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Nack for audio.

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R=stefan@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1507004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4188 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
turaj@webrtc.org
2013-06-06 19:00:09 +00:00
parent d9c4658756
commit a305e9612a
19 changed files with 1522 additions and 55 deletions
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59
webrtc/modules/audio_coding/main/interface/audio_coding_module.h
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@@ -8,8 +8,10 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_INTERFACE_AUDIO_CODING_MODULE_H
#define WEBRTC_MODULES_AUDIO_CODING_MAIN_INTERFACE_AUDIO_CODING_MODULE_H
#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_INTERFACE_AUDIO_CODING_MODULE_H_
#define WEBRTC_MODULES_AUDIO_CODING_MAIN_INTERFACE_AUDIO_CODING_MODULE_H_
#include <vector>
#include "webrtc/common_types.h"
#include "webrtc/modules/audio_coding/main/interface/audio_coding_module_typedefs.h"
@@ -23,8 +25,9 @@ struct CodecInst;
struct WebRtcRTPHeader;
class AudioFrame;
class RTPFragmentationHeader;
class Clock;
#define WEBRTC_10MS_PCM_AUDIO 960 // 16 bits super wideband 48 kHz
#define WEBRTC_10MS_PCM_AUDIO 960 // 16 bits super wideband 48 kHz
// Callback class used for sending data ready to be packetized
class AudioPacketizationCallback {
@@ -63,11 +66,11 @@ class ACMVQMonCallback {
virtual ~ACMVQMonCallback() {}
virtual int32_t NetEqStatistics(
const int32_t id, // current ACM id
const uint16_t MIUsValid, // valid voice duration in ms
const uint16_t MIUsReplaced, // concealed voice duration in ms
const uint8_t eventFlags, // concealed voice flags
const uint16_t delayMS) = 0; // average delay in ms
const int32_t id, // current ACM id
const uint16_t MIUsValid, // valid voice duration in ms
const uint16_t MIUsReplaced, // concealed voice duration in ms
const uint8_t eventFlags, // concealed voice flags
const uint16_t delayMS) = 0; // average delay in ms
};
class AudioCodingModule: public Module {
@@ -77,9 +80,14 @@ class AudioCodingModule: public Module {
public:
///////////////////////////////////////////////////////////////////////////
// Creation and destruction of a ACM
// Creation and destruction of a ACM.
//
// The second method is used for testing where a simulated clock can be
// injected into ACM. ACM will take the owner ship of the object clock and
// delete it when destroyed.
//
static AudioCodingModule* Create(const int32_t id);
static AudioCodingModule* Create(const int32_t id, Clock* clock);
static void Destroy(AudioCodingModule* module);
@@ -814,8 +822,8 @@ class AudioCodingModule: public Module {
//
// Input:
// -desired_freq_hz : the desired sampling frequency, in Hertz, of the
// output audio. If set to -1, the function returns the
// audio at the current sampling frequency.
// output audio. If set to -1, the function returns
// the audio at the current sampling frequency.
//
// Output:
// -audio_frame : output audio frame which contains raw audio data
@@ -966,8 +974,35 @@ class AudioCodingModule: public Module {
// 0 if delay is set successfully.
//
virtual int SetInitialPlayoutDelay(int delay_ms) = 0;
//
// Enable NACK and set the maximum size of the NACK list. If NACK is already
// enable then the maximum NACK list size is modified accordingly.
//
// If the sequence number of last received packet is N, the sequence numbers
// of NACK list are in the range of [N - |max_nack_list_size|, N).
//
// |max_nack_list_size| should be positive (none zero) and less than or
// equal to |Nack::kNackListSizeLimit|. Otherwise, No change is applied and -1
// is returned. 0 is returned at success.
//
virtual int EnableNack(size_t max_nack_list_size) = 0;
// Disable NACK.
virtual void DisableNack() = 0;
//
// Get a list of packets to be retransmitted. |round_trip_time_ms| is an
// estimate of the round-trip-time (in milliseconds). Missing packets which
// will be playout in a shorter time than the round-trip-time (with respect
// to the time this API is called) will not be included in the list.
//
// Negative |round_trip_time_ms| results is an error message and empty list
// is returned.
//
virtual std::vector<uint16_t> GetNackList(int round_trip_time_ms) const = 0;
};
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_INTERFACE_AUDIO_CODING_MODULE_H
#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_INTERFACE_AUDIO_CODING_MODULE_H_