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audio_frame2: added const semantics

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This commit is contained in:
Martin Pulec
2014-08-21 13:45:57 +02:00
parent db9c8cb04f
commit ef733c6f9e
12 changed files with 41 additions and 40 deletions
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4
src/audio/audio.cpp
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@@ -841,7 +841,7 @@ static void *audio_sender_thread(void *arg)
process_statistics(s, &buffer_new);
free(resample_state.resampled.data);
audio_frame2 *uncompressed = &buffer_new;
audio_frame2 *compressed = NULL;
const audio_frame2 *compressed = NULL;
while((compressed = audio_codec_compress(s->audio_coder, uncompressed))) {
audio_tx_send(s->tx_session, s->audio_network_device, compressed);
uncompressed = NULL;
@@ -853,7 +853,7 @@ static void *audio_sender_thread(void *arg)
buffer_new = audio_frame2(&resample_state.resampled);
free(resample_state.resampled.data);
audio_frame2 *uncompressed = &buffer_new;
audio_frame2 *compressed = NULL;
const audio_frame2 *compressed = NULL;
while((compressed = audio_codec_compress(s->audio_coder, uncompressed))) {
//TODO to be dynamic as a function of the selected codec, now only accepting mulaw without checking errors
audio_tx_send_standard(s->tx_session, s->audio_network_device, compressed);

21
src/audio/audio.h
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@@ -92,7 +92,7 @@ typedef struct
{
int bps; /* bytes per sample */
int sample_rate;
char *data; /* data should be at least 4B aligned */
const char *data; /* data should be at least 4B aligned */
int data_len; /* size of useful data in buffer */
audio_codec_t codec;
double duration;
@@ -109,6 +109,7 @@ class audio_frame2
{
public:
audio_frame2();
audio_frame2(audio_frame2&& other) = default;
audio_frame2(struct audio_frame *);
audio_frame2& operator=(audio_frame2&& other) = default;
void init(int nr_channels, audio_codec_t codec, int bps, int sample_rate);
@@ -117,15 +118,15 @@ public:
void replace(int channel, size_t offset, const char *data, size_t length);
void resize(int channel, size_t len);
void reset();
int get_bps();
audio_codec_t get_codec();
const char *get_data(int channel);
size_t get_data_len(int channel);
double get_duration();
int get_channel_count();
int get_sample_count();
int get_sample_rate();
bool has_same_prop_as(audio_frame2 const &frame);
int get_bps() const;
audio_codec_t get_codec() const;
const char *get_data(int channel) const;
size_t get_data_len(int channel) const;
double get_duration() const;
int get_channel_count() const;
int get_sample_count() const;
int get_sample_rate() const;
bool has_same_prop_as(audio_frame2 const &frame) const;
void set_duration(double duration);
private:
int bps; /* bytes per sample */

2
src/audio/codec.cpp
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@@ -223,7 +223,7 @@ struct audio_codec_state *audio_codec_reconfigure(struct audio_codec_state *old,
* @retval pointer pointing to data
* @retval NULL indicating that there are no data left
*/
audio_frame2 *audio_codec_compress(struct audio_codec_state *s, audio_frame2 *frame)
const audio_frame2 *audio_codec_compress(struct audio_codec_state *s, const audio_frame2 *frame)
{
if(frame && s->state_count < frame->get_channel_count()) {
s->state = (void **) realloc(s->state, sizeof(void **) * frame->get_channel_count());

2
src/audio/codec.h
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@@ -85,7 +85,7 @@ struct audio_codec_state *audio_codec_init(audio_codec_t audio_codec, audio_code
struct audio_codec_state *audio_codec_init_cfg(const char *audio_codec_cfg, audio_codec_direction_t);
struct audio_codec_state *audio_codec_reconfigure(struct audio_codec_state *old,
audio_codec_t audio_codec, audio_codec_direction_t);
audio_frame2 *audio_codec_compress(struct audio_codec_state *, audio_frame2 *);
const audio_frame2 *audio_codec_compress(struct audio_codec_state *, const audio_frame2 *);
audio_frame2 *audio_codec_decompress(struct audio_codec_state *, audio_frame2 *);
const int *audio_codec_get_supported_bps(struct audio_codec_state *);
void audio_codec_done(struct audio_codec_state *);

12
src/audio/codec/libavcodec.cpp
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@@ -341,11 +341,11 @@ static audio_channel *libavcodec_compress(void *state, audio_channel * channel)
}
if(s->change_bps_to) {
change_bps(s->tmp.data, s->saved_desc.bps, channel->data,
change_bps((char *) s->tmp.data, s->saved_desc.bps, channel->data,
s->change_bps_to, channel->data_len);
s->tmp.data_len += channel->data_len / s->saved_desc.bps * s->change_bps_to;
} else {
memcpy(s->tmp.data + s->tmp.data_len, channel->data, channel->data_len);
memcpy((char *) s->tmp.data + s->tmp.data_len, channel->data, channel->data_len);
s->tmp.data_len += channel->data_len;
}
}
@@ -382,7 +382,7 @@ static audio_channel *libavcodec_compress(void *state, audio_channel * channel)
}
s->tmp.data_len -= offset;
memmove(s->tmp.data, s->tmp.data + offset, s->tmp.data_len);
memmove((char *) s->tmp.data, s->tmp.data + offset, s->tmp.data_len);
///fprintf(stderr, "%d %d\n", i++% 2, s->output_channel.data_len);
if(s->output_channel.data_len) {
@@ -425,7 +425,7 @@ static audio_channel *libavcodec_decompress(void *state, audio_channel * channel
int data_size = av_samples_get_buffer_size(NULL, channels,
s->av_frame->nb_samples,
s->codec_ctx->sample_fmt, 1);
memcpy(s->output_channel.data + offset, s->av_frame->data[0],
memcpy((char *) s->output_channel.data + offset, s->av_frame->data[0],
data_size);
offset += len;
s->output_channel.data_len += data_size;
@@ -461,8 +461,8 @@ static void libavcodec_done(void *state)
pthread_mutex_unlock(s->libav_global_lock);
rm_release_shared_lock(LAVCD_LOCK_NAME);
free(s->output_channel.data);
free(s->tmp.data);
free((void *) s->output_channel.data);
free((void *) s->tmp.data);
av_free_packet(&s->pkt);
av_freep(&s->samples);
#if LIBAVCODEC_VERSION_MAJOR >= 54

2
src/audio/resampler.cpp
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@@ -34,7 +34,7 @@ void resampler_done(struct resampler *s)
free(s);
}
audio_frame2 *resampler_resample(struct resampler *s, audio_frame2 *frame)
const audio_frame2 *resampler_resample(struct resampler *s, const audio_frame2 *frame)
{
if(s->resample_to == frame->get_sample_rate()) {
return frame;

2
src/audio/resampler.h
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@@ -14,7 +14,7 @@ struct resampler;
struct resampler *resampler_init(int dst_sample_rate);
void resampler_done(struct resampler *);
audio_frame2 *resampler_resample(struct resampler *, audio_frame2 *);
const audio_frame2 *resampler_resample(struct resampler *, const audio_frame2 *);
#ifdef __cplusplus
}

23
src/audio/utils.cpp
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@@ -175,27 +175,27 @@ void audio_frame2::reset()
duration = 0.0;
}
int audio_frame2::get_bps()
int audio_frame2::get_bps() const
{
return bps;
}
audio_codec_t audio_frame2::get_codec()
audio_codec_t audio_frame2::get_codec() const
{
return codec;
}
const char *audio_frame2::get_data(int channel)
const char *audio_frame2::get_data(int channel) const
{
return channels[channel].first.get();
}
size_t audio_frame2::get_data_len(int channel)
size_t audio_frame2::get_data_len(int channel) const
{
return channels[channel].second;
}
double audio_frame2::get_duration()
double audio_frame2::get_duration() const
{
if (codec == AC_PCM) {
int samples = get_sample_count();
@@ -205,12 +205,12 @@ double audio_frame2::get_duration()
}
}
int audio_frame2::get_channel_count()
int audio_frame2::get_channel_count() const
{
return channels.size();
}
int audio_frame2::get_sample_count()
int audio_frame2::get_sample_count() const
{
// for PCM, we can deduce samples count from length of the data
if (codec == AC_PCM) {
@@ -220,12 +220,13 @@ int audio_frame2::get_sample_count()
}
}
int audio_frame2::get_sample_rate()
int audio_frame2::get_sample_rate() const
{
return sample_rate;
}
bool audio_frame2::has_same_prop_as(audio_frame2 const &frame) {
bool audio_frame2::has_same_prop_as(audio_frame2 const &frame) const
{
return bps == frame.bps &&
sample_rate == frame.sample_rate &&
codec == frame.codec &&
@@ -514,9 +515,9 @@ void signed2unsigned(char *out, char *in, int in_len)
}
}
void audio_channel_demux(audio_frame2 *frame, int index, audio_channel *channel)
void audio_channel_demux(const audio_frame2 *frame, int index, audio_channel *channel)
{
channel->data = (char *) frame->get_data(index);
channel->data = frame->get_data(index);
channel->data_len = frame->get_data_len(index);
channel->codec = frame->get_codec();
channel->bps = frame->get_bps();

2
src/audio/utils.h
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@@ -63,7 +63,7 @@ int audio_frame2_get_sample_count(audio_frame2 *frame);
void audio_frame2_reset(audio_frame2 *frame);
double calculate_rms(audio_frame2 *frame, int channel, double *peak);
struct audio_desc audio_desc_from_audio_frame2(audio_frame2 *);
void audio_channel_demux(audio_frame2 *, int, audio_channel*);
void audio_channel_demux(const audio_frame2 *, int, audio_channel*);
void audio_channel_mux(audio_frame2 *, int, audio_channel*);
#endif

2
src/rtp/audio_decoders.cpp
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@@ -513,7 +513,7 @@ int decode_audio_frame(struct coded_data *cdata, void *data)
return FALSE;
}
audio_frame2 *resampled = resampler_resample(decoder->resampler, decompressed);
const audio_frame2 *resampled = resampler_resample(decoder->resampler, decompressed);
size_t new_data_len = s->buffer.data_len + resampled->get_data_len(0) * output_channels;
if(s->buffer.max_size < new_data_len) {

5
src/transmit.cpp
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@@ -67,7 +67,6 @@
#include "perf.h"
#include "audio/audio.h"
#include "audio/codec.h"
#include "audio/utils.h"
#include "crypto/openssl_encrypt.h"
#include "module.h"
#include "rtp/fec.h"
@@ -633,7 +632,7 @@ tx_send_base(struct tx *tx, struct video_frame *frame, struct rtp *rtp_session,
* This multiplication scheme relies upon the fact, that our RTP/pbuf implementation is
* not sensitive to packet duplication. Otherwise, we can get into serious problems.
*/
void audio_tx_send(struct tx* tx, struct rtp *rtp_session, audio_frame2 * buffer)
void audio_tx_send(struct tx* tx, struct rtp *rtp_session, const audio_frame2 * buffer)
{
int pt; /* PT set for audio in our packet format */
unsigned int pos = 0u,
@@ -794,7 +793,7 @@ void audio_tx_send(struct tx* tx, struct rtp *rtp_session, audio_frame2 * buffer
* as the mulaw and A-law standards (dynamic or std PT).
*/
void audio_tx_send_standard(struct tx* tx, struct rtp *rtp_session,
audio_frame2 * buffer) {
const audio_frame2 * buffer) {
//TODO to be more abstract in order to accept A-law too and other supported standards with such implementation
assert(buffer->get_codec() == AC_MULAW || buffer->get_codec() == AC_ALAW);

4
src/transmit.h
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@@ -86,8 +86,8 @@ void tx_send_h264(struct tx *tx_session, struct video_frame *frame, struct rtp *
#endif
#ifdef __cplusplus
void audio_tx_send(struct tx *tx_session, struct rtp *rtp_session, audio_frame2 *buffer);
void audio_tx_send_standard(struct tx* tx, struct rtp *rtp_session, audio_frame2 * buffer);
void audio_tx_send(struct tx *tx_session, struct rtp *rtp_session, const audio_frame2 *buffer);
void audio_tx_send_standard(struct tx* tx, struct rtp *rtp_session, const audio_frame2 * buffer);
#endif
#endif // TRANSMIT_H_