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Code Issues Releases Wiki Activity

ao_coreaudio: move to new log API

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This commit is contained in:
Stefano Pigozzi 2013-08-01 16:47:36 +02:00
parent 5cd5f0cf70
commit 0bd09da570
4 changed files with 101 additions and 99 deletions
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56
audio/out/ao_coreaudio.c
View File

@ -46,10 +46,10 @@ static void audio_pause(struct ao *ao);
static void audio_resume(struct ao *ao);
static void reset(struct ao *ao);
static void print_buffer(struct mp_ring *buffer)
static void print_buffer(struct ao *ao, struct mp_ring *buffer)
{
void *tctx = talloc_new(NULL);
ca_msg(MSGL_V, "%s\n", mp_ring_repr(buffer, tctx));
MP_VERBOSE(ao, "%s\n", mp_ring_repr(buffer, tctx));
talloc_free(tctx);
}
@ -110,7 +110,7 @@ static OSStatus render_cb_lpcm(void *ctx, AudioUnitRenderActionFlags *aflags,
int requested = buf.mDataByteSize;
if (mp_ring_buffered(p->buffer) < requested) {
ca_msg(MSGL_V, "buffer underrun\n");
MP_VERBOSE(ao, "buffer underrun\n");
audio_pause(ao);
} else {
mp_ring_read(p->buffer, buf.mData, requested);
@ -197,7 +197,7 @@ coreaudio_error:
return CONTROL_ERROR;
}
static void print_list(void)
static void print_list(struct ao *ao)
{
char *help = talloc_strdup(NULL, "Available output devices:\n");
@ -226,7 +226,7 @@ static void print_list(void)
talloc_free(devs);
coreaudio_error:
ca_msg(MSGL_INFO, "%s", help);
MP_INFO(ao, "%s", help);
talloc_free(help);
}
@ -238,7 +238,7 @@ static int init(struct ao *ao)
OSStatus err;
struct priv *p = ao->priv;
if (p->opt_list) print_list();
if (p->opt_list) print_list(ao);
struct priv_d *d = talloc_zero(p, struct priv_d);
@ -271,9 +271,8 @@ static int init(struct ao *ao)
err = CA_GET_STR(selected_device, kAudioObjectPropertyName, &device_name);
CHECK_CA_ERROR("could not get selected audio device name");
ca_msg(MSGL_V,
"selected audio output device: %s (%" PRIu32 ")\n",
device_name, selected_device);
MP_VERBOSE(ao, "selected audio output device: %s (%" PRIu32 ")\n",
device_name, selected_device);
talloc_free(device_name);
@ -283,7 +282,7 @@ static int init(struct ao *ao)
bool supports_digital = false;
/* Probe whether device support S/PDIF stream output if input is AC3 stream. */
if (AF_FORMAT_IS_AC3(ao->format)) {
if (ca_device_supports_digital(selected_device))
if (ca_device_supports_digital(ao, selected_device))
supports_digital = true;
}
@ -298,7 +297,7 @@ static int init(struct ao *ao)
uint32_t *bitmaps;
size_t n_bitmaps;
ca_bitmaps_from_layouts(layouts, n_layouts, &bitmaps, &n_bitmaps);
ca_bitmaps_from_layouts(ao, layouts, n_layouts, &bitmaps, &n_bitmaps);
talloc_free(layouts);
struct mp_chmap_sel chmap_sel = {0};
@ -344,7 +343,7 @@ static int init(struct ao *ao)
asbd.mFramesPerPacket * asbd.mChannelsPerFrame *
(asbd.mBitsPerChannel / 8);
ca_print_asbd("source format:", &asbd);
ca_print_asbd(ao, "source format:", &asbd);
if (supports_digital)
return init_digital(ao, asbd);
@ -373,7 +372,7 @@ static int init_lpcm(struct ao *ao, AudioStreamBasicDescription asbd)
AudioComponent comp = AudioComponentFindNext(NULL, &desc);
if (comp == NULL) {
ca_msg(MSGL_ERR, "unable to find audio component\n");
MP_ERR(ao, "unable to find audio component\n");
goto coreaudio_error;
}
@ -417,7 +416,7 @@ static int init_lpcm(struct ao *ao, AudioStreamBasicDescription asbd)
}
p->buffer = mp_ring_new(p, get_ring_size(ao));
print_buffer(p->buffer);
print_buffer(ao, p->buffer);
AURenderCallbackStruct render_cb = (AURenderCallbackStruct) {
.inputProc = render_cb_lpcm,
@ -454,14 +453,14 @@ static int init_digital(struct ao *ao, AudioStreamBasicDescription asbd)
CHECK_CA_WARN("could not check whether device is alive");
if (!is_alive)
ca_msg(MSGL_WARN, "device is not alive\n");
MP_WARN(ao , "device is not alive\n");
p->is_digital = 1;
err = ca_lock_device(p->device, &d->hog_pid);
CHECK_CA_WARN("failed to set hogmode");
err = ca_disable_mixing(p->device, &d->changed_mixing);
err = ca_disable_mixing(ao, p->device, &d->changed_mixing);
CHECK_CA_WARN("failed to disable mixing");
AudioStreamID *streams;
@ -474,7 +473,7 @@ static int init_digital(struct ao *ao, AudioStreamBasicDescription asbd)
CHECK_CA_ERROR("could not get number of streams");
for (int i = 0; i < n_streams && d->stream_idx < 0; i++) {
bool digital = ca_stream_supports_digital(streams[i]);
bool digital = ca_stream_supports_digital(ao, streams[i]);
if (digital) {
err = CA_GET(streams[i], kAudioStreamPropertyPhysicalFormat,
@ -525,11 +524,11 @@ static int init_digital(struct ao *ao, AudioStreamBasicDescription asbd)
talloc_free(streams);
if (d->stream_idx < 0) {
ca_msg(MSGL_WARN, "can't find any digital output stream format\n");
MP_WARN(ao , "can't find any digital output stream format\n");
goto coreaudio_error;
}
if (!ca_change_format(d->stream, d->stream_asbd))
if (!ca_change_format(ao, d->stream, d->stream_asbd))
goto coreaudio_error;
void *changed = (void *) &(d->stream_asbd_changed);
@ -544,8 +543,7 @@ static int init_digital(struct ao *ao, AudioStreamBasicDescription asbd)
ao->format = AF_FORMAT_AC3_LE;
else if (d->stream_asbd.mFormatFlags & kAudioFormatFlagIsBigEndian)
#endif
ca_msg(MSGL_WARN,
"stream has non-native byte order, digital output may fail\n");
MP_WARN(ao, "stream has non-native byte order, output may fail\n");
ao->samplerate = d->stream_asbd.mSampleRate;
ao->bps = ao->samplerate *
@ -553,7 +551,7 @@ static int init_digital(struct ao *ao, AudioStreamBasicDescription asbd)
d->stream_asbd.mFramesPerPacket);
p->buffer = mp_ring_new(p, get_ring_size(ao));
print_buffer(p->buffer);
print_buffer(ao, p->buffer);
err = AudioDeviceCreateIOProcID(p->device,
(AudioDeviceIOProc)render_cb_digital,
@ -580,11 +578,11 @@ static int play(struct ao *ao, void *output_samples, int num_bytes, int flags)
// Check whether we need to reset the digital output stream.
if (p->is_digital && d->stream_asbd_changed) {
d->stream_asbd_changed = 0;
if (ca_stream_supports_digital(d->stream)) {
if (!ca_change_format(d->stream, d->stream_asbd)) {
ca_msg(MSGL_WARN, "can't restore digital output\n");
if (ca_stream_supports_digital(ao, d->stream)) {
if (!ca_change_format(ao, d->stream, d->stream_asbd)) {
MP_WARN(ao , "can't restore digital output\n");
} else {
ca_msg(MSGL_WARN, "restoring digital output succeeded.\n");
MP_WARN(ao, "restoring digital output succeeded.\n");
reset(ao);
}
}
@ -642,10 +640,10 @@ static void uninit(struct ao *ao, bool immed)
err = AudioDeviceDestroyIOProcID(p->device, d->render_cb);
CHECK_CA_WARN("failed to remove device render callback");
if (!ca_change_format(d->stream, d->original_asbd))
ca_msg(MSGL_WARN, "can't revert to original device format");
if (!ca_change_format(ao, d->stream, d->original_asbd))
MP_WARN(ao, "can't revert to original device format");
err = ca_enable_mixing(p->device, d->changed_mixing);
err = ca_enable_mixing(ao, p->device, d->changed_mixing);
CHECK_CA_WARN("can't re-enable mixing");
err = ca_unlock_device(p->device, &d->hog_pid);

6
audio/out/ao_coreaudio_properties.c
View File

@ -60,13 +60,13 @@ OSStatus ca_get_ary(AudioObjectID id, ca_scope scope, ca_sel selector,
};
err = AudioObjectGetPropertyDataSize(id, &p_addr, 0, NULL, &p_size);
CHECK_CA_ERROR("can't fetch property size");
CHECK_CA_ERROR_SILENT_L(coreaudio_error);
*data = talloc_size(NULL, p_size);
*elements = p_size / element_size;
err = ca_get(id, scope, selector, p_size, *data);
CHECK_CA_ERROR_L(coreaudio_error_free, "can't fetch property data");
CHECK_CA_ERROR_SILENT_L(coreaudio_error_free);
return err;
coreaudio_error_free:
@ -81,7 +81,7 @@ OSStatus ca_get_str(AudioObjectID id, ca_scope scope, ca_sel selector,
CFStringRef string;
OSStatus err =
ca_get(id, scope, selector, sizeof(CFStringRef), (void **)&string);
CHECK_CA_ERROR("Can't fetch string property");
CHECK_CA_ERROR_SILENT_L(coreaudio_error);
CFIndex size =
CFStringGetMaximumSizeForEncoding(

107
audio/out/ao_coreaudio_utils.c
View File

@ -52,24 +52,24 @@ char *fourcc_repr(void *talloc_ctx, uint32_t code)
return repr;
}
bool check_ca_st(int level, OSStatus code, const char *message)
bool check_ca_st(struct ao *ao, int level, OSStatus code, const char *message)
{
if (code == noErr) return true;
char *error_string = fourcc_repr(NULL, code);
ca_msg(level, "%s (%s)\n", message, error_string);
mp_msg_log(ao->log, level, "%s (%s)\n", message, error_string);
talloc_free(error_string);
return false;
}
void ca_print_asbd(const char *description,
void ca_print_asbd(struct ao *ao, const char *description,
const AudioStreamBasicDescription *asbd)
{
uint32_t flags = asbd->mFormatFlags;
char *format = fourcc_repr(NULL, asbd->mFormatID);
ca_msg(MSGL_V,
MP_VERBOSE(ao,
"%s %7.1fHz %" PRIu32 "bit [%s]"
"[%" PRIu32 "][%" PRIu32 "][%" PRIu32 "]"
"[%" PRIu32 "][%" PRIu32 "] "
@ -98,7 +98,7 @@ bool ca_format_is_digital(AudioStreamBasicDescription asbd)
return false;
}
bool ca_stream_supports_digital(AudioStreamID stream)
bool ca_stream_supports_digital(struct ao *ao, AudioStreamID stream)
{
AudioStreamRangedDescription *formats = NULL;
size_t n_formats;
@ -111,7 +111,7 @@ bool ca_stream_supports_digital(AudioStreamID stream)
for (int i = 0; i < n_formats; i++) {
AudioStreamBasicDescription asbd = formats[i].mFormat;
ca_print_asbd("supported format:", &(asbd));
ca_print_asbd(ao, "supported format:", &(asbd));
if (ca_format_is_digital(asbd)) {
talloc_free(formats);
return true;
@ -123,7 +123,7 @@ coreaudio_error:
return false;
}
bool ca_device_supports_digital(AudioDeviceID device)
bool ca_device_supports_digital(struct ao *ao, AudioDeviceID device)
{
AudioStreamID *streams = NULL;
size_t n_streams;
@ -135,7 +135,7 @@ bool ca_device_supports_digital(AudioDeviceID device)
CHECK_CA_ERROR("could not get number of streams.");
for (int i = 0; i < n_streams; i++) {
if (ca_stream_supports_digital(streams[i])) {
if (ca_stream_supports_digital(ao, streams[i])) {
talloc_free(streams);
return true;
}
@ -156,8 +156,6 @@ OSStatus ca_property_listener(AudioObjectPropertySelector selector,
for (int i = 0; i < n_addresses; i++) {
if (addresses[i].mSelector == selector) {
ca_msg(MSGL_WARN, "event: property %s changed\n",
fourcc_repr(talloc_ctx, selector));
if (data) *(volatile int *)data = 1;
break;
}
@ -199,8 +197,8 @@ OSStatus ca_unlock_device(AudioDeviceID device, pid_t *pid) {
return noErr;
}
static OSStatus ca_change_mixing(AudioDeviceID device, uint32_t val,
bool *changed) {
static OSStatus ca_change_mixing(struct ao *ao, AudioDeviceID device,
uint32_t val, bool *changed) {
*changed = false;
AudioObjectPropertyAddress p_addr = (AudioObjectPropertyAddress) {
@ -236,14 +234,14 @@ static OSStatus ca_change_mixing(AudioDeviceID device, uint32_t val,
return noErr;
}
OSStatus ca_disable_mixing(AudioDeviceID device, bool *changed) {
return ca_change_mixing(device, 0, changed);
OSStatus ca_disable_mixing(struct ao *ao, AudioDeviceID device, bool *changed) {
return ca_change_mixing(ao, device, 0, changed);
}
OSStatus ca_enable_mixing(AudioDeviceID device, bool changed) {
OSStatus ca_enable_mixing(struct ao *ao, AudioDeviceID device, bool changed) {
if (changed) {
bool dont_care = false;
return ca_change_mixing(device, 1, &dont_care);
return ca_change_mixing(ao, device, 1, &dont_care);
}
return noErr;
@ -275,14 +273,14 @@ OSStatus ca_disable_device_listener(AudioDeviceID device, void *flag) {
return ca_change_device_listening(device, flag, false);
}
bool ca_change_format(AudioStreamID stream,
bool ca_change_format(struct ao *ao, AudioStreamID stream,
AudioStreamBasicDescription change_format)
{
OSStatus err = noErr;
AudioObjectPropertyAddress p_addr;
volatile int stream_format_changed = 0;
ca_print_asbd("setting stream format:", &change_format);
ca_print_asbd(ao, "setting stream format:", &change_format);
/* Install the callback. */
p_addr = (AudioObjectPropertyAddress) {
@ -314,13 +312,13 @@ bool ca_change_format(AudioStreamID stream,
if (stream_format_changed) {
stream_format_changed = 0;
} else {
ca_msg(MSGL_V, "reached timeout\n");
MP_VERBOSE(ao, "reached timeout\n");
}
AudioStreamBasicDescription actual_format;
err = CA_GET(stream, kAudioStreamPropertyPhysicalFormat, &actual_format);
ca_print_asbd("actual format in use:", &actual_format);
ca_print_asbd(ao, "actual format in use:", &actual_format);
if (actual_format.mSampleRate == change_format.mSampleRate &&
actual_format.mFormatID == change_format.mFormatID &&
actual_format.mFramesPerPacket == change_format.mFramesPerPacket) {
@ -338,33 +336,8 @@ bool ca_change_format(AudioStreamID stream,
return format_set;
}
void ca_bitmaps_from_layouts(AudioChannelLayout *layouts, size_t n_layouts,
uint32_t **bitmaps, size_t *n_bitmaps)
{
*n_bitmaps = 0;
*bitmaps = talloc_array_size(NULL, sizeof(uint32_t), n_layouts);
for (int i=0; i < n_layouts; i++) {
uint32_t bitmap = 0;
switch (layouts[i].mChannelLayoutTag) {
case kAudioChannelLayoutTag_UseChannelBitmap:
(*bitmaps)[(*n_bitmaps)++] = layouts[i].mChannelBitmap;
break;
case kAudioChannelLayoutTag_UseChannelDescriptions:
if (ca_bitmap_from_ch_desc(&layouts[i], &bitmap))
(*bitmaps)[(*n_bitmaps)++] = bitmap;
break;
default:
if (ca_bitmap_from_ch_tag(&layouts[i], &bitmap))
(*bitmaps)[(*n_bitmaps)++] = bitmap;
}
}
}
bool ca_bitmap_from_ch_desc(AudioChannelLayout *layout, uint32_t *bitmap)
static bool ca_bitmap_from_ch_desc(struct ao *ao, AudioChannelLayout *layout,
uint32_t *bitmap)
{
// If the channel layout uses channel descriptions, from my
// exepriments there are there three possibile cases:
@ -385,9 +358,8 @@ bool ca_bitmap_from_ch_desc(AudioChannelLayout *layout, uint32_t *bitmap)
if (label == kAudioChannelLabel_UseCoordinates ||
label == kAudioChannelLabel_Unknown ||
label > kAudioChannelLabel_TopBackRight) {
ca_msg(MSGL_V,
"channel label=%d unusable to build channel "
"bitmap, skipping layout\n", label);
MP_VERBOSE(ao, "channel label=%d unusable to build channel "
"bitmap, skipping layout\n", label);
all_channels_valid = false;
} else {
*bitmap |= 1ULL << (label - 1);
@ -397,7 +369,8 @@ bool ca_bitmap_from_ch_desc(AudioChannelLayout *layout, uint32_t *bitmap)
return all_channels_valid;
}
bool ca_bitmap_from_ch_tag(AudioChannelLayout *layout, uint32_t *bitmap)
static bool ca_bitmap_from_ch_tag(struct ao *ao, AudioChannelLayout *layout,
uint32_t *bitmap)
{
// This layout is defined exclusively by it's tag. Use the Audio
// Format Services API to try and convert it to a bitmap that
@ -410,11 +383,37 @@ bool ca_bitmap_from_ch_tag(AudioChannelLayout *layout, uint32_t *bitmap)
sizeof(AudioChannelLayoutTag), &tag,
&bitmap_size, bitmap);
if (err != noErr) {
ca_msg(MSGL_V,
"channel layout tag=%d unusable to build channel "
"bitmap, skipping layout\n", tag);
MP_VERBOSE(ao, "channel layout tag=%d unusable to build channel "
"bitmap, skipping layout\n", tag);
return false;
} else {
return true;
}
}
void ca_bitmaps_from_layouts(struct ao *ao,
AudioChannelLayout *layouts, size_t n_layouts,
uint32_t **bitmaps, size_t *n_bitmaps)
{
*n_bitmaps = 0;
*bitmaps = talloc_array_size(NULL, sizeof(uint32_t), n_layouts);
for (int i=0; i < n_layouts; i++) {
uint32_t bitmap = 0;
switch (layouts[i].mChannelLayoutTag) {
case kAudioChannelLayoutTag_UseChannelBitmap:
(*bitmaps)[(*n_bitmaps)++] = layouts[i].mChannelBitmap;
break;
case kAudioChannelLayoutTag_UseChannelDescriptions:
if (ca_bitmap_from_ch_desc(ao, &layouts[i], &bitmap))
(*bitmaps)[(*n_bitmaps)++] = bitmap;
break;
default:
if (ca_bitmap_from_ch_tag(ao, &layouts[i], &bitmap))
(*bitmaps)[(*n_bitmaps)++] = bitmap;
}
}
}

31
audio/out/ao_coreaudio_utils.h
View File

@ -23,29 +23,34 @@
#include <inttypes.h>
#include <stdbool.h>
#include "core/mp_msg.h"
#include "audio/out/ao.h"
#define ca_msg(a, b ...) mp_msg(MSGT_AO, a, "AO: [coreaudio] " b)
#define CA_CFSTR_ENCODING kCFStringEncodingASCII
char *fourcc_repr(void *talloc_ctx, uint32_t code);
bool check_ca_st(int level, OSStatus code, const char *message);
bool check_ca_st(struct ao *ao, int level, OSStatus code, const char *message);
#define CHECK_CA_ERROR_L(label, message) \
do { \
if (!check_ca_st(MSGL_ERR, err, message)) { \
if (!check_ca_st(ao, MSGL_ERR, err, message)) { \
goto label; \
} \
} while (0)
#define CHECK_CA_ERROR(message) CHECK_CA_ERROR_L(coreaudio_error, message)
#define CHECK_CA_WARN(message) check_ca_st(MSGL_WARN, err, message)
#define CHECK_CA_WARN(message) check_ca_st(ao, MSGL_WARN, err, message)
void ca_print_asbd(const char *description,
#define CHECK_CA_ERROR_SILENT_L(label) \
do { \
if (err != noErr) goto label; \
} while (0)
void ca_print_asbd(struct ao *ao, const char *description,
const AudioStreamBasicDescription *asbd);
bool ca_format_is_digital(AudioStreamBasicDescription asbd);
bool ca_stream_supports_digital(AudioStreamID stream);
bool ca_device_supports_digital(AudioDeviceID device);
bool ca_stream_supports_digital(struct ao *ao, AudioStreamID stream);
bool ca_device_supports_digital(struct ao *ao, AudioDeviceID device);
OSStatus ca_property_listener(AudioObjectPropertySelector selector,
AudioObjectID object, uint32_t n_addresses,
@ -62,17 +67,17 @@ OSStatus ca_device_listener(AudioObjectID object, uint32_t n_addresses,
OSStatus ca_lock_device(AudioDeviceID device, pid_t *pid);
OSStatus ca_unlock_device(AudioDeviceID device, pid_t *pid);
OSStatus ca_disable_mixing(AudioDeviceID device, bool *changed);
OSStatus ca_enable_mixing(AudioDeviceID device, bool changed);
OSStatus ca_disable_mixing(struct ao *ao, AudioDeviceID device, bool *changed);
OSStatus ca_enable_mixing(struct ao *ao, AudioDeviceID device, bool changed);
OSStatus ca_enable_device_listener(AudioDeviceID device, void *flag);
OSStatus ca_disable_device_listener(AudioDeviceID device, void *flag);
bool ca_change_format(AudioStreamID stream,
bool ca_change_format(struct ao *ao, AudioStreamID stream,
AudioStreamBasicDescription change_format);
bool ca_bitmap_from_ch_desc(AudioChannelLayout *layout, uint32_t *bitmap);
bool ca_bitmap_from_ch_tag(AudioChannelLayout *layout, uint32_t *bitmap);
void ca_bitmaps_from_layouts(AudioChannelLayout *layouts, size_t n_layouts,
void ca_bitmaps_from_layouts(struct ao *ao,
AudioChannelLayout *layouts, size_t n_layouts,
uint32_t **bitmaps, size_t *n_bitmaps);
#endif /* MPV_COREAUDIO_UTILS_H */