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/*
* Copyright (c) 2014 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_AUDIO_CODING_ACM2_ACM_SEND_TEST_H_
#define MODULES_AUDIO_CODING_ACM2_ACM_SEND_TEST_H_
#include <memory>
#include <vector>
#include "api/audio/audio_frame.h"
#include "modules/audio_coding/include/audio_coding_module.h"
#include "modules/audio_coding/neteq/tools/packet_source.h"
#include "rtc_base/constructor_magic.h"
#include "system_wrappers/include/clock.h"
namespace webrtc {
class AudioEncoder;
namespace test {
class InputAudioFile;
class Packet;
class AcmSendTestOldApi : public AudioPacketizationCallback,
public PacketSource {
public:
AcmSendTestOldApi(InputAudioFile* audio_source,
int source_rate_hz,
int test_duration_ms);
~AcmSendTestOldApi() override;
// Registers the send codec. Returns true on success, false otherwise.
bool RegisterCodec(const char* payload_name,
int sampling_freq_hz,
int channels,
int payload_type,
int frame_size_samples);
// Registers an external send codec.
void RegisterExternalCodec(
std::unique_ptr<AudioEncoder> external_speech_encoder);
// Inherited from PacketSource.
std::unique_ptr<Packet> NextPacket() override;
// Inherited from AudioPacketizationCallback.
int32_t SendData(AudioFrameType frame_type,
uint8_t payload_type,
uint32_t timestamp,
const uint8_t* payload_data,
size_t payload_len_bytes,
int64_t absolute_capture_timestamp_ms) override;
AudioCodingModule* acm() { return acm_.get(); }
private:
static const int kBlockSizeMs = 10;
// Creates a Packet object from the last packet produced by ACM (and received
// through the SendData method as a callback).
std::unique_ptr<Packet> CreatePacket();
SimulatedClock clock_;
std::unique_ptr<AudioCodingModule> acm_;
InputAudioFile* audio_source_;
int source_rate_hz_;
Update a ton of audio code to use size_t more correctly and in general reduce use of int16_t/uint16_t. This is the upshot of a recommendation by henrik.lundin and kwiberg on an original small change ( https://webrtc-codereview.appspot.com/42569004/#ps1 ) to stop using int16_t just because values could fit in it, and is similar in nature to a previous "mass change to use size_t more" ( https://webrtc-codereview.appspot.com/23129004/ ) which also needed to be split up for review but to land all at once, since, like adding "const", such changes tend to cause a lot of transitive effects. This was be reviewed and approved in pieces: https://codereview.webrtc.org/1224093003 https://codereview.webrtc.org/1224123002 https://codereview.webrtc.org/1224163002 https://codereview.webrtc.org/1225133003 https://codereview.webrtc.org/1225173002 https://codereview.webrtc.org/1227163003 https://codereview.webrtc.org/1227203003 https://codereview.webrtc.org/1227213002 https://codereview.webrtc.org/1227893002 https://codereview.webrtc.org/1228793004 https://codereview.webrtc.org/1228803003 https://codereview.webrtc.org/1228823002 https://codereview.webrtc.org/1228823003 https://codereview.webrtc.org/1228843002 https://codereview.webrtc.org/1230693002 https://codereview.webrtc.org/1231713002 The change is being landed as TBR to all the folks who reviewed the above. BUG=chromium:81439 TEST=none R=andrew@webrtc.org, pbos@webrtc.org TBR=aluebs, andrew, asapersson, henrika, hlundin, jan.skoglund, kwiberg, minyue, pbos, pthatcher Review URL: https://codereview.webrtc.org/1230503003 . Cr-Commit-Position: refs/heads/master@{#9768}
2015-08-24 14:52:23 -07:00
const size_t input_block_size_samples_;
AudioFrame input_frame_;
bool codec_registered_;
int test_duration_ms_;
// The following member variables are set whenever SendData() is called.
AudioFrameType frame_type_;
int payload_type_;
uint32_t timestamp_;
uint16_t sequence_number_;
std::vector<uint8_t> last_payload_vec_;
bool data_to_send_;
RTC_DISALLOW_COPY_AND_ASSIGN(AcmSendTestOldApi);
};
} // namespace test
} // namespace webrtc
#endif // MODULES_AUDIO_CODING_ACM2_ACM_SEND_TEST_H_