEq for 10 milliseconds of decoded audio.
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//
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// Input:
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// -desired_freq_hz : specifies the sampling rate [Hz] of the output
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// audio. If set -1 indicates to resampling is
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// is required and the audio returned at the
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// sampling rate of the decoder.
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//
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// Output:
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// -audio_frame : an audio frame were output data and
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// associated parameters are written to.
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//
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// Return value : 0 if OK.
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// -1 if NetEq returned an error.
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//
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int GetAudio(int desired_freq_hz, AudioFrame* audio_frame);
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//
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// Adds a new codec to the NetEq codec database.
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//
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// Input:
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// - acm_codec_id : ACM codec ID; -1 means external decoder.
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// - payload_type : payload type.
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// - sample_rate_hz : sample rate.
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// - audio_decoder : pointer to a decoder object. If it's null, then
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// NetEq will internally create a decoder object
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// based on the value of |acm_codec_id| (which
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// mustn't be -1). Otherwise, NetEq will use the
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// given decoder for the given payload type. NetEq
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// won't take ownership of the decoder; it's up to
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// the caller to delete it when it's no longer
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// needed.
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//
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// Providing an existing decoder object here is
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// necessary for external decoders, but may also be
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// used for built-in decoders if NetEq doesn't have
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// all the info it needs to construct them properly
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// (e.g. iSAC, where the decoder needs to be paired
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// with an encoder).
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//
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// Return value : 0 if OK.
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// <0 if NetEq returned an error.
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//
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int AddCodec(int acm_codec_id,
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uint8_t payload_type,
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int channels,
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int sample_rate_hz,
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AudioDecoder* audio_decoder);
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//
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// Sets a minimum delay for packet buffer. The given delay is maintained,
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// unless channel condition dictates a higher delay.
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//
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// Input:
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// - delay_ms : minimum delay in milliseconds.
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//
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// Return value : 0 if OK.
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// <0 if NetEq returned an error.
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//
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int SetMinimumDelay(int delay_ms);
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//
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// Sets a maximum delay [ms] for the packet buffer. The target delay does not
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// exceed the given value, even if channel condition requires so.
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//
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// Input:
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// - delay_ms : maximum delay in milliseconds.
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//
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// Return value : 0 if OK.
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// <0 if NetEq returned an error.
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//
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int SetMaximumDelay(int delay_ms);
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//
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// Get least required delay computed based on channel conditions. Note that
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// this is before applying any user-defined limits (specified by calling
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// (SetMinimumDelay() and/or SetMaximumDelay()).
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//
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int LeastRequiredDelayMs() const;
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//
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// Resets the initial delay to zero.
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//
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void ResetInitialDelay();
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//
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// Get the current sampling frequency in Hz.
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//
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// Return value : Sampling frequency in Hz.
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//
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int current_sample_rate_hz() const;
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//
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// Get the current network statistics from NetEq.
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//
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// Output:
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// - statistics : The current network statistics.
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//
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void GetNetworkStatistics(NetworkStatistics* statistics);
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//
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// Enable post-decoding VAD.
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//
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void EnableVad();
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//
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// Disable post-decoding VAD.
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//
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void DisableVad();
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//
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// Returns whether post-decoding VAD is enabled (true) or disabled (false).
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//
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bool vad_enabled() const { return vad_enabled_; }
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//
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// Flushes the NetEq packet and speech buffers.
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//
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void FlushBuffers();
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//
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// Removes a payload-type from the NetEq codec database.
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//
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// Input:
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// - payload_type : the payload-type to be removed.
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//
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// Return value : 0 if OK.
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// -1 if an error occurred.
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//
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int RemoveCodec(uint8_t payload_type);
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//
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// Remove all registered codecs.
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//
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int RemoveAllCodecs();
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//
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// Set ID.
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//
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void set_id(int id); // TODO(turajs): can be inline.
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//
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// Gets the RTP timestamp of the last sample delivered by GetAudio().
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// Returns true if the RTP timestamp is valid, otherwise false.
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//
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bool GetPlayoutTimestamp(uint32_t* timestamp);
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//
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// Return the index of the codec associated with the last non-CNG/non-DTMF
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// received payload. If no non-CNG/non-DTMF payload is received -1 is
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// returned.
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//
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int last_audio_codec_id() const; // TODO(turajs): can be inline.
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//
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// Get the audio codec associated with the last non-CNG/non-DTMF received
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// payload. If no non-CNG/non-DTMF packet is received -1 is returned,
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// otherwise return 0.
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//
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int LastAudioCodec(CodecInst* codec) const;
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//
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// Get a decoder given its registered payload-type.
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//
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// Input:
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// -payload_type : the payload-type of the codec to be retrieved.
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//
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// Output:
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// -codec : codec associated with the given payload-type.
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//
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// Return value : 0 if succeeded.
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// -1 if failed, e.g. given payload-type is not
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// registered.
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//
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int DecoderByPayloadType(uint8_t payload_type,
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CodecInst* codec) const;
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//
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// Enable NACK and set the maximum size of the NACK list. If NACK is already
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// enabled then the maximum NACK list size is modified accordingly.
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//
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// Input:
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// -max_nack_list_size : maximum NACK list size
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// should be positive (none zero) and less than or
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// equal to |Nack::kNackListSizeLimit|
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// Return value
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// : 0 if succeeded.
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// -1 if failed
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//
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int EnableNack(size_t max_nack_list_size);
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// Disable NACK.
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void DisableNack();
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//
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// Get a list of packets to be retransmitted.
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//
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// Input:
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// -round_trip_time_ms : estimate of the round-trip-time (in milliseconds).
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// Return value : list of packets to be retransmitted.
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//
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std::vector<uint16_t> GetNackList(int64_t round_trip_time_ms) const;
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//
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// Get statistics of calls to GetAudio().
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void GetDecodingCallStatistics(AudioDecodingCallStats* stats) const;
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private:
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const Decoder* RtpHeaderToDecoder(const RTPHeader& rtp_header,
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uint8_t payload_type) const
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EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
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uint32_t NowInTimestamp(int decoder_sampling_rate) const;
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rtc::scoped_ptr<CriticalSectionWrapper> crit_sect_;
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int id_; // TODO(henrik.lundin) Make const.
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const Decoder* last_audio_decoder_ GUARDED_BY(crit_sect_);
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AudioFrame::VADActivity previous_audio_activity_ GUARDED_BY(crit_sect_);
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int current_sample_rate_hz_ GUARDED_BY(crit_sect_);
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ACMResampler resampler_ GUARDED_BY(crit_sect_);
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// Used in GetAudio, declared as member to avoid allocating every 10ms.
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// TODO(henrik.lundin) Stack-allocate in GetAudio instead?
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rtc::scoped_ptr<int16_t[]> audio_buffer_ GUARDED_BY(crit_sect_);
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rtc::scoped_ptr<int16_t[]> last_audio_buffer_ GUARDED_BY(crit_sect_);
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CallStatistics call_stats_ GUARDED_BY(crit_sect_);
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NetEq* neteq_;
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// Decoders map is keyed by payload type
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std::map<uint8_t, Decoder> decoders_ GUARDED_BY(crit_sect_);
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bool vad_enabled_;
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Clock* clock_; // TODO(henrik.lundin) Make const if possible.
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bool resampled_last_output_frame_ GUARDED_BY(crit_sect_);
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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_MAIN_ACM2_ACM_RECEIVER_H_
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