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(Re-land) AudioEncoderDecoderIsac: Merge the two config structs

Browse source 
This reverts commit 599beb86, which in turn reverted 7c324cac. What
makes it work this time is that we don't remove the option of setting
bit_rate to 0 in order to ask for the default value.

COAUTHOR=henrik.lundin@webrtc.org
BUG=4228, chromium:478161
R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/48199004

Cr-Commit-Position: refs/heads/master@{#9068}
This commit is contained in:
Karl Wiberg 2015-04-23 14:07:06 +02:00
parent 92f9eacd13
commit d3e8eda839
6 changed files with 105 additions and 127 deletions
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33
webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t.h
View file

@ -25,8 +25,7 @@ class CriticalSectionWrapper;
template <typename T>
class AudioEncoderDecoderIsacT : public AudioEncoder, public AudioDecoder {
public:
// For constructing an encoder in instantaneous mode. Allowed combinations
// are
// Allowed combinations of sample rate, frame size, and bit rate are
// - 16000 Hz, 30 ms, 10000-32000 bps
// - 16000 Hz, 60 ms, 10000-32000 bps
// - 32000 Hz, 30 ms, 10000-56000 bps (if T has super-wideband support)
@ -34,34 +33,24 @@ class AudioEncoderDecoderIsacT : public AudioEncoder, public AudioDecoder {
struct Config {
Config();
bool IsOk() const;
int payload_type;
int sample_rate_hz;
int frame_size_ms;
int bit_rate; // Limit on the short-term average bit rate, in bits/second.
int max_bit_rate;
int bit_rate; // Limit on the short-term average bit rate, in bits/s.
int max_payload_size_bytes;
};
int max_bit_rate;
// For constructing an encoder in channel-adaptive mode. Allowed combinations
// are
// - 16000 Hz, 30 ms, 10000-32000 bps
// - 16000 Hz, 60 ms, 10000-32000 bps
// - 32000 Hz, 30 ms, 10000-56000 bps (if T has super-wideband support)
// - 48000 Hz, 30 ms, 10000-56000 bps (if T has super-wideband support)
struct ConfigAdaptive {
ConfigAdaptive();
bool IsOk() const;
int payload_type;
int sample_rate_hz;
int initial_frame_size_ms;
int initial_bit_rate;
int max_bit_rate;
bool enforce_frame_size; // Prevent adaptive changes to the frame size?
int max_payload_size_bytes;
// If true, the encoder will dynamically adjust frame size and bit rate;
// the configured values are then merely the starting point.
bool adaptive_mode;
// In adaptive mode, prevent adaptive changes to the frame size. (Not used
// in nonadaptive mode.)
bool enforce_frame_size;
};
explicit AudioEncoderDecoderIsacT(const Config& config);
explicit AudioEncoderDecoderIsacT(const ConfigAdaptive& config);
~AudioEncoderDecoderIsacT() override;
// AudioEncoder public methods.

84
webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h
View file

@ -30,8 +30,10 @@ AudioEncoderDecoderIsacT<T>::Config::Config()
sample_rate_hz(16000),
frame_size_ms(30),
bit_rate(kDefaultBitRate),
max_payload_size_bytes(-1),
max_bit_rate(-1),
max_payload_size_bytes(-1) {
adaptive_mode(false),
enforce_frame_size(false) {
}
template <typename T>
@ -47,7 +49,7 @@ bool AudioEncoderDecoderIsacT<T>::Config::IsOk() const {
if (max_payload_size_bytes > 400)
return false;
return (frame_size_ms == 30 || frame_size_ms == 60) &&
((bit_rate >= 10000 && bit_rate <= 32000) || bit_rate == 0);
(bit_rate == 0 || (bit_rate >= 10000 && bit_rate <= 32000));
case 32000:
case 48000:
if (max_bit_rate > 160000)
@ -56,46 +58,7 @@ bool AudioEncoderDecoderIsacT<T>::Config::IsOk() const {
return false;
return T::has_swb &&
(frame_size_ms == 30 &&
((bit_rate >= 10000 && bit_rate <= 56000) || bit_rate == 0));
default:
return false;
}
}
template <typename T>
AudioEncoderDecoderIsacT<T>::ConfigAdaptive::ConfigAdaptive()
: payload_type(kIsacPayloadType),
sample_rate_hz(16000),
initial_frame_size_ms(30),
initial_bit_rate(kDefaultBitRate),
max_bit_rate(-1),
enforce_frame_size(false),
max_payload_size_bytes(-1) {
}
template <typename T>
bool AudioEncoderDecoderIsacT<T>::ConfigAdaptive::IsOk() const {
if (max_bit_rate < 32000 && max_bit_rate != -1)
return false;
if (max_payload_size_bytes < 120 && max_payload_size_bytes != -1)
return false;
switch (sample_rate_hz) {
case 16000:
if (max_bit_rate > 53400)
return false;
if (max_payload_size_bytes > 400)
return false;
return (initial_frame_size_ms == 30 || initial_frame_size_ms == 60) &&
initial_bit_rate >= 10000 && initial_bit_rate <= 32000;
case 32000:
case 48000:
if (max_bit_rate > 160000)
return false;
if (max_payload_size_bytes > 600)
return false;
return T::has_swb &&
(initial_frame_size_ms == 30 && initial_bit_rate >= 10000 &&
initial_bit_rate <= 56000);
(bit_rate == 0 || (bit_rate >= 10000 && bit_rate <= 56000)));
default:
return false;
}
@ -110,11 +73,16 @@ AudioEncoderDecoderIsacT<T>::AudioEncoderDecoderIsacT(const Config& config)
packet_in_progress_(false) {
CHECK(config.IsOk());
CHECK_EQ(0, T::Create(&isac_state_));
CHECK_EQ(0, T::EncoderInit(isac_state_, 1));
CHECK_EQ(0, T::EncoderInit(isac_state_, config.adaptive_mode ? 0 : 1));
CHECK_EQ(0, T::SetEncSampRate(isac_state_, config.sample_rate_hz));
CHECK_EQ(0, T::Control(isac_state_, config.bit_rate == 0 ? kDefaultBitRate
: config.bit_rate,
config.frame_size_ms));
const int bit_rate = config.bit_rate == 0 ? kDefaultBitRate : config.bit_rate;
if (config.adaptive_mode) {
CHECK_EQ(0, T::ControlBwe(isac_state_, bit_rate,
config.frame_size_ms, config.enforce_frame_size));
} else {
CHECK_EQ(0, T::Control(isac_state_, bit_rate, config.frame_size_ms));
}
// When config.sample_rate_hz is set to 48000 Hz (iSAC-fb), the decoder is
// still set to 32000 Hz, since there is no full-band mode in the decoder.
CHECK_EQ(0, T::SetDecSampRate(isac_state_,
@ -124,29 +92,7 @@ AudioEncoderDecoderIsacT<T>::AudioEncoderDecoderIsacT(const Config& config)
T::SetMaxPayloadSize(isac_state_, config.max_payload_size_bytes));
if (config.max_bit_rate != -1)
CHECK_EQ(0, T::SetMaxRate(isac_state_, config.max_bit_rate));
}
template <typename T>
AudioEncoderDecoderIsacT<T>::AudioEncoderDecoderIsacT(
const ConfigAdaptive& config)
: payload_type_(config.payload_type),
state_lock_(CriticalSectionWrapper::CreateCriticalSection()),
decoder_sample_rate_hz_(0),
lock_(CriticalSectionWrapper::CreateCriticalSection()),
packet_in_progress_(false) {
CHECK(config.IsOk());
CHECK_EQ(0, T::Create(&isac_state_));
CHECK_EQ(0, T::EncoderInit(isac_state_, 0));
CHECK_EQ(0, T::SetEncSampRate(isac_state_, config.sample_rate_hz));
CHECK_EQ(0, T::ControlBwe(isac_state_, config.initial_bit_rate,
config.initial_frame_size_ms,
config.enforce_frame_size));
CHECK_EQ(0, T::SetDecSampRate(isac_state_, config.sample_rate_hz));
if (config.max_payload_size_bytes != -1)
CHECK_EQ(0,
T::SetMaxPayloadSize(isac_state_, config.max_payload_size_bytes));
if (config.max_bit_rate != -1)
CHECK_EQ(0, T::SetMaxRate(isac_state_, config.max_bit_rate));
CHECK_EQ(0, T::DecoderInit(isac_state_));
}
template <typename T>

56
webrtc/modules/audio_coding/codecs/isac/main/source/audio_encoder_isac_unittest.cc Normal file
View file

@ -0,0 +1,56 @@
/*
* Copyright (c) 2015 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.
*/
#include <limits>
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/modules/audio_coding/codecs/isac/main/interface/audio_encoder_isac.h"
namespace webrtc {
namespace {
void TestBadConfig(const AudioEncoderDecoderIsac::Config& config) {
EXPECT_FALSE(config.IsOk());
}
void TestGoodConfig(const AudioEncoderDecoderIsac::Config& config) {
EXPECT_TRUE(config.IsOk());
AudioEncoderDecoderIsac ed(config);
}
// Wrap subroutine calls that test things in this, so that the error messages
// will be accompanied by stack traces that make it possible to tell which
// subroutine invocation caused the failure.
#define S(x) do { SCOPED_TRACE(#x); x; } while (0)
} // namespace
TEST(AudioEncoderIsacTest, TestConfigBitrate) {
AudioEncoderDecoderIsac::Config config;
// The default value is some real, positive value.
EXPECT_GT(config.bit_rate, 1);
S(TestGoodConfig(config));
// 0 is another way to ask for the default value.
config.bit_rate = 0;
S(TestGoodConfig(config));
// Try some unreasonable values and watch them fail.
config.bit_rate = -1;
S(TestBadConfig(config));
config.bit_rate = 1;
S(TestBadConfig(config));
config.bit_rate = std::numeric_limits<int>::max();
S(TestBadConfig(config));
}
} // namespace webrtc

55
webrtc/modules/audio_coding/main/acm2/acm_generic_codec.cc
View file

@ -293,51 +293,34 @@ void ACMGenericCodec::ResetAudioEncoder() {
#endif
#ifdef WEBRTC_CODEC_ISACFX
} else if (!STR_CASE_CMP(codec_inst.plname, "ISAC")) {
DCHECK_EQ(codec_inst.plfreq, 16000);
is_isac_ = true;
AudioEncoderDecoderIsacFix* enc_dec;
if (codec_inst.rate == -1) {
// Adaptive mode.
AudioEncoderDecoderIsacFix::ConfigAdaptive config;
config.payload_type = codec_inst.pltype;
enc_dec = new AudioEncoderDecoderIsacFix(config);
} else {
// Channel independent mode.
AudioEncoderDecoderIsacFix::Config config;
AudioEncoderDecoderIsacFix::Config config;
config.payload_type = codec_inst.pltype;
config.sample_rate_hz = codec_inst.plfreq;
config.frame_size_ms = rtc::CheckedDivExact(codec_inst.pacsize, 16);
if (codec_inst.rate != -1)
config.bit_rate = codec_inst.rate;
config.frame_size_ms = codec_inst.pacsize / 16;
config.payload_type = codec_inst.pltype;
enc_dec = new AudioEncoderDecoderIsacFix(config);
}
config.max_payload_size_bytes = max_payload_size_bytes_;
config.max_bit_rate = max_rate_bps_;
config.adaptive_mode = (codec_inst.rate == -1);
auto* enc_dec = new AudioEncoderDecoderIsacFix(config);
decoder_proxy_.SetDecoder(enc_dec);
audio_encoder_.reset(enc_dec);
#endif
#ifdef WEBRTC_CODEC_ISAC
} else if (!STR_CASE_CMP(codec_inst.plname, "ISAC")) {
is_isac_ = true;
AudioEncoderDecoderIsac* enc_dec;
if (codec_inst.rate == -1) {
// Adaptive mode.
AudioEncoderDecoderIsac::ConfigAdaptive config;
config.sample_rate_hz = codec_inst.plfreq;
config.initial_frame_size_ms = rtc::CheckedDivExact(
1000 * codec_inst.pacsize, config.sample_rate_hz);
config.max_payload_size_bytes = max_payload_size_bytes_;
config.max_bit_rate = max_rate_bps_;
config.payload_type = codec_inst.pltype;
enc_dec = new AudioEncoderDecoderIsac(config);
} else {
// Channel independent mode.
AudioEncoderDecoderIsac::Config config;
config.sample_rate_hz = codec_inst.plfreq;
AudioEncoderDecoderIsac::Config config;
config.payload_type = codec_inst.pltype;
config.sample_rate_hz = codec_inst.plfreq;
config.frame_size_ms =
rtc::CheckedDivExact(1000 * codec_inst.pacsize, config.sample_rate_hz);
if (codec_inst.rate != -1)
config.bit_rate = codec_inst.rate;
config.frame_size_ms = rtc::CheckedDivExact(1000 * codec_inst.pacsize,
config.sample_rate_hz);
config.max_payload_size_bytes = max_payload_size_bytes_;
config.max_bit_rate = max_rate_bps_;
config.payload_type = codec_inst.pltype;
enc_dec = new AudioEncoderDecoderIsac(config);
}
config.max_payload_size_bytes = max_payload_size_bytes_;
config.max_bit_rate = max_rate_bps_;
config.adaptive_mode = (codec_inst.rate == -1);
auto* enc_dec = new AudioEncoderDecoderIsac(config);
decoder_proxy_.SetDecoder(enc_dec);
audio_encoder_.reset(enc_dec);
#endif

3
webrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc
View file

@ -361,6 +361,7 @@ class AudioDecoderIsacFloatTest : public AudioDecoderTest {
AudioEncoderDecoderIsac::Config config;
config.payload_type = payload_type_;
config.sample_rate_hz = codec_input_rate_hz_;
config.adaptive_mode = false;
config.frame_size_ms =
1000 * static_cast<int>(frame_size_) / codec_input_rate_hz_;
@ -380,6 +381,7 @@ class AudioDecoderIsacSwbTest : public AudioDecoderTest {
AudioEncoderDecoderIsac::Config config;
config.payload_type = payload_type_;
config.sample_rate_hz = codec_input_rate_hz_;
config.adaptive_mode = false;
config.frame_size_ms =
1000 * static_cast<int>(frame_size_) / codec_input_rate_hz_;
@ -399,6 +401,7 @@ class AudioDecoderIsacFixTest : public AudioDecoderTest {
AudioEncoderDecoderIsacFix::Config config;
config.payload_type = payload_type_;
config.sample_rate_hz = codec_input_rate_hz_;
config.adaptive_mode = false;
config.frame_size_ms =
1000 * static_cast<int>(frame_size_) / codec_input_rate_hz_;

1
webrtc/modules/modules.gyp
View file

@ -108,6 +108,7 @@
'audio_coding/codecs/isac/fix/source/filterbanks_unittest.cc',
'audio_coding/codecs/isac/fix/source/lpc_masking_model_unittest.cc',
'audio_coding/codecs/isac/fix/source/transform_unittest.cc',
'audio_coding/codecs/isac/main/source/audio_encoder_isac_unittest.cc',
'audio_coding/codecs/isac/main/source/isac_unittest.cc',
'audio_coding/codecs/opus/audio_encoder_opus_unittest.cc',
'audio_coding/codecs/opus/opus_unittest.cc',