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Reland r8248 "Introduce ACMGenericCodecWrapper"

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This effectively reverts r8249.

This new class inherits from ACMGenericCodec. The purpose is to wrap
AudioEncoder objects into an ACMGenericCodec interface. This is a
temporary construction that will be used during the ACM redesign work.

BUG=4228
COAUTHOR=kwiberg@webrtc.org
TBR=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/38919004

Cr-Commit-Position: refs/heads/master@{#8255}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8255 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
henrik.lundin@webrtc.org
2015-02-05 18:29:39 +00:00
parent ec4521cdb4
commit f45c8ca88b
10 changed files with 941 additions and 39 deletions
Download Patch File Download Diff File Expand all files Collapse all files

15
webrtc/modules/audio_coding/codecs/audio_encoder.cc
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@ -9,6 +9,7 @@
*/
#include "webrtc/modules/audio_coding/codecs/audio_encoder.h"
#include "webrtc/base/checks.h"
namespace webrtc {
@ -18,6 +19,20 @@ AudioEncoder::EncodedInfo::EncodedInfo() : EncodedInfoLeaf() {
AudioEncoder::EncodedInfo::~EncodedInfo() {
}
bool AudioEncoder::Encode(uint32_t rtp_timestamp,
const int16_t* audio,
size_t num_samples_per_channel,
size_t max_encoded_bytes,
uint8_t* encoded,
EncodedInfo* info) {
CHECK_EQ(num_samples_per_channel,
static_cast<size_t>(sample_rate_hz() / 100));
bool ret =
EncodeInternal(rtp_timestamp, audio, max_encoded_bytes, encoded, info);
CHECK_LE(info->encoded_bytes, max_encoded_bytes);
return ret;
}
int AudioEncoder::rtp_timestamp_rate_hz() const {
return sample_rate_hz();
}

10
webrtc/modules/audio_coding/codecs/audio_encoder.h
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@ -14,7 +14,6 @@
#include <algorithm>
#include <vector>
#include "webrtc/base/checks.h"
#include "webrtc/typedefs.h"
namespace webrtc {
@ -63,14 +62,7 @@ class AudioEncoder {
size_t num_samples_per_channel,
size_t max_encoded_bytes,
uint8_t* encoded,
EncodedInfo* info) {
CHECK_EQ(num_samples_per_channel,
static_cast<size_t>(sample_rate_hz() / 100));
bool ret =
EncodeInternal(rtp_timestamp, audio, max_encoded_bytes, encoded, info);
CHECK_LE(info->encoded_bytes, max_encoded_bytes);
return ret;
}
EncodedInfo* info);
// Return the input sample rate in Hz and the number of input channels.
// These are constants set at instantiation time.

1
webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.cc
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@ -12,6 +12,7 @@
#include <limits>
#include "webrtc/base/checks.h"
#include "webrtc/modules/audio_coding/codecs/g711/include/g711_interface.h"
namespace webrtc {

1
webrtc/modules/audio_coding/codecs/isac/fix/interface/audio_encoder_isacfix.h
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@ -11,6 +11,7 @@
#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_INTERFACE_AUDIO_ENCODER_ISACFIX_H_
#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_INTERFACE_AUDIO_ENCODER_ISACFIX_H_
#include "webrtc/base/checks.h"
#include "webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t.h"
#include "webrtc/modules/audio_coding/codecs/isac/fix/interface/isacfix.h"

1
webrtc/modules/audio_coding/codecs/isac/main/interface/audio_encoder_isac.h
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@ -11,6 +11,7 @@
#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_INTERFACE_AUDIO_ENCODER_ISAC_H_
#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_INTERFACE_AUDIO_ENCODER_ISAC_H_
#include "webrtc/base/checks.h"
#include "webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t.h"
#include "webrtc/modules/audio_coding/codecs/isac/main/interface/isac.h"

2
webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.cc
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@ -12,6 +12,8 @@
#include <string.h>
#include "webrtc/base/checks.h"
namespace webrtc {
AudioEncoderCopyRed::AudioEncoderCopyRed(const Config& config)

1
webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red_unittest.cc
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@ -9,6 +9,7 @@
*/
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/base/checks.h"
#include "webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.h"
#include "webrtc/modules/audio_coding/codecs/mock/mock_audio_encoder.h"
#include "webrtc/system_wrappers/interface/scoped_ptr.h"